From calvin.ellison at voxox.com Sun Apr 4 20:09:58 2021 From: calvin.ellison at voxox.com (Calvin Ellison) Date: Sun, 4 Apr 2021 13:09:58 -0700 Subject: [OpenSIPS-Users] regex value too long OR how to properly quote/escape Redis raw query Message-ID: I need to post a JSON object to Elsasticsearch, and I am using Redis as a queue via cache_raw_query and the RPUSH command. The JSON object contains spaces, which appears to cause the OpenSIPS Redis client to break the string into pieces and push each of them to Redis separately. 1. Maybe I'm not quoting correctly? RPUSH snarf-cdrs 'object_goes_here' It doesn't seem to matter if I enclose the Resid list value in single quotes or not. I initially resolved this problem by escaping spaces with their escaped Unicode equivalent. This has the desired result in Elasticsearch, but now my JSON object is bigger and the regex replacement fails. Is there a workaround? $var(replaceSpace) = "/ /\\\u00a0/sg"; $avp(body) = $(json_compact(body){re.subst,$var(replaceSpace)}); DBG:core:tr_eval_re: Trying to apply regexp [/ /\\u00a0/sg] on : [{"index":"snarf-2021.04.03","time":"2021-04-03T03:34:18","TID":"006508","BLOCKED":1,"SVCPORT":"8.38.43.182","CALLID":"bf0ea503853749618c747a995b7f0102","SOURCEIP":"192.168.47.130","MEDIAIP":"192.168.47.130","ANI":"2132850555","DNIS":"2132850555","SNARF":"NONE","FRAUD":0,"TCPA":0,"NEIGHLATA":"1","NEIGHRC":"1","NEIGHBLOCK":"1","NEIGHPREFIX":"1","ATFN":0,"BTFN":0,"ALRN":"2132620105","AState":"CA","ANetwork":" BANDWIDTH.COM CLEC- LLC - CA","AOCN":"979E","ARatecenter":"LSAN DA 01","AClass":"L","ALATA":"730","ACountry":"US","AReachable":"1","AReason":"SS7 ID","ADNC":"0","AGood":"1","ADNO":"0","BLRN":"2132620105","BState":"CA","BNetwork":" BANDWIDTH.COM CLEC- LLC - CA","BOCN":"979E","BRatecenter":"LSAN DA 01","BClass":"L","BLATA":"730","BCountry":"US","BReachable":"1","BReason":"SS7 ID","BDNC":"0","BGood":"1","BDNO":"0","TRIGGERS":["neighBlock"],"RULES":{"tid":"006508","rep":"none","fraud":0,"tcpa":95,"aclass":false,"adno":false,"atfn":false,"bclass":false,"bdnc":false,"bgood":false,"neighBlock":true,"neighLata":false,"neighRc":false,"neighPrefix":false}}] Apr 3 03:34:18 ve-lab /usr/sbin/opensips[14291]: DBG:core:tr_eval_re: we must compile the regexp Apr 3 03:34:18 ve-lab /usr/sbin/opensips[14291]: DBG:core:subst_parser: ok, se is 0x7f8737c1f0e8 Apr 3 03:34:18 ve-lab /usr/sbin/opensips[14291]: ERROR:core:tr_eval_re: regex value too long [{"index":"snarf-2021.04.03","time":"2021-04-03T03:34:18","TID":"006508","BLOCKED":1,"SVCPORT":"8.38.43.182","CALLID":"bf0ea503853749618c747a995b7f0102","SOURCEIP":"192.168.47.130","MEDIAIP":"192.168.47.130","ANI":"2132850555","DNIS":"2132850555","SNARF":"NONE","FRAUD":0,"TCPA":0,"NEIGHLATA":"1","NEIGHRC":"1","NEIGHBLOCK":"1","NEIGHPREFIX":"1","ATFN":0,"BTFN":0,"ALRN":"2132620105","AState":"CA","ANetwork":" BANDWIDTH.COM CLEC- LLC - CA","AOCN":"979E","ARatecenter":"LSAN DA 01","AClass":"L","ALATA":"730","ACountry":"US","AReachable":"1","AReason":"SS7 ID","ADNC":"0","AGood":"1","ADNO":"0","BLRN":"2132620105","BState":"CA","BNetwork":" BANDWIDTH.COM CLEC- LLC - CA","BOCN":"979E","BRatecenter":"LSAN DA 01","BClass":"L","BLATA":"730","BCountry":"US","BReachable":"1","BReason":"SS7 ID","BDNC":"0","BGood":"1","BDNO":"0","TRIGGERS":["neighBlock"],"RULES":{"tid":"006508","rep":"none","fraud":0,"tcpa":95,"aclass":false,"adno":false,"atfn":false,"bclass":false,"bdnc":false,"bgood":false,"neighBlock":true,"neighLata":false,"neighRc":false,"neighPrefix":false}}] opensips 3.1.1 Ubuntu 18.04.5 LTS Linux ve-lab 4.15.0-140-generic #144-Ubuntu SMP Fri Mar 19 14:12:35 UTC 2021 x86_64 x86_64 x86_64 GNU/Linux Regards, *Calvin Ellison* Systems Architect calvin.ellison at voxox.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Tue Apr 6 09:42:41 2021 From: john.quick at smartvox.co.uk (John Quick) Date: Tue, 6 Apr 2021 10:42:41 +0100 Subject: [OpenSIPS-Users] Tips for troubleshooting OpenSIPS as a Teams SBC Message-ID: <000d01d72ac9$33528c80$99f7a580$@smartvox.co.uk> Hello all, The article I published last week talks about common issues you might encounter when commissioning a Microsoft Teams SBC solution built using OpenSIPS. It is designed to be read alongside the article by Alexey Vasilyev. https://kb.smartvox.co.uk/opensips/opensips-as-ms-teams-sbc/ John Quick Smartvox Limited From volga629 at networklab.ca Tue Apr 6 15:55:59 2021 From: volga629 at networklab.ca (volga629) Date: Tue, 6 Apr 2021 12:55:59 -0300 Subject: [OpenSIPS-Users] mid_registrar lookup In-Reply-To: <3076b43b-f15e-d23c-b2ce-1a57b8e7e364@opensips.org> References: <913584ce-6b5d-6b7b-461f-15b4dedcf4f2@skillsearch.ca> <079a32e4-a3f1-d4f5-d66f-94c28da12e7c@skillsearch.ca> <37f6d7d3-c5b4-c759-f4ed-f42d818860e0@skillsearch.ca> <3076b43b-f15e-d23c-b2ce-1a57b8e7e364@opensips.org> Message-ID: <21a0c40e-3930-0f50-91c2-3aeba2b58749@skillsearch.ca> An HTML attachment was scrubbed... URL: From kingsley at dns99.co.uk Thu Apr 8 10:55:39 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Thu, 08 Apr 2021 11:55:39 +0100 Subject: [OpenSIPS-Users] OpenSIPS 3.1 DB logging INVITEs but not BYEs Message-ID: <7b0ae2267494f20b1ac1fa43caabc9c335eef65f.camel@dns99.co.uk> Hi, I'm using OpenSIPS 3.1 with logging to DB and using the dialog module. It is logging INVITE messages, but not BYE and I can't see why. I'm new to OpenSIPS so this is puzzling me a bit. This is what I'm setting in the acc module: loadmodule "acc.so" modparam("acc", "early_media", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 0) modparam("acc", "db_url", "mysql://blah...") My main route script starts with this: script_trace(1, "method $rm"); I'm doing this when an INVITE comes in: t_newtran(); create_dialog("pPB"); do_accounting("db", "cdr|missed|failed"); I have this, but it made no difference (although "BYE RECEIVED" did appear in the opensips log): if (is_method("BYE")) { # do accounting even if the transaction fails xlog("BYE RECEIVED"); do_accounting("db", "cdr|missed|failed"); } I have this event route: event_route[E_DLG_STATE_CHANGED] { switch ($param(7)) { case 1: xlog("** DLG ** INVITE SENT"); break; case 2: xlog("** DLG ** PROVISIONAL REPLY"); break; case 3: xlog("** DLG ** PROVISIONALLY ANSWERED"); break; case 4: xlog("** DLG ** CONFIRMED ANSWERED"); break; case 5: xlog("** DLG ** BYE RECEIVED - ENDED"); break; } } and I see this in the opensips log when the call ends (I see "do_accounting" in the trace): /usr/sbin/opensips[22551]: ** DLG ** BYE RECEIVED - ENDED /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:225][core if] -> (method BYE) /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:220][module validate_dialog] -> (method BYE) /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:230][core if] -> (method BYE) /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:225][module is_method] -> (method BYE) /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:238][core if] -> (method BYE) /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:230][module is_method] -> (method BYE) /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:233][module do_accounting] -> (method BYE) /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:238][route relay] -> (method BYE) /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:431][core if] -> (method BYE) /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:428][module t_relay] -> (method BYE) /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:431][core exit] -> (method BYE) But the acc table only ever gets INVITEs logged; here are records for the last 2 test calls: MariaDB [opensips]> SELECT id,method,callid,sip_code FROm acc ORDER BY id DESC LIMIT 4; +-----+--------+--------------------------------------+----------+ | id | method | callid | sip_code | +-----+--------+--------------------------------------+----------+ | 234 | INVITE | 18d9cfb9-5add-4824-9d19-b828c10df9f4 | 200 | | 233 | INVITE | 18d9cfb9-5add-4824-9d19-b828c10df9f4 | 183 | | 232 | INVITE | a3a3f3b3-3ac0-45de-8cb7-f208b2d420d0 | 200 | | 231 | INVITE | a3a3f3b3-3ac0-45de-8cb7-f208b2d420d0 | 183 | +-----+--------+--------------------------------------+----------+ Am I missing something obvious? Cheers, Kingsley. From marcin at voipplus.net Thu Apr 8 12:10:55 2021 From: marcin at voipplus.net (Marcin Groszek) Date: Thu, 8 Apr 2021 07:10:55 -0500 Subject: [OpenSIPS-Users] OpenSIPS 3.1 DB logging INVITEs but not BYEs In-Reply-To: <7b0ae2267494f20b1ac1fa43caabc9c335eef65f.camel@dns99.co.uk> References: <7b0ae2267494f20b1ac1fa43caabc9c335eef65f.camel@dns99.co.uk> Message-ID: <28a72cb1-79d0-7f41-8909-a2ebc83752b6@voipplus.net> remove t_newtran(); On 4/8/2021 5:55 AM, Kingsley Tart wrote: > Hi, > > I'm using OpenSIPS 3.1 with logging to DB and using the dialog module. > > It is logging INVITE messages, but not BYE and I can't see why. I'm new > to OpenSIPS so this is puzzling me a bit. > > > This is what I'm setting in the acc module: > > loadmodule "acc.so" > modparam("acc", "early_media", 1) > modparam("acc", "report_cancels", 1) > modparam("acc", "detect_direction", 0) > modparam("acc", "db_url", "mysql://blah...") > > > My main route script starts with this: > > script_trace(1, "method $rm"); > > > I'm doing this when an INVITE comes in: > > t_newtran(); > create_dialog("pPB"); > do_accounting("db", "cdr|missed|failed"); > > > I have this, but it made no difference (although "BYE RECEIVED" did > appear in the opensips log): > > if (is_method("BYE")) { > # do accounting even if the transaction fails > xlog("BYE RECEIVED"); > do_accounting("db", "cdr|missed|failed"); > } > > > I have this event route: > > event_route[E_DLG_STATE_CHANGED] { > switch ($param(7)) { > case 1: > xlog("** DLG ** INVITE SENT"); > break; > case 2: > xlog("** DLG ** PROVISIONAL REPLY"); > break; > case 3: > xlog("** DLG ** PROVISIONALLY ANSWERED"); > break; > case 4: > xlog("** DLG ** CONFIRMED ANSWERED"); > break; > case 5: > xlog("** DLG ** BYE RECEIVED - ENDED"); > break; > } > } > > > and I see this in the opensips log when the call ends (I see > "do_accounting" in the trace): > > /usr/sbin/opensips[22551]: ** DLG ** BYE RECEIVED - ENDED > /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:225][core if] -> (method BYE) > /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:220][module validate_dialog] -> (method BYE) > /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:230][core if] -> (method BYE) > /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:225][module is_method] -> (method BYE) > /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:238][core if] -> (method BYE) > /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:230][module is_method] -> (method BYE) > /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:233][module do_accounting] -> (method BYE) > /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:238][route relay] -> (method BYE) > /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:431][core if] -> (method BYE) > /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:428][module t_relay] -> (method BYE) > /usr/sbin/opensips[22552]: [Script Trace][/etc/opensips/opensips.cfg:431][core exit] -> (method BYE) > > > But the acc table only ever gets INVITEs logged; here are records for > the last 2 test calls: > > MariaDB [opensips]> SELECT id,method,callid,sip_code FROm acc ORDER BY > id DESC LIMIT 4; > +-----+--------+--------------------------------------+----------+ > | id | method | callid | sip_code | > +-----+--------+--------------------------------------+----------+ > | 234 | INVITE | 18d9cfb9-5add-4824-9d19-b828c10df9f4 | 200 | > | 233 | INVITE | 18d9cfb9-5add-4824-9d19-b828c10df9f4 | 183 | > | 232 | INVITE | a3a3f3b3-3ac0-45de-8cb7-f208b2d420d0 | 200 | > | 231 | INVITE | a3a3f3b3-3ac0-45de-8cb7-f208b2d420d0 | 183 | > +-----+--------+--------------------------------------+----------+ > > > Am I missing something obvious? > > Cheers, > Kingsley. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Best Regards: Marcin Groszek Business Voip Resource. http://www.voipplus.net From kingsley at dns99.co.uk Thu Apr 8 13:14:44 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Thu, 08 Apr 2021 14:14:44 +0100 Subject: [OpenSIPS-Users] OpenSIPS 3.1 DB logging INVITEs but not BYEs In-Reply-To: <28a72cb1-79d0-7f41-8909-a2ebc83752b6@voipplus.net> References: <7b0ae2267494f20b1ac1fa43caabc9c335eef65f.camel@dns99.co.uk> <28a72cb1-79d0-7f41-8909-a2ebc83752b6@voipplus.net> Message-ID: Thanks, though that didn't seem to have made any difference :( It *is* logging durations though (something you mentioned in your other email). Cheers, Kingsley. On Thu, 2021-04-08 at 07:10 -0500, Marcin Groszek wrote: > remove > > t_newtran(); > > > On 4/8/2021 5:55 AM, Kingsley Tart wrote: > > Hi, > > > > I'm using OpenSIPS 3.1 with logging to DB and using the dialog > > module. > > > > It is logging INVITE messages, but not BYE and I can't see why. I'm > > new > > to OpenSIPS so this is puzzling me a bit. > > > > > > This is what I'm setting in the acc module: > > > > loadmodule "acc.so" > > modparam("acc", "early_media", 1) > > modparam("acc", "report_cancels", 1) > > modparam("acc", "detect_direction", 0) > > modparam("acc", "db_url", "mysql://blah...") > > > > > > My main route script starts with this: > > > > script_trace(1, "method $rm"); > > > > > > I'm doing this when an INVITE comes in: > > > > t_newtran(); > > create_dialog("pPB"); > > do_accounting("db", "cdr|missed|failed"); > > > > > > I have this, but it made no difference (although "BYE RECEIVED" did > > appear in the opensips log): > > > > if (is_method("BYE")) { > > # do accounting even if the transaction fails > > xlog("BYE RECEIVED"); > > do_accounting("db", "cdr|missed|failed"); > > } > > > > > > I have this event route: > > > > event_route[E_DLG_STATE_CHANGED] { > > switch ($param(7)) { > > case 1: > > xlog("** DLG ** INVITE SENT"); > > break; > > case 2: > > xlog("** DLG ** PROVISIONAL REPLY"); > > break; > > case 3: > > xlog("** DLG ** PROVISIONALLY ANSWERED"); > > break; > > case 4: > > xlog("** DLG ** CONFIRMED ANSWERED"); > > break; > > case 5: > > xlog("** DLG ** BYE RECEIVED - ENDED"); > > break; > > } > > } > > > > > > and I see this in the opensips log when the call ends (I see > > "do_accounting" in the trace): > > > > /usr/sbin/opensips[22551]: ** DLG ** BYE RECEIVED - ENDED > > /usr/sbin/opensips[22552]: [Script > > Trace][/etc/opensips/opensips.cfg:225][core if] -> (method BYE) > > /usr/sbin/opensips[22552]: [Script > > Trace][/etc/opensips/opensips.cfg:220][module validate_dialog] -> > > (method BYE) > > /usr/sbin/opensips[22552]: [Script > > Trace][/etc/opensips/opensips.cfg:230][core if] -> (method BYE) > > /usr/sbin/opensips[22552]: [Script > > Trace][/etc/opensips/opensips.cfg:225][module is_method] -> (method > > BYE) > > /usr/sbin/opensips[22552]: [Script > > Trace][/etc/opensips/opensips.cfg:238][core if] -> (method BYE) > > /usr/sbin/opensips[22552]: [Script > > Trace][/etc/opensips/opensips.cfg:230][module is_method] -> (method > > BYE) > > /usr/sbin/opensips[22552]: [Script > > Trace][/etc/opensips/opensips.cfg:233][module do_accounting] -> > > (method BYE) > > /usr/sbin/opensips[22552]: [Script > > Trace][/etc/opensips/opensips.cfg:238][route relay] -> (method BYE) > > /usr/sbin/opensips[22552]: [Script > > Trace][/etc/opensips/opensips.cfg:431][core if] -> (method BYE) > > /usr/sbin/opensips[22552]: [Script > > Trace][/etc/opensips/opensips.cfg:428][module t_relay] -> (method > > BYE) > > /usr/sbin/opensips[22552]: [Script > > Trace][/etc/opensips/opensips.cfg:431][core exit] -> (method BYE) > > > > > > But the acc table only ever gets INVITEs logged; here are records > > for > > the last 2 test calls: > > > > MariaDB [opensips]> SELECT id,method,callid,sip_code FROm acc ORDER > > BY > > id DESC LIMIT 4; > > +-----+--------+--------------------------------------+----------+ > > > id | method | callid | sip_code | > > > > +-----+--------+--------------------------------------+----------+ > > > 234 | INVITE | 18d9cfb9-5add-4824-9d19-b828c10df9f4 | 200 | > > > 233 | INVITE | 18d9cfb9-5add-4824-9d19-b828c10df9f4 | 183 | > > > 232 | INVITE | a3a3f3b3-3ac0-45de-8cb7-f208b2d420d0 | 200 | > > > 231 | INVITE | a3a3f3b3-3ac0-45de-8cb7-f208b2d420d0 | 183 | > > > > +-----+--------+--------------------------------------+----------+ > > > > > > Am I missing something obvious? > > > > Cheers, > > Kingsley. > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > From marcin at voipplus.net Thu Apr 8 14:17:30 2021 From: marcin at voipplus.net (Marcin Groszek) Date: Thu, 8 Apr 2021 09:17:30 -0500 Subject: [OpenSIPS-Users] OpenSIPS 3.1 DB logging INVITEs but not BYEs In-Reply-To: References: Message-ID: <18D1DEDB-79DE-46E7-892A-68E589CB625A@voipplus.net> Because you are using dialog that invite entry in acc table is written at the time of bye. I prefer a single entry in acc table for each call, not 2 Best regards: Marcin Groszek > On Apr 8, 2021, at 8:15 AM, Kingsley Tart wrote: > > Thanks, though that didn't seem to have made any difference :( > > It *is* logging durations though (something you mentioned in your other > email). > > Cheers, > Kingsley. > >> On Thu, 2021-04-08 at 07:10 -0500, Marcin Groszek wrote: >> remove >> >> t_newtran(); >> >> >>> On 4/8/2021 5:55 AM, Kingsley Tart wrote: >>> Hi, >>> >>> I'm using OpenSIPS 3.1 with logging to DB and using the dialog >>> module. >>> >>> It is logging INVITE messages, but not BYE and I can't see why. I'm >>> new >>> to OpenSIPS so this is puzzling me a bit. >>> >>> >>> This is what I'm setting in the acc module: >>> >>> loadmodule "acc.so" >>> modparam("acc", "early_media", 1) >>> modparam("acc", "report_cancels", 1) >>> modparam("acc", "detect_direction", 0) >>> modparam("acc", "db_url", "mysql://blah...") >>> >>> >>> My main route script starts with this: >>> >>> script_trace(1, "method $rm"); >>> >>> >>> I'm doing this when an INVITE comes in: >>> >>> t_newtran(); >>> create_dialog("pPB"); >>> do_accounting("db", "cdr|missed|failed"); >>> >>> >>> I have this, but it made no difference (although "BYE RECEIVED" did >>> appear in the opensips log): >>> >>> if (is_method("BYE")) { >>> # do accounting even if the transaction fails >>> xlog("BYE RECEIVED"); >>> do_accounting("db", "cdr|missed|failed"); >>> } >>> >>> >>> I have this event route: >>> >>> event_route[E_DLG_STATE_CHANGED] { >>> switch ($param(7)) { >>> case 1: >>> xlog("** DLG ** INVITE SENT"); >>> break; >>> case 2: >>> xlog("** DLG ** PROVISIONAL REPLY"); >>> break; >>> case 3: >>> xlog("** DLG ** PROVISIONALLY ANSWERED"); >>> break; >>> case 4: >>> xlog("** DLG ** CONFIRMED ANSWERED"); >>> break; >>> case 5: >>> xlog("** DLG ** BYE RECEIVED - ENDED"); >>> break; >>> } >>> } >>> >>> >>> and I see this in the opensips log when the call ends (I see >>> "do_accounting" in the trace): >>> >>> /usr/sbin/opensips[22551]: ** DLG ** BYE RECEIVED - ENDED >>> /usr/sbin/opensips[22552]: [Script >>> Trace][/etc/opensips/opensips.cfg:225][core if] -> (method BYE) >>> /usr/sbin/opensips[22552]: [Script >>> Trace][/etc/opensips/opensips.cfg:220][module validate_dialog] -> >>> (method BYE) >>> /usr/sbin/opensips[22552]: [Script >>> Trace][/etc/opensips/opensips.cfg:230][core if] -> (method BYE) >>> /usr/sbin/opensips[22552]: [Script >>> Trace][/etc/opensips/opensips.cfg:225][module is_method] -> (method >>> BYE) >>> /usr/sbin/opensips[22552]: [Script >>> Trace][/etc/opensips/opensips.cfg:238][core if] -> (method BYE) >>> /usr/sbin/opensips[22552]: [Script >>> Trace][/etc/opensips/opensips.cfg:230][module is_method] -> (method >>> BYE) >>> /usr/sbin/opensips[22552]: [Script >>> Trace][/etc/opensips/opensips.cfg:233][module do_accounting] -> >>> (method BYE) >>> /usr/sbin/opensips[22552]: [Script >>> Trace][/etc/opensips/opensips.cfg:238][route relay] -> (method BYE) >>> /usr/sbin/opensips[22552]: [Script >>> Trace][/etc/opensips/opensips.cfg:431][core if] -> (method BYE) >>> /usr/sbin/opensips[22552]: [Script >>> Trace][/etc/opensips/opensips.cfg:428][module t_relay] -> (method >>> BYE) >>> /usr/sbin/opensips[22552]: [Script >>> Trace][/etc/opensips/opensips.cfg:431][core exit] -> (method BYE) >>> >>> >>> But the acc table only ever gets INVITEs logged; here are records >>> for >>> the last 2 test calls: >>> >>> MariaDB [opensips]> SELECT id,method,callid,sip_code FROm acc ORDER >>> BY >>> id DESC LIMIT 4; >>> +-----+--------+--------------------------------------+----------+ >>>> id | method | callid | sip_code | >>> >>> +-----+--------+--------------------------------------+----------+ >>>> 234 | INVITE | 18d9cfb9-5add-4824-9d19-b828c10df9f4 | 200 | >>>> 233 | INVITE | 18d9cfb9-5add-4824-9d19-b828c10df9f4 | 183 | >>>> 232 | INVITE | a3a3f3b3-3ac0-45de-8cb7-f208b2d420d0 | 200 | >>>> 231 | INVITE | a3a3f3b3-3ac0-45de-8cb7-f208b2d420d0 | 183 | >>> >>> +-----+--------+--------------------------------------+----------+ >>> >>> >>> Am I missing something obvious? >>> >>> Cheers, >>> Kingsley. >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From john.quick at smartvox.co.uk Thu Apr 8 14:45:50 2021 From: john.quick at smartvox.co.uk (John Quick) Date: Thu, 8 Apr 2021 15:45:50 +0100 Subject: [OpenSIPS-Users] AUTH_DB module database queries Message-ID: <000d01d72c85$e1b3f9e0$a51beda0$@smartvox.co.uk> I am auditing an existing opensips.cfg script trying to identify all possible places where a database query is executed. The purpose is to optimise performance by reducing the number of DB queries and eliminating any unnecessary synchronous database queries. I would like to know more about what happens when the script calls the proxy_authorize() function. I assume it must be querying the subscriber table, but is this a synchronous query? Does the AUTH_DB module do any caching of data or will it always read directly from the DB table? Is there any difference between v2.4 and v3.1 here? Is there anything on the roadmap that may be relevant? The behaviour of proxy_authorize() could be relevant to system performance and reliability in the following situations: - An OpenSIPS system under heavy load with many call attempts per second - Where an unexpected network or DB-server issue results in slow DB query responses - potentially crashing OpenSIPS - If a "rogue" customer device or malicious hacking attempt generates large numbers of requests which all call proxy_authorize() John Quick Smartvox Limited From bogdan at opensips.org Thu Apr 8 15:38:34 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 8 Apr 2021 18:38:34 +0300 Subject: [OpenSIPS-Users] AUTH_DB module database queries In-Reply-To: <000d01d72c85$e1b3f9e0$a51beda0$@smartvox.co.uk> References: <000d01d72c85$e1b3f9e0$a51beda0$@smartvox.co.uk> Message-ID: <01b12e3e-5125-31ec-8b6e-19f987bec6ee@opensips.org> Hi John, See inline answers. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/8/21 5:45 PM, John Quick wrote: > I am auditing an existing opensips.cfg script trying to identify all > possible places where a database query is executed. > The purpose is to optimise performance by reducing the number of DB queries > and eliminating any unnecessary synchronous database queries. > > I would like to know more about what happens when the script calls the > proxy_authorize() function. > I assume it must be querying the subscriber table, but is this a synchronous > query? yes, it is a blocking query. > Does the AUTH_DB module do any caching of data or will it always read > directly from the DB table? no caching > Is there any difference between v2.4 and v3.1 here? no > Is there anything on the roadmap that may be relevant? no > > The behaviour of proxy_authorize() could be relevant to system performance > and reliability in the following situations: > - An OpenSIPS system under heavy load with many call attempts per second > - Where an unexpected network or DB-server issue results in slow DB query > responses - potentially crashing OpenSIPS > - If a "rogue" customer device or malicious hacking attempt generates large > numbers of requests which all call proxy_authorize() you can replace the proxy_authorize() with pv_proxy_authorize() (where you provide the pwd from script) and manage the pwd retrieval from script level, with async queries + caching. > > John Quick > Smartvox Limited > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From john.quick at smartvox.co.uk Thu Apr 8 15:49:36 2021 From: john.quick at smartvox.co.uk (John Quick) Date: Thu, 8 Apr 2021 16:49:36 +0100 Subject: [OpenSIPS-Users] AUTH_DB module database queries In-Reply-To: <01b12e3e-5125-31ec-8b6e-19f987bec6ee@opensips.org> References: <000d01d72c85$e1b3f9e0$a51beda0$@smartvox.co.uk> <01b12e3e-5125-31ec-8b6e-19f987bec6ee@opensips.org> Message-ID: <001001d72c8e$ca34b8f0$5e9e2ad0$@smartvox.co.uk> Thanks Bogdan. John Quick Smartvox Limited -----Original Message----- From: Bogdan-Andrei Iancu Sent: 08 April 2021 16:39 To: john.quick at smartvox.co.uk; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] AUTH_DB module database queries Hi John, See inline answers. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/8/21 5:45 PM, John Quick wrote: > I am auditing an existing opensips.cfg script trying to identify all > possible places where a database query is executed. > The purpose is to optimise performance by reducing the number of DB > queries and eliminating any unnecessary synchronous database queries. > > I would like to know more about what happens when the script calls the > proxy_authorize() function. > I assume it must be querying the subscriber table, but is this a > synchronous query? yes, it is a blocking query. > Does the AUTH_DB module do any caching of data or will it always read > directly from the DB table? no caching > Is there any difference between v2.4 and v3.1 here? no > Is there anything on the roadmap that may be relevant? no > > The behaviour of proxy_authorize() could be relevant to system > performance and reliability in the following situations: > - An OpenSIPS system under heavy load with many call attempts per second > - Where an unexpected network or DB-server issue results in slow DB > query responses - potentially crashing OpenSIPS > - If a "rogue" customer device or malicious hacking attempt > generates large numbers of requests which all call proxy_authorize() you can replace the proxy_authorize() with pv_proxy_authorize() (where you provide the pwd from script) and manage the pwd retrieval from script level, with async queries + caching. > > John Quick > Smartvox Limited > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From kingsley at dns99.co.uk Thu Apr 8 16:09:06 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Thu, 08 Apr 2021 17:09:06 +0100 Subject: [OpenSIPS-Users] OpenSIPS 3.1 DB logging INVITEs but not BYEs In-Reply-To: <18D1DEDB-79DE-46E7-892A-68E589CB625A@voipplus.net> References: <18D1DEDB-79DE-46E7-892A-68E589CB625A@voipplus.net> Message-ID: I would prefer a single record as well. I don't understand why using the dialog module makes it write a new "INVITE" record at BYE time instead of either writing a BYE record, or allowing me to update it with a few custom fields, one of which would be t_end or something. Cheers, Kingsley. On Thu, 2021-04-08 at 09:17 -0500, Marcin Groszek wrote: > Because you are using dialog that invite entry in acc table is > written at the time of bye. > I prefer a single entry in acc table for each call, not 2 > > Best regards: > Marcin Groszek > > > On Apr 8, 2021, at 8:15 AM, Kingsley Tart > > wrote: > > > > Thanks, though that didn't seem to have made any difference :( > > > > It *is* logging durations though (something you mentioned in your > > other > > email). > > > > Cheers, > > Kingsley. From kingsley at dns99.co.uk Thu Apr 8 17:06:36 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Thu, 08 Apr 2021 18:06:36 +0100 Subject: [OpenSIPS-Users] Matching dialog IDs Message-ID: <376c743d9b1c2fc41580592c07199b673ab87835.camel@dns99.co.uk> Hi, With OpenSIPS 3.1, on one example call I noted that: * did in Record-Route header was 5af.b5c82a2 * dlg_id in DB table was 17201388227675 * event_route[E_DLG_STATE_CHANGED] got hash_entry=4005 (param 1) and hash_id=44207195 (param 2) I see these 3 sets of values are all essentially the same thing but in a different format. Is this guaranteed behaviour? eg in the event route, this expression: $(param(1){s.dec2hex}{s.reverse}{s.tolower}) + "." + $(param(2){s.dec2hex}{s.reverse}{s.tolower}); evaluates to "5af.b5c82a2", ie the did in the Record-Route header, and that in MySQL I could derive the dlg_id value in the DB table (which was 17201388227675) by doing XXXX*POW(2,32)+YYYYYYY where XXXX is the value of $param(1) and YYYYYYY is the value of $param(2). There may be a better way to do all this, but I'm trying to figure out ways I can update the DB with some additional info at BYE time and haven't currently found out how to do this more properly. Cheers, Kingsley. From marcin at voipplus.net Thu Apr 8 17:26:22 2021 From: marcin at voipplus.net (Marcin Groszek) Date: Thu, 8 Apr 2021 12:26:22 -0500 Subject: [OpenSIPS-Users] OpenSIPS 3.1 DB logging INVITEs but not BYEs In-Reply-To: References: <18D1DEDB-79DE-46E7-892A-68E589CB625A@voipplus.net> Message-ID: <865adb51-94e7-d818-77df-561563af25cc@voipplus.net> At initial invite some information are stored like a start time, this is needed to calculate the duration at the call end. I believe if you do: do_accounting("db","cdr") this will write only 1 invite entry in acc at the end of the call, but if you do: do_accounting("db") this will write your invite and bye as 2 entries in acc table. make a call, select * from acc at your db hung up the call and select again you will see when the entries are written. You may also use ngrep -port 3306 -dlo to monitor db access by opensips but you have to change localhost to 127.0.0.1 in modparam. modparam("acc", "db_url", "mysql://opensips:opensipsrw at 127.0.0.1/opensips") On 4/8/2021 11:09 AM, Kingsley Tart wrote: > I would prefer a single record as well. I don't understand why using > the dialog module makes it write a new "INVITE" record at BYE time > instead of either writing a BYE record, or allowing me to update it > with a few custom fields, one of which would be t_end or something. > > Cheers, > Kingsley. > > On Thu, 2021-04-08 at 09:17 -0500, Marcin Groszek wrote: >> Because you are using dialog that invite entry in acc table is >> written at the time of bye. >> I prefer a single entry in acc table for each call, not 2 >> >> Best regards: >> Marcin Groszek >> >>> On Apr 8, 2021, at 8:15 AM, Kingsley Tart >>> wrote: >>> >>> Thanks, though that didn't seem to have made any difference :( >>> >>> It *is* logging durations though (something you mentioned in your >>> other >>> email). >>> >>> Cheers, >>> Kingsley. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Best Regards: Marcin Groszek Business Voip Resource. http://www.voipplus.net From calvin.ellison at voxox.com Thu Apr 8 21:02:55 2021 From: calvin.ellison at voxox.com (Calvin Ellison) Date: Thu, 8 Apr 2021 14:02:55 -0700 Subject: [OpenSIPS-Users] regex value too long OR how to properly quote/escape Redis raw query In-Reply-To: References: Message-ID: Any suggestion for resolving "regex value too long" when using the re.subst transformation? Any way to fix the Redis client so it does not break on spaces when the value is single-quoted? From what I understand, modern Redis servers support single-quoted strings. On Sun, Apr 4, 2021 at 1:09 PM Calvin Ellison wrote: > I need to post a JSON object to Elsasticsearch, and I am using Redis as a > queue via cache_raw_query and the RPUSH command. The JSON object contains > spaces, which appears to cause the OpenSIPS Redis client to break the > string into pieces and push each of them to Redis separately. > > 1. Maybe I'm not quoting correctly? > RPUSH snarf-cdrs 'object_goes_here' > > It doesn't seem to matter if I enclose the Resid list value in single > quotes or not. > > I initially resolved this problem by escaping spaces with their escaped > Unicode equivalent. This has the desired result in Elasticsearch, but now > my JSON object is bigger and the regex replacement fails. Is there a > workaround? > > $var(replaceSpace) = "/ /\\\u00a0/sg"; > $avp(body) = $(json_compact(body){re.subst,$var(replaceSpace)}); > > DBG:core:tr_eval_re: Trying to apply regexp [/ /\\u00a0/sg] on : > [{"index":"snarf-2021.04.03","time":"2021-04-03T03:34:18","TID":"006508","BLOCKED":1,"SVCPORT":"8.38.43.182","CALLID":"bf0ea503853749618c747a995b7f0102","SOURCEIP":"192.168.47.130","MEDIAIP":"192.168.47.130","ANI":"2132850555","DNIS":"2132850555","SNARF":"NONE","FRAUD":0,"TCPA":0,"NEIGHLATA":"1","NEIGHRC":"1","NEIGHBLOCK":"1","NEIGHPREFIX":"1","ATFN":0,"BTFN":0,"ALRN":"2132620105","AState":"CA","ANetwork":" > BANDWIDTH.COM CLEC- LLC - CA","AOCN":"979E","ARatecenter":"LSAN DA > 01","AClass":"L","ALATA":"730","ACountry":"US","AReachable":"1","AReason":"SS7 > ID","ADNC":"0","AGood":"1","ADNO":"0","BLRN":"2132620105","BState":"CA","BNetwork":" > BANDWIDTH.COM CLEC- LLC - CA","BOCN":"979E","BRatecenter":"LSAN DA > 01","BClass":"L","BLATA":"730","BCountry":"US","BReachable":"1","BReason":"SS7 > ID","BDNC":"0","BGood":"1","BDNO":"0","TRIGGERS":["neighBlock"],"RULES":{"tid":"006508","rep":"none","fraud":0,"tcpa":95,"aclass":false,"adno":false,"atfn":false,"bclass":false,"bdnc":false,"bgood":false,"neighBlock":true,"neighLata":false,"neighRc":false,"neighPrefix":false}}] > Apr 3 03:34:18 ve-lab /usr/sbin/opensips[14291]: DBG:core:tr_eval_re: we > must compile the regexp > Apr 3 03:34:18 ve-lab /usr/sbin/opensips[14291]: DBG:core:subst_parser: > ok, se is 0x7f8737c1f0e8 > Apr 3 03:34:18 ve-lab /usr/sbin/opensips[14291]: ERROR:core:tr_eval_re: > regex value too long > [{"index":"snarf-2021.04.03","time":"2021-04-03T03:34:18","TID":"006508","BLOCKED":1,"SVCPORT":"8.38.43.182","CALLID":"bf0ea503853749618c747a995b7f0102","SOURCEIP":"192.168.47.130","MEDIAIP":"192.168.47.130","ANI":"2132850555","DNIS":"2132850555","SNARF":"NONE","FRAUD":0,"TCPA":0,"NEIGHLATA":"1","NEIGHRC":"1","NEIGHBLOCK":"1","NEIGHPREFIX":"1","ATFN":0,"BTFN":0,"ALRN":"2132620105","AState":"CA","ANetwork":" > BANDWIDTH.COM CLEC- LLC - CA","AOCN":"979E","ARatecenter":"LSAN DA > 01","AClass":"L","ALATA":"730","ACountry":"US","AReachable":"1","AReason":"SS7 > ID","ADNC":"0","AGood":"1","ADNO":"0","BLRN":"2132620105","BState":"CA","BNetwork":" > BANDWIDTH.COM CLEC- LLC - CA","BOCN":"979E","BRatecenter":"LSAN DA > 01","BClass":"L","BLATA":"730","BCountry":"US","BReachable":"1","BReason":"SS7 > ID","BDNC":"0","BGood":"1","BDNO":"0","TRIGGERS":["neighBlock"],"RULES":{"tid":"006508","rep":"none","fraud":0,"tcpa":95,"aclass":false,"adno":false,"atfn":false,"bclass":false,"bdnc":false,"bgood":false,"neighBlock":true,"neighLata":false,"neighRc":false,"neighPrefix":false}}] > > opensips 3.1.1 > Ubuntu 18.04.5 LTS > Linux ve-lab 4.15.0-140-generic #144-Ubuntu SMP Fri Mar 19 14:12:35 UTC > 2021 x86_64 x86_64 x86_64 GNU/Linux > > > Regards, > > *Calvin Ellison* > Systems Architect > calvin.ellison at voxox.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kingsley at dns99.co.uk Thu Apr 8 23:04:56 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Fri, 09 Apr 2021 00:04:56 +0100 Subject: [OpenSIPS-Users] Matching dialog IDs Message-ID: <7101010a5423f6e4992a5077f0450872576e1db9.camel@dns99.co.uk> Hi, With OpenSIPS 3.1, on one example call I noted that: * did in Record-Route header was 5af.b5c82a2 * dlg_id in DB table was 17201388227675 * event_route[E_DLG_STATE_CHANGED] got hash_entry=4005 (param 1) and hash_id=44207195 (param 2) I see these 3 sets of values are all essentially the same thing but in a different format. In a way it feels wrong to ask this but, is this guaranteed behaviour? eg in the event route, this expression: $(param(1){s.dec2hex}{s.reverse}{s.tolower}) + "." + $(param(2){s.dec2hex}{s.reverse}{s.tolower}); evaluates to "5af.b5c82a2", ie the did in the Record-Route header, and that in MySQL I could derive the dlg_id value in the DB table (which was 17201388227675) by doing XXXX*POW(2,32)+YYYYYYY where XXXX is the value of $param(1) and YYYYYYY is the value of $param(2). There may be a better way to do all this, but I'm trying to figure out ways I can update the DB with some additional info at BYE time and haven't currently found out how to do this more properly. (again, this feels the wrong way to do it, but I have not yet found a better way). Cheers, Kingsley. From razvan at opensips.org Fri Apr 9 08:01:08 2021 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 9 Apr 2021 11:01:08 +0300 Subject: [OpenSIPS-Users] Matching dialog IDs In-Reply-To: <7101010a5423f6e4992a5077f0450872576e1db9.camel@dns99.co.uk> References: <7101010a5423f6e4992a5077f0450872576e1db9.camel@dns99.co.uk> Message-ID: <6364cebd-6c4c-7ba2-6519-57325ead9c82@opensips.org> Hi, Kingsley! I am afraid your approach is the correct one, and currently there's no other workaround, rather than the one you already did :(. Last year I've reworked the did and dlg_id in a few places to have a more consistent value, but somehow I missed the event's one. Ideally, we should have only two formats: the hex one (the one in Record-Route, dlg_list and $DLG_did), and the db_id one (the integer in DB). I will try to rework this in the master branch, any chance you could open a feature request for it? Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 4/9/21 2:04 AM, Kingsley Tart wrote: > Hi, > > With OpenSIPS 3.1, on one example call I noted that: > > * did in Record-Route header was 5af.b5c82a2 > * dlg_id in DB table was 17201388227675 > * event_route[E_DLG_STATE_CHANGED] got hash_entry=4005 (param 1) and > hash_id=44207195 (param 2) > > I see these 3 sets of values are all essentially the same thing but in > a different format. > > In a way it feels wrong to ask this but, is this guaranteed behaviour? > > > eg in the event route, this expression: > > $(param(1){s.dec2hex}{s.reverse}{s.tolower}) + "." + > $(param(2){s.dec2hex}{s.reverse}{s.tolower}); > > evaluates to "5af.b5c82a2", ie the did in the Record-Route header, and > that in MySQL I could derive the dlg_id value in the DB table (which > was 17201388227675) by doing XXXX*POW(2,32)+YYYYYYY where XXXX is the > value of $param(1) and YYYYYYY is the value of $param(2). > > There may be a better way to do all this, but I'm trying to figure out > ways I can update the DB with some additional info at BYE time and > haven't currently found out how to do this more properly. > > (again, this feels the wrong way to do it, but I have not yet found a > better way). > > Cheers, > Kingsley. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From bullehs at gmail.com Fri Apr 9 09:39:00 2021 From: bullehs at gmail.com (HS) Date: Fri, 9 Apr 2021 14:39:00 +0500 Subject: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications Message-ID: Hello all. We are using Opensips 3.0 (with rtpproxy) and been looking to implement Push Notifications for iOS devices. Did come across quite a few resources, but they seem either too old ~2013 time or about V3.1. Has anyone implemented something to make calls to devices with the UAC in background mode on V3.0 pls? Or suggestions on how to make this work on v3.0? http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/ Appreciate the help. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From kingsley at dns99.co.uk Fri Apr 9 10:58:27 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Fri, 09 Apr 2021 11:58:27 +0100 Subject: [OpenSIPS-Users] Matching dialog IDs In-Reply-To: <6364cebd-6c4c-7ba2-6519-57325ead9c82@opensips.org> References: <7101010a5423f6e4992a5077f0450872576e1db9.camel@dns99.co.uk> <6364cebd-6c4c-7ba2-6519-57325ead9c82@opensips.org> Message-ID: <3012d8c576b6308258ac605f7c058c83beadaf64.camel@dns99.co.uk> What would you like me to specifically request, and how do I submit a feature request for OpenSIPS? Cheers, Kingsley. On Fri, 2021-04-09 at 11:01 +0300, Răzvan Crainea wrote: > Hi, Kingsley! > > I am afraid your approach is the correct one, and currently there's no > other workaround, rather than the one you already did :(. > Last year I've reworked the did and dlg_id in a few places to have a > more consistent value, but somehow I missed the event's one. Ideally, we > should have only two formats: the hex one (the one in Record-Route, > dlg_list and $DLG_did), and the db_id one (the integer in DB). > I will try to rework this in the master branch, any chance you could > open a feature request for it? > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 4/9/21 2:04 AM, Kingsley Tart wrote: > > Hi, > > > > With OpenSIPS 3.1, on one example call I noted that: > > > > * did in Record-Route header was 5af.b5c82a2 > > * dlg_id in DB table was 17201388227675 > > * event_route[E_DLG_STATE_CHANGED] got hash_entry=4005 (param 1) and > > hash_id=44207195 (param 2) > > > > I see these 3 sets of values are all essentially the same thing but in > > a different format. > > > > In a way it feels wrong to ask this but, is this guaranteed behaviour? > > > > > > eg in the event route, this expression: > > > > $(param(1){s.dec2hex}{s.reverse}{s.tolower}) + "." + > > $(param(2){s.dec2hex}{s.reverse}{s.tolower}); > > > > evaluates to "5af.b5c82a2", ie the did in the Record-Route header, and > > that in MySQL I could derive the dlg_id value in the DB table (which > > was 17201388227675) by doing XXXX*POW(2,32)+YYYYYYY where XXXX is the > > value of $param(1) and YYYYYYY is the value of $param(2). > > > > There may be a better way to do all this, but I'm trying to figure out > > ways I can update the DB with some additional info at BYE time and > > haven't currently found out how to do this more properly. > > > > (again, this feels the wrong way to do it, but I have not yet found a > > better way). > > > > Cheers, > > Kingsley. > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Fri Apr 9 11:01:42 2021 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 9 Apr 2021 14:01:42 +0300 Subject: [OpenSIPS-Users] Matching dialog IDs In-Reply-To: <3012d8c576b6308258ac605f7c058c83beadaf64.camel@dns99.co.uk> References: <7101010a5423f6e4992a5077f0450872576e1db9.camel@dns99.co.uk> <6364cebd-6c4c-7ba2-6519-57325ead9c82@opensips.org> <3012d8c576b6308258ac605f7c058c83beadaf64.camel@dns99.co.uk> Message-ID: <0d662004-a7bd-f1f8-23d5-f3fb8a3ae263@opensips.org> Hi, Kingsley! Just open an issue on OpenSIPS github page[1], it would be easier for us to keep track of them. You will find the procedure there. I think the feature is to also provide the dialog id in the event. [1] https://github.com/OpenSIPS/opensips/issues Thanks, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 4/9/21 1:58 PM, Kingsley Tart wrote: > What would you like me to specifically request, and how do I submit a > feature request for OpenSIPS? > > Cheers, > Kingsley. > > On Fri, 2021-04-09 at 11:01 +0300, Răzvan Crainea wrote: >> Hi, Kingsley! >> >> I am afraid your approach is the correct one, and currently there's no >> other workaround, rather than the one you already did :(. >> Last year I've reworked the did and dlg_id in a few places to have a >> more consistent value, but somehow I missed the event's one. Ideally, we >> should have only two formats: the hex one (the one in Record-Route, >> dlg_list and $DLG_did), and the db_id one (the integer in DB). >> I will try to rework this in the master branch, any chance you could >> open a feature request for it? >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Core Developer >> http://www.opensips-solutions.com >> >> On 4/9/21 2:04 AM, Kingsley Tart wrote: >>> Hi, >>> >>> With OpenSIPS 3.1, on one example call I noted that: >>> >>> * did in Record-Route header was 5af.b5c82a2 >>> * dlg_id in DB table was 17201388227675 >>> * event_route[E_DLG_STATE_CHANGED] got hash_entry=4005 (param 1) and >>> hash_id=44207195 (param 2) >>> >>> I see these 3 sets of values are all essentially the same thing but in >>> a different format. >>> >>> In a way it feels wrong to ask this but, is this guaranteed behaviour? >>> >>> >>> eg in the event route, this expression: >>> >>> $(param(1){s.dec2hex}{s.reverse}{s.tolower}) + "." + >>> $(param(2){s.dec2hex}{s.reverse}{s.tolower}); >>> >>> evaluates to "5af.b5c82a2", ie the did in the Record-Route header, and >>> that in MySQL I could derive the dlg_id value in the DB table (which >>> was 17201388227675) by doing XXXX*POW(2,32)+YYYYYYY where XXXX is the >>> value of $param(1) and YYYYYYY is the value of $param(2). >>> >>> There may be a better way to do all this, but I'm trying to figure out >>> ways I can update the DB with some additional info at BYE time and >>> haven't currently found out how to do this more properly. >>> >>> (again, this feels the wrong way to do it, but I have not yet found a >>> better way). >>> >>> Cheers, >>> Kingsley. >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From razvan at opensips.org Fri Apr 9 11:10:23 2021 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 9 Apr 2021 14:10:23 +0300 Subject: [OpenSIPS-Users] OpenSIPS 3.1 DB logging INVITEs but not BYEs In-Reply-To: <865adb51-94e7-d818-77df-561563af25cc@voipplus.net> References: <18D1DEDB-79DE-46E7-892A-68E589CB625A@voipplus.net> <865adb51-94e7-d818-77df-561563af25cc@voipplus.net> Message-ID: Hi, Marcin, Kingsley! Indeed, if you'll be running only do_accounting("db", "cdr"), you will only get one record, at the end of the call, containing the `INVITE` method. That's more or less hardcoded there, and can't be changed. Giving a second thought about it, I somehow agree that's not 100% suggestive or correct, although I was the one who wrote that code :). The idea was to create a CDR format compatible with previous formats of acc, which were requiring a method. But since a call consists of several methods, the easiest way to pick one was to pick the first one, namely INVITE. Hence the CDR always has an INVITE method. If you have other thoughts on this, any feedback is welcome. Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 4/8/21 8:26 PM, Marcin Groszek wrote: > At initial invite some information are stored like a start time, this is > needed to calculate the duration at the call end. > > I believe if you do: do_accounting("db","cdr") this will write only 1 > invite entry in acc at the end of the call, > > but if you do: do_accounting("db") this will write your invite and bye > as 2 entries in acc table. > > make a call, select * from acc at your db hung up the call and select > again you will see when the entries are written. > > You may also use ngrep -port 3306 -dlo to monitor db access by opensips > but you have to change localhost to 127.0.0.1 in modparam. > > modparam("acc", "db_url", "mysql://opensips:opensipsrw at 127.0.0.1/opensips") > > > > On 4/8/2021 11:09 AM, Kingsley Tart wrote: >> I would prefer a single record as well. I don't understand why using >> the dialog module makes it write a new "INVITE" record at BYE time >> instead of either writing a BYE record, or allowing me to update it >> with a few custom fields, one of which would be t_end or something. >> >> Cheers, >> Kingsley. >> >> On Thu, 2021-04-08 at 09:17 -0500, Marcin Groszek wrote: >>> Because you are using dialog that invite entry in acc table is >>> written at the time of bye. >>> I prefer a single entry in acc table for each call, not 2 >>> >>> Best regards: >>> Marcin Groszek >>> >>>> On Apr 8, 2021, at 8:15 AM, Kingsley Tart >>>> wrote: >>>> >>>> Thanks, though that didn't seem to have made any difference :( >>>> >>>> It *is* logging durations though (something you mentioned in your >>>> other >>>> email). >>>> >>>> Cheers, >>>> Kingsley. >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> From razvan at opensips.org Fri Apr 9 11:17:34 2021 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 9 Apr 2021 14:17:34 +0300 Subject: [OpenSIPS-Users] regex value too long OR how to properly quote/escape Redis raw query In-Reply-To: References: Message-ID: Hi, Calvin! The "regex value too long" error should have already been fixed[1], but I somehow forgot to inform you :(. Can you please run another test now and let us know if there are other issues? [1] https://github.com/OpenSIPS/opensips/commit/a8f8aabc245ca86bd40a9eddf9befcd74a087be1 Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 4/9/21 12:02 AM, Calvin Ellison wrote: > Any suggestion for resolving "regex value too long" when using > the re.subst transformation? > > Any way to fix the Redis client so it does not break on spaces when the > value is single-quoted? From what I understand, modern Redis servers > support single-quoted strings. > > > On Sun, Apr 4, 2021 at 1:09 PM Calvin Ellison > wrote: > > I need to post a JSON object to Elsasticsearch, and I am using Redis > as a queue via cache_raw_query and the RPUSH command. The JSON > object contains spaces, which appears to cause the OpenSIPS Redis > client to break the string into pieces and push each of them to > Redis separately. > > 1. Maybe I'm not quoting correctly? > RPUSH snarf-cdrs 'object_goes_here' > > It doesn't seem to matter if I enclose the Resid list value in > single quotes or not. > > I initially resolved this problem by escaping spaces with their > escaped Unicode equivalent. This has the desired result in > Elasticsearch, but now my JSON object is bigger and the regex > replacement fails. Is there a workaround? > > $var(replaceSpace) = "/ /\\\u00a0/sg"; > $avp(body) = $(json_compact(body){re.subst,$var(replaceSpace)}); > > DBG:core:tr_eval_re: Trying to apply regexp [/ /\\u00a0/sg] on : > [{"index":"snarf-2021.04.03","time":"2021-04-03T03:34:18","TID":"006508","BLOCKED":1,"SVCPORT":"8.38.43.182","CALLID":"bf0ea503853749618c747a995b7f0102","SOURCEIP":"192.168.47.130","MEDIAIP":"192.168.47.130","ANI":"2132850555","DNIS":"2132850555","SNARF":"NONE","FRAUD":0,"TCPA":0,"NEIGHLATA":"1","NEIGHRC":"1","NEIGHBLOCK":"1","NEIGHPREFIX":"1","ATFN":0,"BTFN":0,"ALRN":"2132620105","AState":"CA","ANetwork":"BANDWIDTH.COM > CLEC- LLC - > CA","AOCN":"979E","ARatecenter":"LSAN DA > 01","AClass":"L","ALATA":"730","ACountry":"US","AReachable":"1","AReason":"SS7 > ID","ADNC":"0","AGood":"1","ADNO":"0","BLRN":"2132620105","BState":"CA","BNetwork":"BANDWIDTH.COM > CLEC- LLC - > CA","BOCN":"979E","BRatecenter":"LSAN DA > 01","BClass":"L","BLATA":"730","BCountry":"US","BReachable":"1","BReason":"SS7 > ID","BDNC":"0","BGood":"1","BDNO":"0","TRIGGERS":["neighBlock"],"RULES":{"tid":"006508","rep":"none","fraud":0,"tcpa":95,"aclass":false,"adno":false,"atfn":false,"bclass":false,"bdnc":false,"bgood":false,"neighBlock":true,"neighLata":false,"neighRc":false,"neighPrefix":false}}] > Apr  3 03:34:18 ve-lab /usr/sbin/opensips[14291]: > DBG:core:tr_eval_re: we must compile the regexp > Apr  3 03:34:18 ve-lab /usr/sbin/opensips[14291]: > DBG:core:subst_parser: ok, se is 0x7f8737c1f0e8 > Apr  3 03:34:18 ve-lab /usr/sbin/opensips[14291]: > ERROR:core:tr_eval_re: regex value too long > [{"index":"snarf-2021.04.03","time":"2021-04-03T03:34:18","TID":"006508","BLOCKED":1,"SVCPORT":"8.38.43.182","CALLID":"bf0ea503853749618c747a995b7f0102","SOURCEIP":"192.168.47.130","MEDIAIP":"192.168.47.130","ANI":"2132850555","DNIS":"2132850555","SNARF":"NONE","FRAUD":0,"TCPA":0,"NEIGHLATA":"1","NEIGHRC":"1","NEIGHBLOCK":"1","NEIGHPREFIX":"1","ATFN":0,"BTFN":0,"ALRN":"2132620105","AState":"CA","ANetwork":"BANDWIDTH.COM > CLEC- LLC - > CA","AOCN":"979E","ARatecenter":"LSAN DA > 01","AClass":"L","ALATA":"730","ACountry":"US","AReachable":"1","AReason":"SS7 > ID","ADNC":"0","AGood":"1","ADNO":"0","BLRN":"2132620105","BState":"CA","BNetwork":"BANDWIDTH.COM > CLEC- LLC - > CA","BOCN":"979E","BRatecenter":"LSAN DA > 01","BClass":"L","BLATA":"730","BCountry":"US","BReachable":"1","BReason":"SS7 > ID","BDNC":"0","BGood":"1","BDNO":"0","TRIGGERS":["neighBlock"],"RULES":{"tid":"006508","rep":"none","fraud":0,"tcpa":95,"aclass":false,"adno":false,"atfn":false,"bclass":false,"bdnc":false,"bgood":false,"neighBlock":true,"neighLata":false,"neighRc":false,"neighPrefix":false}}] > > opensips 3.1.1 > Ubuntu 18.04.5 LTS > Linux ve-lab 4.15.0-140-generic #144-Ubuntu SMP Fri Mar 19 14:12:35 > UTC 2021 x86_64 x86_64 x86_64 GNU/Linux > > > Regards, > > *Calvin Ellison* > Systems Architect > calvin.ellison at voxox.com > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From kingsley at dns99.co.uk Fri Apr 9 11:41:07 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Fri, 09 Apr 2021 12:41:07 +0100 Subject: [OpenSIPS-Users] OpenSIPS 3.1 DB logging INVITEs but not BYEs In-Reply-To: <865adb51-94e7-d818-77df-561563af25cc@voipplus.net> References: <18D1DEDB-79DE-46E7-892A-68E589CB625A@voipplus.net> <865adb51-94e7-d818-77df-561563af25cc@voipplus.net> Message-ID: <063e14ab3d117e70d0a7b4bbd43ebb4c53e1ec12.camel@dns99.co.uk> Hi, I don't know whether you know but MariaDB v10 (that's the version I happen to be using; I don't know about other versions or specifically MySQL) has a general log that can be enabled and if on, logs all queries, thus avoiding the need to use ngrep; eg: MariaDB [opensips]> SHOW VARIABLES LIKE '%general%'; +------------------+--------------------+ | Variable_name | Value | +------------------+--------------------+ | general_log | OFF | | general_log_file | bt-ipx-proxy-b.log | +------------------+--------------------+ 2 rows in set (0.001 sec) MariaDB [opensips]> SET @@GLOBAL.general_log=1; Query OK, 0 rows affected (0.012 sec) MariaDB [opensips]> SHOW VARIABLES LIKE '%general%'; +------------------+--------------------+ | Variable_name | Value | +------------------+--------------------+ | general_log | ON | | general_log_file | bt-ipx-proxy-b.log | +------------------+--------------------+ 2 rows in set (0.002 sec) MariaDB [opensips]> exit Bye # cat /var/lib/mysql/bt-ipx-proxy-b.log /usr/sbin/mysqld, Version: 10.3.27-MariaDB-0+deb10u1 (Debian 10). started with: Tcp port: 3306 Unix socket: /run/mysqld/mysqld.sock Time Id Command Argument 210409 12:34:59 334515 Query SHOW VARIABLES LIKE '%general%' 210409 12:36:35 334515 Quit (ie it logged my querying of the variables the second time now that the setting was enabled). Obviously it's not something you'd want to leave enabled for long in some environments :) Cheers, Kingsley. On Thu, 2021-04-08 at 12:26 -0500, Marcin Groszek wrote: > At initial invite some information are stored like a start time, this is > needed to calculate the duration at the call end. > > I believe if you do: do_accounting("db","cdr") this will write only 1 > invite entry in acc at the end of the call, > > but if you do: do_accounting("db") this will write your invite and bye > as 2 entries in acc table. > > make a call, select * from acc at your db hung up the call and select > again you will see when the entries are written. > > You may also use ngrep -port 3306 -dlo to monitor db access by opensips > but you have to change localhost to 127.0.0.1 in modparam. > > modparam("acc", "db_url", "mysql://opensips:opensipsrw at 127.0.0.1/opensips") From kingsley at dns99.co.uk Fri Apr 9 11:45:00 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Fri, 09 Apr 2021 12:45:00 +0100 Subject: [OpenSIPS-Users] OpenSIPS 3.1 DB logging INVITEs but not BYEs In-Reply-To: References: <18D1DEDB-79DE-46E7-892A-68E589CB625A@voipplus.net> <865adb51-94e7-d818-77df-561563af25cc@voipplus.net> Message-ID: <60fe1d13b3b5e16cfe36e39e9af0c5196d6e9e16.camel@dns99.co.uk> Hi, I found that changing this: modparam("acc", "early_media", 1) to this: modparam("acc", "early_media", 0) Has stopped the multiple records. With early_media on, a 180 or 183 created a new record. I already have that info in custom fields anyway by having this (just testing, so "doodle" was as good a name as any at the time :) ): onreply_route[doodle] { if (t_check_status("^1[0-9][0-9]$")) { switch ($T_reply_code) { case 180: $acc_extra(t_ringing) = $Ts; break; case 183: $acc_extra(t_progress) = $Ts; break; } } else if (t_check_status("^2[0-9][0-9]$")) { $acc_extra(t_answer) = $Ts; } else { $acc_extra(t_clear) = $Ts; } } though the t_clear field only gets logged for missed calls, not answered ones (but the call duration for answered calls IS logged in the duration field, so I suppose t_clear is superfluous in the acc table, really). I tried putting the setting of $acc_extra(t_clear) into event_route[E_DLG_STATE_CHANGED] when new state=5 but this didn't work either. I don't actually mind not getting the BYE logged specifically as long as I can tell when it happened (as I can from the duration field). It's just that I was following an example in Bogdan's book about OpenSIPS 2.1 that seemed to expect a BYE to be logged separately. Cheers, Kingsley. On Fri, 2021-04-09 at 14:10 +0300, Răzvan Crainea wrote: > Hi, Marcin, Kingsley! > > Indeed, if you'll be running only do_accounting("db", "cdr"), you will > only get one record, at the end of the call, containing the `INVITE` > method. That's more or less hardcoded there, and can't be changed. > Giving a second thought about it, I somehow agree that's not 100% > suggestive or correct, although I was the one who wrote that code :). > The idea was to create a CDR format compatible with previous formats of > acc, which were requiring a method. But since a call consists of several > methods, the easiest way to pick one was to pick the first one, namely > INVITE. Hence the CDR always has an INVITE method. > If you have other thoughts on this, any feedback is welcome. > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 4/8/21 8:26 PM, Marcin Groszek wrote: > > At initial invite some information are stored like a start time, this is > > needed to calculate the duration at the call end. > > > > I believe if you do: do_accounting("db","cdr") this will write only 1 > > invite entry in acc at the end of the call, > > > > but if you do: do_accounting("db") this will write your invite and bye > > as 2 entries in acc table. > > > > make a call, select * from acc at your db hung up the call and select > > again you will see when the entries are written. > > > > You may also use ngrep -port 3306 -dlo to monitor db access by opensips > > but you have to change localhost to 127.0.0.1 in modparam. > > > > modparam("acc", "db_url", "mysql://opensips:opensipsrw at 127.0.0.1/opensips") > > > > > > > > On 4/8/2021 11:09 AM, Kingsley Tart wrote: > > > I would prefer a single record as well. I don't understand why using > > > the dialog module makes it write a new "INVITE" record at BYE time > > > instead of either writing a BYE record, or allowing me to update it > > > with a few custom fields, one of which would be t_end or something. > > > > > > Cheers, > > > Kingsley. > > > > > > On Thu, 2021-04-08 at 09:17 -0500, Marcin Groszek wrote: > > > > Because you are using dialog that invite entry in acc table is > > > > written at the time of bye. > > > > I prefer a single entry in acc table for each call, not 2 > > > > > > > > Best regards: > > > > Marcin Groszek > > > > > > > > > On Apr 8, 2021, at 8:15 AM, Kingsley Tart > > > > > wrote: > > > > > > > > > > Thanks, though that didn't seem to have made any difference :( > > > > > > > > > > It *is* logging durations though (something you mentioned in your > > > > > other > > > > > email). > > > > > > > > > > Cheers, > > > > > Kingsley. > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From kingsley at dns99.co.uk Fri Apr 9 13:22:33 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Fri, 09 Apr 2021 14:22:33 +0100 Subject: [OpenSIPS-Users] Matching dialog IDs In-Reply-To: <0d662004-a7bd-f1f8-23d5-f3fb8a3ae263@opensips.org> References: <7101010a5423f6e4992a5077f0450872576e1db9.camel@dns99.co.uk> <6364cebd-6c4c-7ba2-6519-57325ead9c82@opensips.org> <3012d8c576b6308258ac605f7c058c83beadaf64.camel@dns99.co.uk> <0d662004-a7bd-f1f8-23d5-f3fb8a3ae263@opensips.org> Message-ID: Here you go: https://github.com/OpenSIPS/opensips/issues/2463 Cheers, Kingsley. On Fri, 2021-04-09 at 14:01 +0300, Răzvan Crainea wrote: > Hi, Kingsley! > > Just open an issue on OpenSIPS github page[1], it would be easier for us > to keep track of them. You will find the procedure there. > I think the feature is to also provide the dialog id in the event. > > [1] https://github.com/OpenSIPS/opensips/issues > > Thanks, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 4/9/21 1:58 PM, Kingsley Tart wrote: > > What would you like me to specifically request, and how do I submit a > > feature request for OpenSIPS? > > > > Cheers, > > Kingsley. > > > > On Fri, 2021-04-09 at 11:01 +0300, Răzvan Crainea wrote: > > > Hi, Kingsley! > > > > > > I am afraid your approach is the correct one, and currently there's no > > > other workaround, rather than the one you already did :(. > > > Last year I've reworked the did and dlg_id in a few places to have a > > > more consistent value, but somehow I missed the event's one. Ideally, we > > > should have only two formats: the hex one (the one in Record-Route, > > > dlg_list and $DLG_did), and the db_id one (the integer in DB). > > > I will try to rework this in the master branch, any chance you could > > > open a feature request for it? > > > > > > Best regards, > > > > > > Răzvan Crainea > > > OpenSIPS Core Developer > > > http://www.opensips-solutions.com > > > > > > On 4/9/21 2:04 AM, Kingsley Tart wrote: > > > > Hi, > > > > > > > > With OpenSIPS 3.1, on one example call I noted that: > > > > > > > > * did in Record-Route header was 5af.b5c82a2 > > > > * dlg_id in DB table was 17201388227675 > > > > * event_route[E_DLG_STATE_CHANGED] got hash_entry=4005 (param 1) and > > > > hash_id=44207195 (param 2) > > > > > > > > I see these 3 sets of values are all essentially the same thing but in > > > > a different format. > > > > > > > > In a way it feels wrong to ask this but, is this guaranteed behaviour? > > > > > > > > > > > > eg in the event route, this expression: > > > > > > > > $(param(1){s.dec2hex}{s.reverse}{s.tolower}) + "." + > > > > $(param(2){s.dec2hex}{s.reverse}{s.tolower}); > > > > > > > > evaluates to "5af.b5c82a2", ie the did in the Record-Route header, and > > > > that in MySQL I could derive the dlg_id value in the DB table (which > > > > was 17201388227675) by doing XXXX*POW(2,32)+YYYYYYY where XXXX is the > > > > value of $param(1) and YYYYYYY is the value of $param(2). > > > > > > > > There may be a better way to do all this, but I'm trying to figure out > > > > ways I can update the DB with some additional info at BYE time and > > > > haven't currently found out how to do this more properly. > > > > > > > > (again, this feels the wrong way to do it, but I have not yet found a > > > > better way). > > > > > > > > Cheers, > > > > Kingsley. > > > > > > > > From xaled at web.de Fri Apr 9 15:49:49 2021 From: xaled at web.de (xaled at web.de) Date: Fri, 9 Apr 2021 17:49:49 +0200 Subject: [OpenSIPS-Users] use of scenario_id in b2b_init_request Message-ID: <000201d72d57$f98c86e0$eca594a0$@web.de> Hi, I'm trying to understand the B2B module in 3.2 and don't quite get the meaning of scenario_id in b2b_init_request function. There are multiple examples that use different values in scenario_id https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2 b2b_init_request("prepaid"); or b2b_init_request("refer"); but the values "prepaid" or "refer" not used to differentiate anything within route[b2b_logic_request] or anywhere else in documentation. I get the meaning of req_route parameter of b2b_init_request function to address different B2b logic using different routes, but what does the scenario_id used for? Thanks, Xaled From vladp at opensips.org Fri Apr 9 17:11:33 2021 From: vladp at opensips.org (Vlad Patrascu) Date: Fri, 9 Apr 2021 20:11:33 +0300 Subject: [OpenSIPS-Users] use of scenario_id in b2b_init_request In-Reply-To: <000201d72d57$f98c86e0$eca594a0$@web.de> References: <000201d72d57$f98c86e0$eca594a0$@web.de> Message-ID: <12103fb4-4b04-13b1-6e5c-f9495359e1fc@opensips.org> Hi Xaled, The scenario ID is just a transparent identifier for the current B2B session. You may use it to identify which session belongs to which of your B2B scenarios. You can see this value in the output of b2b_list MI function and also you can retrieve it in the script with $b2b_logic.scenario (this may be useful if you use a single B2B request route, as in the example here: [1]). Another purpose of the "scenario_id" parameter of b2b_init_request() is to be able to choose the internal topology hiding scenario, by providing the special value "top hiding". [1] https://opensips.org/docs/modules/3.2.x/b2b_logic.html#b2b_logic.scenario Regards, -- Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 09.04.2021 18:49, xaled at web.de wrote: > Hi, > > I'm trying to understand the B2B module in 3.2 and don't quite get the meaning of scenario_id in b2b_init_request function. > > There are multiple examples that use different values in scenario_id > https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2 > > b2b_init_request("prepaid"); or b2b_init_request("refer"); but the values "prepaid" or "refer" not used to differentiate anything within route[b2b_logic_request] or anywhere else in documentation. > > I get the meaning of req_route parameter of b2b_init_request function to address different B2b logic using different routes, but what does the scenario_id used for? > > Thanks, > Xaled > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From calvin.ellison at voxox.com Fri Apr 9 17:24:58 2021 From: calvin.ellison at voxox.com (Calvin Ellison) Date: Fri, 9 Apr 2021 10:24:58 -0700 Subject: [OpenSIPS-Users] regex value too long OR how to properly quote/escape Redis raw query In-Reply-To: References: Message-ID: Thanks, Răzvan! Better late than never :) Should we expect to see this fix in 3.1 LTS soon? Regards, *Calvin Ellison* Systems Architect calvin.ellison at voxox.com +1 (213) 285-0555 ----------------------------------------------- *voxox.com * 5825 Oberlin Drive, Suite 5 San Diego, CA 92121 [image: Voxox] On Fri, Apr 9, 2021 at 4:18 AM Răzvan Crainea wrote: > Hi, Calvin! > > The "regex value too long" error should have already been fixed[1], but > I somehow forgot to inform you :(. Can you please run another test now > and let us know if there are other issues? > > [1] > > https://github.com/OpenSIPS/opensips/commit/a8f8aabc245ca86bd40a9eddf9befcd74a087be1 > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 4/9/21 12:02 AM, Calvin Ellison wrote: > > Any suggestion for resolving "regex value too long" when using > > the re.subst transformation? > > > > Any way to fix the Redis client so it does not break on spaces when the > > value is single-quoted? From what I understand, modern Redis servers > > support single-quoted strings. > > > > > > On Sun, Apr 4, 2021 at 1:09 PM Calvin Ellison > > wrote: > > > > I need to post a JSON object to Elsasticsearch, and I am using Redis > > as a queue via cache_raw_query and the RPUSH command. The JSON > > object contains spaces, which appears to cause the OpenSIPS Redis > > client to break the string into pieces and push each of them to > > Redis separately. > > > > 1. Maybe I'm not quoting correctly? > > RPUSH snarf-cdrs 'object_goes_here' > > > > It doesn't seem to matter if I enclose the Resid list value in > > single quotes or not. > > > > I initially resolved this problem by escaping spaces with their > > escaped Unicode equivalent. This has the desired result in > > Elasticsearch, but now my JSON object is bigger and the regex > > replacement fails. Is there a workaround? > > > > $var(replaceSpace) = "/ /\\\u00a0/sg"; > > $avp(body) = $(json_compact(body){re.subst,$var(replaceSpace)}); > > > > DBG:core:tr_eval_re: Trying to apply regexp [/ /\\u00a0/sg] on : > > > [{"index":"snarf-2021.04.03","time":"2021-04-03T03:34:18","TID":"006508","BLOCKED":1,"SVCPORT":"8.38.43.182","CALLID":"bf0ea503853749618c747a995b7f0102","SOURCEIP":"192.168.47.130","MEDIAIP":"192.168.47.130","ANI":"2132850555","DNIS":"2132850555","SNARF":"NONE","FRAUD":0,"TCPA":0,"NEIGHLATA":"1","NEIGHRC":"1","NEIGHBLOCK":"1","NEIGHPREFIX":"1","ATFN":0,"BTFN":0,"ALRN":"2132620105","AState":"CA","ANetwork":" > BANDWIDTH.COM > > CLEC- LLC - > > CA","AOCN":"979E","ARatecenter":"LSAN DA > > > 01","AClass":"L","ALATA":"730","ACountry":"US","AReachable":"1","AReason":"SS7 > > > ID","ADNC":"0","AGood":"1","ADNO":"0","BLRN":"2132620105","BState":"CA","BNetwork":" > BANDWIDTH.COM > > CLEC- LLC - > > CA","BOCN":"979E","BRatecenter":"LSAN DA > > > 01","BClass":"L","BLATA":"730","BCountry":"US","BReachable":"1","BReason":"SS7 > > > ID","BDNC":"0","BGood":"1","BDNO":"0","TRIGGERS":["neighBlock"],"RULES":{"tid":"006508","rep":"none","fraud":0,"tcpa":95,"aclass":false,"adno":false,"atfn":false,"bclass":false,"bdnc":false,"bgood":false,"neighBlock":true,"neighLata":false,"neighRc":false,"neighPrefix":false}}] > > Apr 3 03:34:18 ve-lab /usr/sbin/opensips[14291]: > > DBG:core:tr_eval_re: we must compile the regexp > > Apr 3 03:34:18 ve-lab /usr/sbin/opensips[14291]: > > DBG:core:subst_parser: ok, se is 0x7f8737c1f0e8 > > Apr 3 03:34:18 ve-lab /usr/sbin/opensips[14291]: > > ERROR:core:tr_eval_re: regex value too long > > > [{"index":"snarf-2021.04.03","time":"2021-04-03T03:34:18","TID":"006508","BLOCKED":1,"SVCPORT":"8.38.43.182","CALLID":"bf0ea503853749618c747a995b7f0102","SOURCEIP":"192.168.47.130","MEDIAIP":"192.168.47.130","ANI":"2132850555","DNIS":"2132850555","SNARF":"NONE","FRAUD":0,"TCPA":0,"NEIGHLATA":"1","NEIGHRC":"1","NEIGHBLOCK":"1","NEIGHPREFIX":"1","ATFN":0,"BTFN":0,"ALRN":"2132620105","AState":"CA","ANetwork":" > BANDWIDTH.COM > > CLEC- LLC - > > CA","AOCN":"979E","ARatecenter":"LSAN DA > > > 01","AClass":"L","ALATA":"730","ACountry":"US","AReachable":"1","AReason":"SS7 > > > ID","ADNC":"0","AGood":"1","ADNO":"0","BLRN":"2132620105","BState":"CA","BNetwork":" > BANDWIDTH.COM > > CLEC- LLC - > > CA","BOCN":"979E","BRatecenter":"LSAN DA > > > 01","BClass":"L","BLATA":"730","BCountry":"US","BReachable":"1","BReason":"SS7 > > > ID","BDNC":"0","BGood":"1","BDNO":"0","TRIGGERS":["neighBlock"],"RULES":{"tid":"006508","rep":"none","fraud":0,"tcpa":95,"aclass":false,"adno":false,"atfn":false,"bclass":false,"bdnc":false,"bgood":false,"neighBlock":true,"neighLata":false,"neighRc":false,"neighPrefix":false}}] > > > > opensips 3.1.1 > > Ubuntu 18.04.5 LTS > > Linux ve-lab 4.15.0-140-generic #144-Ubuntu SMP Fri Mar 19 14:12:35 > > UTC 2021 x86_64 x86_64 x86_64 GNU/Linux > > > > > > Regards, > > > > *Calvin Ellison* > > Systems Architect > > calvin.ellison at voxox.com > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Fri Apr 9 21:06:55 2021 From: xaled at web.de (xaled at web.de) Date: Fri, 9 Apr 2021 23:06:55 +0200 Subject: [OpenSIPS-Users] use of scenario_id in b2b_init_request In-Reply-To: <12103fb4-4b04-13b1-6e5c-f9495359e1fc@opensips.org> References: <000201d72d57$f98c86e0$eca594a0$@web.de> <12103fb4-4b04-13b1-6e5c-f9495359e1fc@opensips.org> Message-ID: <012c01d72d84$45dc08a0$d19419e0$@web.de> Hi Vlad, thank you, I missed the $b2b_logic.scenario variable that can be used to differentiate logic within route[b2b_logic_request]. Now it is clear to me. It would be good to include it as an example in the B2B tutorial or B2B module documentation. Greetings, Xaled -----Original Message----- From: Users On Behalf Of Vlad Patrascu Sent: Friday, April 9, 2021 7:12 PM To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] use of scenario_id in b2b_init_request Hi Xaled, The scenario ID is just a transparent identifier for the current B2B session. You may use it to identify which session belongs to which of your B2B scenarios. You can see this value in the output of b2b_list MI function and also you can retrieve it in the script with $b2b_logic.scenario (this may be useful if you use a single B2B request route, as in the example here: [1]). Another purpose of the "scenario_id" parameter of b2b_init_request() is to be able to choose the internal topology hiding scenario, by providing the special value "top hiding". [1] https://opensips.org/docs/modules/3.2.x/b2b_logic.html#b2b_logic.scenario Regards, -- Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 09.04.2021 18:49, xaled at web.de wrote: > Hi, > > I'm trying to understand the B2B module in 3.2 and don't quite get the meaning of scenario_id in b2b_init_request function. > > There are multiple examples that use different values in scenario_id > https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2 > > b2b_init_request("prepaid"); or b2b_init_request("refer"); but the values "prepaid" or "refer" not used to differentiate anything within route[b2b_logic_request] or anywhere else in documentation. > > I get the meaning of req_route parameter of b2b_init_request function to address different B2b logic using different routes, but what does the scenario_id used for? > > Thanks, > Xaled > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From xaled at web.de Sun Apr 11 18:32:34 2021 From: xaled at web.de (xaled at web.de) Date: Sun, 11 Apr 2021 20:32:34 +0200 Subject: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module Message-ID: <019b01d72f01$0a957a00$1fc06e00$@web.de> Hi, would it be possible to respect loose routing on the caller side of B2B module and set route header on the callee side accordingly Something like this: Incoming INVITE: INVITE sip:+12345678 at 1.2.3.4;user=phone SIP/2.0 Via: SIP/2.0/TCP 4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb Route: , Wanted outgoing INVITE: INVITE sip:+87654321 at 1.1.1.1;user=phone SIP/2.0 Via: SIP/2.0/TCP test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb Route: I tried setting loose_route in both the original route and in the resulting b2b route but without success. route[b2b_test_logic] { <------>loose_route(); if ($rm != "BYE") { # for requests other than BYE, no special actions needs to be done, # just pass the request to the peer b2b_pass_request(); exit; } ... Thanks, Xaled From xaled at web.de Sun Apr 11 21:36:21 2021 From: xaled at web.de (xaled at web.de) Date: Sun, 11 Apr 2021 23:36:21 +0200 Subject: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module In-Reply-To: <019b01d72f01$0a957a00$1fc06e00$@web.de> References: <019b01d72f01$0a957a00$1fc06e00$@web.de> Message-ID: <01f901d72f1a$b6fd55b0$24f80110$@web.de> Hi, I tried to use the extra_hdrs variable of b2b_client_new to set up the Route header extra_hdrs (var, optional) - AVP variable holding a list of extra headers (the header names) to be added for any request sent to this entity. But was confused by the "header names" part and it looks like that it will not accept headers with values. What is then the use of the variable? xlog("INFO: B2B $b2b_logic.ctx(next_route)"); INFO: B2B Route: $avp(route_header) = $b2b_logic.ctx(next_route); b2b_client_new("media", "sip:+87654321 at 1.1.1.1;user=phone", "sip:$b2b_logic.ctx(server_ip)", ,$avp(route_header)); ERROR:b2b_logic:b2bl_entity_new: header names without values! The need to have an avp for extra_hdrs is an unnecessary limitation, or is there a specific reason for it? Thanks, Xaled -----Original Message----- From: Users On Behalf Of xaled at web.de Sent: Sunday, April 11, 2021 8:33 PM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module Hi, would it be possible to respect loose routing on the caller side of B2B module and set route header on the callee side accordingly Something like this: Incoming INVITE: INVITE sip:+12345678 at 1.2.3.4;user=phone SIP/2.0 Via: SIP/2.0/TCP 4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb Route: , Wanted outgoing INVITE: INVITE sip:+87654321 at 1.1.1.1;user=phone SIP/2.0 Via: SIP/2.0/TCP test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb Route: I tried setting loose_route in both the original route and in the resulting b2b route but without success. route[b2b_test_logic] { <------>loose_route(); if ($rm != "BYE") { # for requests other than BYE, no special actions needs to be done, # just pass the request to the peer b2b_pass_request(); exit; } ... Thanks, Xaled _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From xaled at web.de Sun Apr 11 21:59:25 2021 From: xaled at web.de (xaled at web.de) Date: Sun, 11 Apr 2021 23:59:25 +0200 Subject: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example Message-ID: <020701d72f1d$f0055f80$d0101e80$@web.de> Hi, I tested the 3.2 Prepaid example from B2B tutorial https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2 In Scenario Schema there is a first reinvite from OpenSIPS B2B to caller - this reINVITE in my tests comes without SDP. Should not there be an SDP, or am I missing something in this scenario? INVITE sip:TEST001 at 1.2.3.201:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 1.2.5.183:5060;branch=z9hG4bKb5b7.1ffbf7b1.0 To: ;tag=f79aacce From: ;tag=B2B.81.186.1618177460 CSeq: 3 INVITE Call-ID: 80a9a9cfbe9920f8 at 1.2.3.201 Max-Forwards: 70 Content-Length: 0 Contact: Thanks, Xaled From homstr at gmail.com Sun Apr 11 23:06:30 2021 From: homstr at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdC10Lkg0KXQsNC80YHRgtGA?=) Date: Mon, 12 Apr 2021 01:06:30 +0200 Subject: [OpenSIPS-Users] Working with MAX-FORWARDS in B2B topology hidding Message-ID: Hello. Please tell me who faced the setting of the MAX-FORWARDS parameter pass when using the B2B module We have found that when using the module, this counter starts counting again from the initial value of 70, which leads to looping of calls in the network. example from trace in the input of opensips we have INVITE sip:543174317 at 94.241.67.137 SIP/2.0 *Max-Forwards: 18* Via: SIP/2.0/UDP 94.241.67.136:5062;rport;branch=z9hG4bK759843122 From: ;tag=298433023 To: Call-ID: 752394752 at 94.241.67.136:5062 CSeq: 3633945 INVITE User-Agent: YATE/6.2.1 Contact: Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, PRACK, INFO Supported: 100rel Content-Type: multipart/mixed;boundary=734768646_1362544590 Content-Length: 443 but on the output there is: INVITE sip:543174317 at 94.241.67.137 SIP/2.0 Via: SIP/2.0/UDP 94.241.67.137:5060;branch=z9hG4bK5e7e.38ff3a06.0 To: sip:543174317 at 94.241.67.137 From: ;tag=153c35a5fde4948908525582635dfe04 CSeq: 3633946 INVITE Call-ID: B2B.506.2825450.1617906940 *Max-Forwards: 70* Content-Length: 185 User-Agent: OpenSIPS (2.4.6 (x86_64/linux)) Content-Type: application/sdp Supported: 100rel Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, PRACK, INFO Contact: The module documentation says that this header should be passed from the dialog of one side to the other side. Max-Forwards (it is decreased by 1) Thanks in advance, with regards, Alexey Homstr -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Apr 12 07:08:47 2021 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 12 Apr 2021 10:08:47 +0300 Subject: [OpenSIPS-Users] Working with MAX-FORWARDS in B2B topology hidding In-Reply-To: References: Message-ID: Hi, Alexey! Unfortunately the behavior is mis-documented - the Max-Forwards seems to always be enforced to 70, regardless what's on the a-leg. Please open a bug report on our tracking list [1] to keep an eye on this. [1] https://github.com/OpenSIPS/opensips/issues Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 4/12/21 2:06 AM, Алексей Хамстр wrote: > Hello. > Please tell me who faced the setting of the MAX-FORWARDS parameter pass > when using the B2B module > We have found that when using the module, this counter starts counting > again from the initial value of 70, which leads to looping of calls in > the network. > example from trace > > in the input of opensips we have > > INVITE sip:543174317 at 94.241.67.137 > SIP/2.0 > *Max-Forwards: 18* > Via: SIP/2.0/UDP 94.241.67.136:5062;rport;branch=z9hG4bK759843122 > From: >;tag=298433023 > To: > > Call-ID: 752394752 at 94.241.67.136:5062 > CSeq: 3633945 INVITE > User-Agent: YATE/6.2.1 > Contact: > > Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, PRACK, INFO > Supported: 100rel > Content-Type: multipart/mixed;boundary=734768646_1362544590 > Content-Length: 443 > > but on the output there is: > > INVITE sip:543174317 at 94.241.67.137 > SIP/2.0 > Via: SIP/2.0/UDP 94.241.67.137:5060;branch=z9hG4bK5e7e.38ff3a06.0 > To: sip:543174317 at 94.241.67.137 > From: >;tag=153c35a5fde4948908525582635dfe04 > CSeq: 3633946 INVITE > Call-ID: B2B.506.2825450.1617906940 > *Max-Forwards: 70* > Content-Length: 185 > User-Agent: OpenSIPS (2.4.6 (x86_64/linux)) > Content-Type: application/sdp > Supported: 100rel > Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, PRACK, INFO > Contact: > > > The module documentation says that this header should be passed from the > dialog of one side to the other side. > Max-Forwards (it is decreased by 1) > > Thanks in advance, > with regards, > Alexey Homstr > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From xaled at web.de Tue Apr 13 09:52:13 2021 From: xaled at web.de (xaled at web.de) Date: Tue, 13 Apr 2021 11:52:13 +0200 Subject: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module In-Reply-To: <01f901d72f1a$b6fd55b0$24f80110$@web.de> References: <019b01d72f01$0a957a00$1fc06e00$@web.de> <01f901d72f1a$b6fd55b0$24f80110$@web.de> Message-ID: <04d501d7304a$ae58fe60$0b0afb20$@web.de> Hi, Could somebody provide any advice on the issue? Thanks, Xaled -----Original Message----- From: Users On Behalf Of xaled at web.de Sent: Sunday, April 11, 2021 11:36 PM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module Hi, I tried to use the extra_hdrs variable of b2b_client_new to set up the Route header extra_hdrs (var, optional) - AVP variable holding a list of extra headers (the header names) to be added for any request sent to this entity. But was confused by the "header names" part and it looks like that it will not accept headers with values. What is then the use of the variable? xlog("INFO: B2B $b2b_logic.ctx(next_route)"); INFO: B2B Route: $avp(route_header) = $b2b_logic.ctx(next_route); b2b_client_new("media", "sip:+87654321 at 1.1.1.1;user=phone", "sip:$b2b_logic.ctx(server_ip)", ,$avp(route_header)); ERROR:b2b_logic:b2bl_entity_new: header names without values! The need to have an avp for extra_hdrs is an unnecessary limitation, or is there a specific reason for it? Thanks, Xaled -----Original Message----- From: Users On Behalf Of xaled at web.de Sent: Sunday, April 11, 2021 8:33 PM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module Hi, would it be possible to respect loose routing on the caller side of B2B module and set route header on the callee side accordingly Something like this: Incoming INVITE: INVITE sip:+12345678 at 1.2.3.4;user=phone SIP/2.0 Via: SIP/2.0/TCP 4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb Route: , Wanted outgoing INVITE: INVITE sip:+87654321 at 1.1.1.1;user=phone SIP/2.0 Via: SIP/2.0/TCP test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb Route: I tried setting loose_route in both the original route and in the resulting b2b route but without success. route[b2b_test_logic] { <------>loose_route(); if ($rm != "BYE") { # for requests other than BYE, no special actions needs to be done, # just pass the request to the peer b2b_pass_request(); exit; } ... Thanks, Xaled _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From saurabhc at 3clogic.com Wed Apr 14 10:03:25 2021 From: saurabhc at 3clogic.com (Saurabh Chopra) Date: Wed, 14 Apr 2021 15:33:25 +0530 Subject: [OpenSIPS-Users] Issue with opensips-cli . In-Reply-To: References: <4f177b0b-8ad8-2400-a1f8-9d726ab1dc9e@opensips.org> <91629096-8de4-9a6e-8458-8855d36a629f@opensips.org> Message-ID: Hi Opensips Team/ Razvan, As per Sasmita's previous mail, we are still stuck in a situation where, 1. If we are using "FIFO" as a communication_type we are not getting data from a remote machine instead getting data from a local host. 2. If we are using "HTTP" as a communication_type then we are unable to switch to a remote machine. For both scenarios logs have been attached in *pastebin*. Kindly advise and provide us with an early resolution. We will be waiting for your prompt response Best Regards Saurabh Chopra +918861979979 On Tue, Mar 30, 2021 at 11:31 AM Sasmita Panda wrote: > Hi , > > I have changed my instance.cfg like bellow . > > [10.0.0.1] > prompt_name: opensips-cli at 10.0.0.1 > communication_type: http > url: http://10.0.0.1:8888/json > > With this config above I am getting some error . I am not able to switch > to the remote instance as well . > > https://pastebin.com/1fDuHbBk > > > if I am giving below config . > > [10.0.0.1] > prompt_name: opensips-cli at 10.0.0.1 > communication_type: fifo > fifo_file: /tmp/opensips_fifo > > I am able to switch to the remote instance . After switching when I am > running any command its giving the result for localhost not for the remote > instance ;. > > > What should I do ? Am I missing any dependency library ? Please do help . > > > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > On Fri, Mar 26, 2021 at 5:58 PM Sasmita Panda wrote: > >> same result . >> >> my instance.cfg contains : >> cat instance.cfg >> [10.0.1.1] >> prompt_name: opensips-cli at 10.0.1.1 >> fifo_file: /tmp/opensips_fifo >> >> https://pastebin.com/FQBipyZJ >> This is the result . >> >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> On Fri, Mar 26, 2021 at 5:38 PM Răzvan Crainea >> wrote: >> >>> You are not switching the instance correctly: should have used `instance >>> switch 10.0.1.1` >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Core Developer >>> http://www.opensips-solutions.com >>> >>> On 3/26/21 1:54 PM, Sasmita Panda wrote: >>> > I am not getting any error for this . I have pasted the logs in >>> pastebin . >>> > >>> > https://pastebin.com/4tzXzrrG >>> > >>> > What needs to be done in the remote machine 10.0.1.1 . I have allowed >>> > 8888 tcp port . I saw default http url is like http://10.0.1.1:8888/mi >>> > . >>> > Do suggest if I am missing anything . >>> > >>> > >>> > */Thanks & Regards/* >>> > /Sasmita Panda/ >>> > /Senior Network Testing and Software Engineer/ >>> > /3CLogic , ph:07827611765/ >>> > >>> > >>> > On Fri, Mar 26, 2021 at 5:00 PM Răzvan Crainea >> > > wrote: >>> > >>> > Hi, Sasmita! >>> > >>> > Can you post the logs of an execution for the 10.0.1.1 instance run >>> > with >>> > debugging (-d parameter)? Also, please post them on pastebin[1] or >>> in a >>> > different place, not in the mail itself. >>> > >>> > [1] https://pastebin.com/ >>> > >>> > Best regards, >>> > >>> > Răzvan Crainea >>> > OpenSIPS Core Developer >>> > http://www.opensips-solutions.com < >>> http://www.opensips-solutions.com> >>> > >>> > On 3/26/21 11:16 AM, Sasmita Panda wrote: >>> > > Hi All , >>> > > >>> > > I am using opensips-cli ( 0.1.0 ) and opensips-3.1 . >>> > > >>> > > I am able to get all the data of the local host through cli but >>> > when I >>> > > am connecting to a remote host even though it's getting >>> connected >>> > but no >>> > > data is visible for the remote instance . It only showing the >>> > data of >>> > > the instance on which I am running the cli . >>> > > >>> > > >>> > > my opensips-cli.cfg >>> > > *[default] >>> > > database_modules: All >>> > > prompt_intro: Welcome to OpenSIPS >>> > > >>> > > database_admin_url: mysql://root:cccl0g1c at localhost >>> > > #database_url: mysql://opensips:opensipsrw at localhost >>> > > database_schema_path: /opt/opensips-3.1/scripts >>> > > >>> > > [10.0.1.1] >>> > > prompt_name: opensips-cli at 10.0.1.1 >> opensips-cli at 10.0.1.1> >>> > > >>> > > prompt_intro: OpenSIPS CLI for 10.0.1.1 instance >>> > > communication_type: fifo >>> > > url: http://10.0.1.1:8888/mi >>> > >* >>> > > >>> > > Am I doing anything wrong ? Please do help me . >>> > > >>> > > >>> > > */Thanks & Regards/* >>> > > /Sasmita Panda/ >>> > > /Senior Network Testing and Software Engineer/ >>> > > /3CLogic , ph:07827611765/ >>> > > >>> > > _______________________________________________ >>> > > Users mailing list >>> > > Users at lists.opensips.org >>> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> > >>> > > >>> > >>> > _______________________________________________ >>> > Users mailing list >>> > Users at lists.opensips.org >>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> > >>> > >>> > >>> > _______________________________________________ >>> > Users mailing list >>> > Users at lists.opensips.org >>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> > >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 14 10:30:17 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Apr 2021 13:30:17 +0300 Subject: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module In-Reply-To: <019b01d72f01$0a957a00$1fc06e00$@web.de> References: <019b01d72f01$0a957a00$1fc06e00$@web.de> Message-ID: Hi Xaled, The record routing is incompatible with the b2b, by SIP definition. The record routing is done only by a SIP proxy, not by a SIP b2b2. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/11/21 9:32 PM, xaled at web.de wrote: > Hi, > > would it be possible to respect loose routing on the caller side of B2B module and set route header on the callee side accordingly > > Something like this: > > Incoming INVITE: > > INVITE sip:+12345678 at 1.2.3.4;user=phone SIP/2.0 > Via: SIP/2.0/TCP 4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb > Route: , > > Wanted outgoing INVITE: > > INVITE sip:+87654321 at 1.1.1.1;user=phone SIP/2.0 > Via: SIP/2.0/TCP test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb > Route: > > I tried setting loose_route in both the original route and in the resulting b2b route but without success. > > route[b2b_test_logic] { > > <------>loose_route(); > > if ($rm != "BYE") { > # for requests other than BYE, no special actions needs to be done, > # just pass the request to the peer > b2b_pass_request(); > exit; > } > ... > > Thanks, > Xaled > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From callum.guy at x-on.co.uk Wed Apr 14 10:44:17 2021 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 14 Apr 2021 11:44:17 +0100 Subject: [OpenSIPS-Users] TLS certificate reloading Message-ID: Hi All, I recently encountered an issue where our certificates were renewed, following which I issued: *opensips-cli -x mi tls_reload* The CLI action indicated success however on closer inspection of the handshake we could see the previous certificate was continuing to be presented. Previously I have had success with the reload operation. In this situation only a full restart resolved the issue and loaded the updated certificates. We are storing certificates in the database and only use domain default with a type 1 and type 2 record, both using the same certificate. The following global params are set: tcp_connection_lifetime=720 tcp_connect_timeout=200 We also use extremely long module timeout settings to deal with an outrageously slow peer (slow on first connect): modparam("tls_mgm", "tls_send_timeout", 2000) modparam("tls_mgm", "tls_handshake_timeout", 2000) Is tls_reload expected to work under all conditions or is there something else we need to do (i.e. tear down existing connections)? All log messages indicated success and as we are using lets encrypt certs the subject/issuer remained the same so only a packet capture revealed the actual serial number of the cert - as an aside it would be useful to have the SN reported in the reload operation. Thanks, Callum -- *0333 332 0000  |  x-on.co.uk   |   **      **  |  Coronavirus **  |   Practice Index Reviews * THE ITSPA AWARDS 2020 AND Best ITSP - Mid Market, Best Software and Best Vertical Solution are trade marks of the Internet Telephony Services Providers' Association, used under licence. *Our new office address: 22 Riduna Park, Melton IP12 1QT.* X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexei.vasilyev at gmail.com Wed Apr 14 10:50:13 2021 From: alexei.vasilyev at gmail.com (Alexey Vasilyev) Date: Wed, 14 Apr 2021 03:50:13 -0700 (MST) Subject: [OpenSIPS-Users] TLS certificate reloading In-Reply-To: References: Message-ID: <1618397413667-0.post@n2.nabble.com> Hi, yes there is an issue. Please follow here: https://github.com/OpenSIPS/opensips/issues/2433 ----- --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From bogdan at opensips.org Wed Apr 14 10:53:45 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Apr 2021 13:53:45 +0300 Subject: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications In-Reply-To: References: Message-ID: <48bbe2f1-0131-03e1-69ca-34fd872d066f@opensips.org> See https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/ https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/9/21 12:39 PM, HS wrote: > Hello all. > > We are using Opensips 3.0 (with rtpproxy) and been looking to > implement Push Notifications for iOS devices. Did come across quite a > few resources, but they seem either too old ~2013 time or about V3.1. > Has anyone implemented something to make calls to devices with the UAC > in background mode on V3.0 pls? > > Or suggestions on how to make this work on v3.0? > http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/ > > Appreciate the help. > > Thanks. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Wed Apr 14 10:58:50 2021 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 14 Apr 2021 11:58:50 +0100 Subject: [OpenSIPS-Users] TLS certificate reloading In-Reply-To: <1618397413667-0.post@n2.nabble.com> References: <1618397413667-0.post@n2.nabble.com> Message-ID: Excellent, thanks for beginning this to my attention - will track. On Wed, 14 Apr 2021 at 11:52, Alexey Vasilyev wrote: > Hi, yes there is an issue. > > Please follow here: https://github.com/OpenSIPS/opensips/issues/2433 > > > > > ----- > --- > Alexey Vasilyev > -- > Sent from: > http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- *0333 332 0000  |  x-on.co.uk   |   **      **  |  Coronavirus **  |   Practice Index Reviews * THE ITSPA AWARDS 2020 AND Best ITSP - Mid Market, Best Software and Best Vertical Solution are trade marks of the Internet Telephony Services Providers' Association, used under licence. *Our new office address: 22 Riduna Park, Melton IP12 1QT.* X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 14 11:03:36 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Apr 2021 14:03:36 +0300 Subject: [OpenSIPS-Users] The next OpenSIPS Bootcamp online training In-Reply-To: References: Message-ID: <117695bb-9372-b25e-eabf-eb8e27bfba2f@opensips.org> Heads-up ALL, the deadline for the early-bird discount is almost here. IF you care about your pocket, register now ;-) https://opensips.org/training/OpenSIPS_eBootcamp_2021/#mu-pricing Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 3/31/21 2:30 PM, Bogdan-Andrei Iancu wrote: > > > > 17th May-28th May 2021, > > > online, worldwide > > > > *OpenSIPS Bootcamp* > > Join the core developers and founder of the OpenSIPS project for an > *OpenSIPS Bootcamp* > online session > - an in-cloud training, a ten days, 4 hours per day (40 hours) > intensive and practical training, covering installation, configuration > and administration on OpenSIPS. > > All the knowledge transferred to the students will be strongly backed > up by practice sessions where you will get hands-on experience in > handling OpenSIPS. The training is structured to be offer 50% / 50% > between the theoretical and practical sessions. > > Check Syllabus > > > *Early Birds open* > > The Early Bird 10% discount is available for registrations before > /*16th of April 2021*/, so do not mis the opportunity. The number of > seats is limited, so be sure and book a seat now. Keep in mind that a > 10% group discount is also available - grab your work mate and start > learning more OpenSIPS together . > . > > Register Now > > > *Certified training saves time and money* > > OpenSIPS mistakes are easily avoided if you get proper training! > Companies that use OpenSIPS waste time and money when they don't have > a trained engineer on staff. Searching on Google, waiting on IRC, even > the latency in mailing list replies takes it's toll over time. Take > this rare opportunity to train your employees with the project members > themselves. > > > Any questions? do not hesitate to contact us > ! > > ------------------------------------------------------------------------ > You received this email as part of your relationship with the OpenSIPS > Project. > If you do not want to receive any more news, please email to > unsubscribe . > > > -- > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Wed Apr 14 11:54:12 2021 From: xaled at web.de (xaled at web.de) Date: Wed, 14 Apr 2021 13:54:12 +0200 Subject: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module In-Reply-To: References: <019b01d72f01$0a957a00$1fc06e00$@web.de> Message-ID: <06ac01d73124$e2f3ef60$a8dbce20$@web.de> Hi Bogdan, I got it, the actual question is how can I add new headers (not only Route, but any header) with values to the outgoing B2B INVITE request. Please see my other mail. Thanks, Xaled -----Original Message----- From: Bogdan-Andrei Iancu Sent: Wednesday, April 14, 2021 12:30 PM To: xaled at web.de; users at lists.opensips.org Subject: Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module Hi Xaled, The record routing is incompatible with the b2b, by SIP definition. The record routing is done only by a SIP proxy, not by a SIP b2b2. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/11/21 9:32 PM, xaled at web.de wrote: > Hi, > > would it be possible to respect loose routing on the caller side of > B2B module and set route header on the callee side accordingly > > Something like this: > > Incoming INVITE: > > INVITE sip:+12345678 at 1.2.3.4;user=phone SIP/2.0 > Via: SIP/2.0/TCP > 4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb > Route: , > > > Wanted outgoing INVITE: > > INVITE sip:+87654321 at 1.1.1.1;user=phone SIP/2.0 > Via: SIP/2.0/TCP > test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb > Route: > > I tried setting loose_route in both the original route and in the resulting b2b route but without success. > > route[b2b_test_logic] { > > <------>loose_route(); > > if ($rm != "BYE") { > # for requests other than BYE, no special actions needs to be done, > # just pass the request to the peer > b2b_pass_request(); > exit; > } > ... > > Thanks, > Xaled > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Wed Apr 14 12:05:01 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Apr 2021 15:05:01 +0300 Subject: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module In-Reply-To: <06ac01d73124$e2f3ef60$a8dbce20$@web.de> References: <019b01d72f01$0a957a00$1fc06e00$@web.de> <06ac01d73124$e2f3ef60$a8dbce20$@web.de> Message-ID: <8d7442e2-f77c-05dc-b8f7-6f1126b65d07@opensips.org> Use the "local_route" for that https://www.opensips.org/Documentation/Script-Routes-3-1#toc6 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/14/21 2:54 PM, xaled at web.de wrote: > Hi Bogdan, > > I got it, the actual question is how can I add new headers (not only Route, but any header) with values to the outgoing B2B INVITE request. Please see my other mail. > > Thanks, > Xaled > > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: Wednesday, April 14, 2021 12:30 PM > To: xaled at web.de; users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module > > Hi Xaled, > > The record routing is incompatible with the b2b, by SIP definition. The record routing is done only by a SIP proxy, not by a SIP b2b2. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/11/21 9:32 PM, xaled at web.de wrote: >> Hi, >> >> would it be possible to respect loose routing on the caller side of >> B2B module and set route header on the callee side accordingly >> >> Something like this: >> >> Incoming INVITE: >> >> INVITE sip:+12345678 at 1.2.3.4;user=phone SIP/2.0 >> Via: SIP/2.0/TCP >> 4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb >> Route: , >> >> >> Wanted outgoing INVITE: >> >> INVITE sip:+87654321 at 1.1.1.1;user=phone SIP/2.0 >> Via: SIP/2.0/TCP >> test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb >> Route: >> >> I tried setting loose_route in both the original route and in the resulting b2b route but without success. >> >> route[b2b_test_logic] { >> >> <------>loose_route(); >> >> if ($rm != "BYE") { >> # for requests other than BYE, no special actions needs to be done, >> # just pass the request to the peer >> b2b_pass_request(); >> exit; >> } >> ... >> >> Thanks, >> Xaled >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From xaled at web.de Wed Apr 14 12:14:55 2021 From: xaled at web.de (xaled at web.de) Date: Wed, 14 Apr 2021 14:14:55 +0200 Subject: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module References: <019b01d72f01$0a957a00$1fc06e00$@web.de> Message-ID: <06bb01d73127$c7fef760$57fce620$@web.de> Hi Bogdan, I understood and got it working with extra_hdrs and extra_hdr_bodies. Thank you, Xaled -----Original Message----- From: xaled at web.de Sent: Wednesday, April 14, 2021 1:54 PM To: 'Bogdan-Andrei Iancu' ; 'users at lists.opensips.org' Subject: RE: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module Hi Bogdan, I got it, the actual question is how can I add new headers (not only Route, but any header) with values to the outgoing B2B INVITE request. Please see my other mail. Thanks, Xaled -----Original Message----- From: Bogdan-Andrei Iancu Sent: Wednesday, April 14, 2021 12:30 PM To: xaled at web.de; users at lists.opensips.org Subject: Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module Hi Xaled, The record routing is incompatible with the b2b, by SIP definition. The record routing is done only by a SIP proxy, not by a SIP b2b2. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/11/21 9:32 PM, xaled at web.de wrote: > Hi, > > would it be possible to respect loose routing on the caller side of > B2B module and set route header on the callee side accordingly > > Something like this: > > Incoming INVITE: > > INVITE sip:+12345678 at 1.2.3.4;user=phone SIP/2.0 > Via: SIP/2.0/TCP > 4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb > Route: , > > > Wanted outgoing INVITE: > > INVITE sip:+87654321 at 1.1.1.1;user=phone SIP/2.0 > Via: SIP/2.0/TCP > test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb > Route: > > I tried setting loose_route in both the original route and in the resulting b2b route but without success. > > route[b2b_test_logic] { > > <------>loose_route(); > > if ($rm != "BYE") { > # for requests other than BYE, no special actions needs to be done, > # just pass the request to the peer > b2b_pass_request(); > exit; > } > ... > > Thanks, > Xaled > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From xaled at web.de Wed Apr 14 12:17:46 2021 From: xaled at web.de (xaled at web.de) Date: Wed, 14 Apr 2021 14:17:46 +0200 Subject: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example In-Reply-To: <020701d72f1d$f0055f80$d0101e80$@web.de> References: <020701d72f1d$f0055f80$d0101e80$@web.de> Message-ID: <06bd01d73128$2db357e0$891a07a0$@web.de> Hi, could somebody please shed some light on this scenario? Thanks, Xaled -----Original Message----- From: Users On Behalf Of xaled at web.de Sent: Sunday, April 11, 2021 11:59 PM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example Hi, I tested the 3.2 Prepaid example from B2B tutorial https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2 In Scenario Schema there is a first reinvite from OpenSIPS B2B to caller - this reINVITE in my tests comes without SDP. Should not there be an SDP, or am I missing something in this scenario? INVITE sip:TEST001 at 1.2.3.201:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 1.2.5.183:5060;branch=z9hG4bKb5b7.1ffbf7b1.0 To: ;tag=f79aacce From: ;tag=B2B.81.186.1618177460 CSeq: 3 INVITE Call-ID: 80a9a9cfbe9920f8 at 1.2.3.201 Max-Forwards: 70 Content-Length: 0 Contact: Thanks, Xaled _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Wed Apr 14 13:24:49 2021 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 14 Apr 2021 16:24:49 +0300 Subject: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example In-Reply-To: <06bd01d73128$2db357e0$891a07a0$@web.de> References: <020701d72f1d$f0055f80$d0101e80$@web.de> <06bd01d73128$2db357e0$891a07a0$@web.de> Message-ID: Hi, Xaled! No, there should bot be an SDP, because you don't have an SDP (the media server is gone, and you don't know the callee's SDP). Instead, the module is doing a late negotiation to find out advertised caller's SDP, and present it to the new callee. Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 4/14/21 3:17 PM, xaled at web.de wrote: > Hi, > > could somebody please shed some light on this scenario? > > Thanks, > Xaled > > -----Original Message----- > From: Users On Behalf Of xaled at web.de > Sent: Sunday, April 11, 2021 11:59 PM > To: 'OpenSIPS users mailling list' > Subject: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example > > Hi, > > I tested the 3.2 Prepaid example from B2B tutorial > > https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2 > > In Scenario Schema there is a first reinvite from OpenSIPS B2B to caller - this reINVITE in my tests comes without SDP. Should not there be an SDP, or am I missing something in this scenario? > > INVITE sip:TEST001 at 1.2.3.201:5060;transport=tcp SIP/2.0 > Via: SIP/2.0/TCP 1.2.5.183:5060;branch=z9hG4bKb5b7.1ffbf7b1.0 > To: ;tag=f79aacce > From: ;tag=B2B.81.186.1618177460 > CSeq: 3 INVITE > Call-ID: 80a9a9cfbe9920f8 at 1.2.3.201 > Max-Forwards: 70 > Content-Length: 0 > Contact: > > Thanks, > Xaled > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From bullehs at gmail.com Wed Apr 14 16:33:38 2021 From: bullehs at gmail.com (HS) Date: Wed, 14 Apr 2021 21:33:38 +0500 Subject: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications In-Reply-To: <48bbe2f1-0131-03e1-69ca-34fd872d066f@opensips.org> References: <48bbe2f1-0131-03e1-69ca-34fd872d066f@opensips.org> Message-ID: Bogdan-Andrei, Thanks a lot for sharing these links. I had actually looked at them previously, all are for Opensips 3.1 - any possibility of getting PN to work with Openips 3.0 please? Thanks. On Wed, Apr 14, 2021 at 3:53 PM Bogdan-Andrei Iancu wrote: > See > > > https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/ > > > https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ > > > https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/9/21 12:39 PM, HS wrote: > > Hello all. > > We are using Opensips 3.0 (with rtpproxy) and been looking to implement > Push Notifications for iOS devices. Did come across quite a few resources, > but they seem either too old ~2013 time or about V3.1. Has anyone > implemented something to make calls to devices with the UAC in background > mode on V3.0 pls? > > Or suggestions on how to make this work on v3.0? > > http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/ > > Appreciate the help. > > Thanks. > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Wed Apr 14 16:44:04 2021 From: xaled at web.de (xaled at web.de) Date: Wed, 14 Apr 2021 18:44:04 +0200 Subject: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example In-Reply-To: References: <020701d72f1d$f0055f80$d0101e80$@web.de> <06bd01d73128$2db357e0$891a07a0$@web.de> Message-ID: <072701d7314d$615ee210$241ca630$@web.de> Hi Răzvan, Thank you for the reply. Indeed I was able to get the prepaid B2B scenario running with all the needed headers once I understood the mechanics of B2B module. Thank you and the team for the great module that will allow me to implement some very interesting scenarios! Greetings, Xaled -----Original Message----- From: Users On Behalf Of Razvan Crainea Sent: Wednesday, April 14, 2021 3:25 PM To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example Hi, Xaled! No, there should bot be an SDP, because you don't have an SDP (the media server is gone, and you don't know the callee's SDP). Instead, the module is doing a late negotiation to find out advertised caller's SDP, and present it to the new callee. Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 4/14/21 3:17 PM, xaled at web.de wrote: > Hi, > > could somebody please shed some light on this scenario? > > Thanks, > Xaled > > -----Original Message----- > From: Users On Behalf Of > xaled at web.de > Sent: Sunday, April 11, 2021 11:59 PM > To: 'OpenSIPS users mailling list' > Subject: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example > > Hi, > > I tested the 3.2 Prepaid example from B2B tutorial > > https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2 > > In Scenario Schema there is a first reinvite from OpenSIPS B2B to caller - this reINVITE in my tests comes without SDP. Should not there be an SDP, or am I missing something in this scenario? > > INVITE sip:TEST001 at 1.2.3.201:5060;transport=tcp SIP/2.0 > Via: SIP/2.0/TCP 1.2.5.183:5060;branch=z9hG4bKb5b7.1ffbf7b1.0 > To: ;tag=f79aacce > From: > ;tag=B2B.81.186.1618177460 > CSeq: 3 INVITE > Call-ID: 80a9a9cfbe9920f8 at 1.2.3.201 > Max-Forwards: 70 > Content-Length: 0 > Contact: > > Thanks, > Xaled > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From callum.guy at x-on.co.uk Wed Apr 14 19:55:49 2021 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 14 Apr 2021 20:55:49 +0100 Subject: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications In-Reply-To: References: <48bbe2f1-0131-03e1-69ca-34fd872d066f@opensips.org> Message-ID: Thinking outside the box here but... maybe update to 3.1.1? https://www.opensips.org/Documentation/Migration-3-0-0-to-3-1-0 On Wed, 14 Apr 2021 at 17:35, HS wrote: > Bogdan-Andrei, > > Thanks a lot for sharing these links. I had actually looked at them > previously, all are for Opensips 3.1 - any possibility of getting PN to > work with Openips 3.0 please? > > Thanks. > > On Wed, Apr 14, 2021 at 3:53 PM Bogdan-Andrei Iancu > wrote: > >> See >> >> >> https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/ >> >> >> https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ >> >> >> https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp 2021 online >> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >> >> On 4/9/21 12:39 PM, HS wrote: >> >> Hello all. >> >> We are using Opensips 3.0 (with rtpproxy) and been looking to implement >> Push Notifications for iOS devices. Did come across quite a few resources, >> but they seem either too old ~2013 time or about V3.1. Has anyone >> implemented something to make calls to devices with the UAC in background >> mode on V3.0 pls? >> >> Or suggestions on how to make this work on v3.0? >> >> http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/ >> >> Appreciate the help. >> >> Thanks. >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- *0333 332 0000  |  x-on.co.uk   |   **      **  |  Coronavirus **  |   Practice Index Reviews * THE ITSPA AWARDS 2020 AND Best ITSP - Mid Market, Best Software and Best Vertical Solution are trade marks of the Internet Telephony Services Providers' Association, used under licence. *Our new office address: 22 Riduna Park, Melton IP12 1QT.* X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 15 06:00:57 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 15 Apr 2021 09:00:57 +0300 Subject: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications In-Reply-To: References: <48bbe2f1-0131-03e1-69ca-34fd872d066f@opensips.org> Message-ID: The last link is for OpenSIPS 2.3 and higher. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/14/21 7:33 PM, HS wrote: > Bogdan-Andrei, > > Thanks a lot for sharing these links. I had actually looked at them > previously, all are for Opensips 3.1 - any possibility of getting PN > to work with Openips 3.0 please? > > Thanks. > > On Wed, Apr 14, 2021 at 3:53 PM Bogdan-Andrei Iancu > > wrote: > > See > > https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/ > > https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ > > https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/9/21 12:39 PM, HS wrote: >> Hello all. >> >> We are using Opensips 3.0 (with rtpproxy) and been looking to >> implement Push Notifications for iOS devices. Did come across >> quite a few resources, but they seem either too old ~2013 time or >> about V3.1. Has anyone implemented something to make calls to >> devices with the UAC in background mode on V3.0 pls? >> >> Or suggestions on how to make this work on v3.0? >> http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/ >> >> Appreciate the help. >> >> Thanks. >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 15 06:04:24 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 15 Apr 2021 09:04:24 +0300 Subject: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications In-Reply-To: References: <48bbe2f1-0131-03e1-69ca-34fd872d066f@opensips.org> Message-ID: All 3.1.x are the alike when comes to usage. The difference between the minor versions (on the same branch/master version) is about bug fixes. So you can use that link to migrate to any 3.1.x Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/14/21 10:55 PM, Callum Guy wrote: > Thinking outside the box here but... maybe update to 3.1.1? > > https://www.opensips.org/Documentation/Migration-3-0-0-to-3-1-0 > > On Wed, 14 Apr 2021 at 17:35, HS > wrote: > > Bogdan-Andrei, > > Thanks a lot for sharing these links. I had actually looked at > them previously, all are for Opensips 3.1 - any possibility of > getting PN to work with Openips 3.0 please? > > Thanks. > > On Wed, Apr 14, 2021 at 3:53 PM Bogdan-Andrei Iancu > > wrote: > > See > > https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/ > > https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ > > https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/9/21 12:39 PM, HS wrote: >> Hello all. >> >> We are using Opensips 3.0 (with rtpproxy) and been looking to >> implement Push Notifications for iOS devices. Did come across >> quite a few resources, but they seem either too old ~2013 >> time or about V3.1. Has anyone implemented something to make >> calls to devices with the UAC in background mode on V3.0 pls? >> >> Or suggestions on how to make this work on v3.0? >> http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/ >> >> Appreciate the help. >> >> Thanks. >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > *^0333 332 0000  | x-on.co.uk   | > _**_^ > **^  | > Coronavirus > **^  > | Practice Index Reviews * > > THE ITSPA AWARDS 2020 AND Best > ITSP - Mid Market, Best Software and Best Vertical Solution are trade > marks of the Internet Telephony Services Providers' Association, used > under licence. > > *Our new office address: 22 Riduna Park, Melton IP12 1QT.* > > X-on is a trading name of Storacall Technology Ltd a limited company > registered in England and Wales. > Registered Office : Avaland House, 110 London Road, Apsley, Hemel > Hempstead, Herts, HP3 9SD. 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URL: From bogdan at opensips.org Thu Apr 15 09:37:39 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 15 Apr 2021 12:37:39 +0300 Subject: [OpenSIPS-Users] [blogpost] Tips & Tricks for Advanced SIP Capturing Message-ID: <20da1ac9-37f5-4613-1fb3-41fcec63842f@opensips.org> Hi OpenSIPS'ers! With the new upcoming OpenSIPS 3.2 release, we are looking to address more challenges related to SIP capturing / tracing: multi-call correlation, SIP server chaining and B2B-wise tracing. https://blog.opensips.org/2021/04/15/tips-tricks-for-advanced-sip-capturing/ Enjoy, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ From saurabhc at 3clogic.com Thu Apr 15 09:51:33 2021 From: saurabhc at 3clogic.com (Saurabh Chopra) Date: Thu, 15 Apr 2021 15:21:33 +0530 Subject: [OpenSIPS-Users] Issue with opensips-cli . In-Reply-To: References: <4f177b0b-8ad8-2400-a1f8-9d726ab1dc9e@opensips.org> <91629096-8de4-9a6e-8458-8855d36a629f@opensips.org> Message-ID: Hi Guys, Any update would be appreciated regarding the above matter. Best Regards Saurabh Chopra +918861979979 On Wed, Apr 14, 2021 at 3:33 PM Saurabh Chopra wrote: > Hi Opensips Team/ Razvan, > > As per Sasmita's previous mail, we are still stuck in a situation where, > > 1. If we are using "FIFO" as a communication_type we are not getting data > from a remote machine instead getting data from a local host. > 2. If we are using "HTTP" as a communication_type then we are unable to > switch to a remote machine. > > For both scenarios logs have been attached in *pastebin*. > > Kindly advise and provide us with an early resolution. We will be > waiting for your prompt response > > Best Regards > Saurabh Chopra > +918861979979 > > > On Tue, Mar 30, 2021 at 11:31 AM Sasmita Panda wrote: > >> Hi , >> >> I have changed my instance.cfg like bellow . >> >> [10.0.0.1] >> prompt_name: opensips-cli at 10.0.0.1 >> communication_type: http >> url: http://10.0.0.1:8888/json >> >> With this config above I am getting some error . I am not able to switch >> to the remote instance as well . >> >> https://pastebin.com/1fDuHbBk >> >> >> if I am giving below config . >> >> [10.0.0.1] >> prompt_name: opensips-cli at 10.0.0.1 >> communication_type: fifo >> fifo_file: /tmp/opensips_fifo >> >> I am able to switch to the remote instance . After switching when I am >> running any command its giving the result for localhost not for the remote >> instance ;. >> >> >> What should I do ? Am I missing any dependency library ? Please do help . >> >> >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> On Fri, Mar 26, 2021 at 5:58 PM Sasmita Panda wrote: >> >>> same result . >>> >>> my instance.cfg contains : >>> cat instance.cfg >>> [10.0.1.1] >>> prompt_name: opensips-cli at 10.0.1.1 >>> fifo_file: /tmp/opensips_fifo >>> >>> https://pastebin.com/FQBipyZJ >>> This is the result . >>> >>> >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Senior Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> >>> On Fri, Mar 26, 2021 at 5:38 PM Răzvan Crainea >>> wrote: >>> >>>> You are not switching the instance correctly: should have used >>>> `instance >>>> switch 10.0.1.1` >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Core Developer >>>> http://www.opensips-solutions.com >>>> >>>> On 3/26/21 1:54 PM, Sasmita Panda wrote: >>>> > I am not getting any error for this . I have pasted the logs in >>>> pastebin . >>>> > >>>> > https://pastebin.com/4tzXzrrG >>>> > >>>> > What needs to be done in the remote machine 10.0.1.1 . I have allowed >>>> > 8888 tcp port . I saw default http url is like >>>> http://10.0.1.1:8888/mi >>>> > . >>>> > Do suggest if I am missing anything . >>>> > >>>> > >>>> > */Thanks & Regards/* >>>> > /Sasmita Panda/ >>>> > /Senior Network Testing and Software Engineer/ >>>> > /3CLogic , ph:07827611765/ >>>> > >>>> > >>>> > On Fri, Mar 26, 2021 at 5:00 PM Răzvan Crainea >>> > > wrote: >>>> > >>>> > Hi, Sasmita! >>>> > >>>> > Can you post the logs of an execution for the 10.0.1.1 instance >>>> run >>>> > with >>>> > debugging (-d parameter)? Also, please post them on pastebin[1] >>>> or in a >>>> > different place, not in the mail itself. >>>> > >>>> > [1] https://pastebin.com/ >>>> > >>>> > Best regards, >>>> > >>>> > Răzvan Crainea >>>> > OpenSIPS Core Developer >>>> > http://www.opensips-solutions.com < >>>> http://www.opensips-solutions.com> >>>> > >>>> > On 3/26/21 11:16 AM, Sasmita Panda wrote: >>>> > > Hi All , >>>> > > >>>> > > I am using opensips-cli ( 0.1.0 ) and opensips-3.1 . >>>> > > >>>> > > I am able to get all the data of the local host through cli but >>>> > when I >>>> > > am connecting to a remote host even though it's getting >>>> connected >>>> > but no >>>> > > data is visible for the remote instance . It only showing the >>>> > data of >>>> > > the instance on which I am running the cli . >>>> > > >>>> > > >>>> > > my opensips-cli.cfg >>>> > > *[default] >>>> > > database_modules: All >>>> > > prompt_intro: Welcome to OpenSIPS >>>> > > >>>> > > database_admin_url: mysql://root:cccl0g1c at localhost >>>> > > #database_url: mysql://opensips:opensipsrw at localhost >>>> > > database_schema_path: /opt/opensips-3.1/scripts >>>> > > >>>> > > [10.0.1.1] >>>> > > prompt_name: opensips-cli at 10.0.1.1 >>> opensips-cli at 10.0.1.1> >>>> > > >>>> > > prompt_intro: OpenSIPS CLI for 10.0.1.1 instance >>>> > > communication_type: fifo >>>> > > url: http://10.0.1.1:8888/mi >>>> > >* >>>> > > >>>> > > Am I doing anything wrong ? Please do help me . >>>> > > >>>> > > >>>> > > */Thanks & Regards/* >>>> > > /Sasmita Panda/ >>>> > > /Senior Network Testing and Software Engineer/ >>>> > > /3CLogic , ph:07827611765/ >>>> > > >>>> > > _______________________________________________ >>>> > > Users mailing list >>>> > > Users at lists.opensips.org >>>> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> > >>>> > > >>>> > >>>> > _______________________________________________ >>>> > Users mailing list >>>> > Users at lists.opensips.org >>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> > >>>> > >>>> > >>>> > _______________________________________________ >>>> > Users mailing list >>>> > Users at lists.opensips.org >>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> > >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Apr 15 10:39:12 2021 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 15 Apr 2021 13:39:12 +0300 Subject: [OpenSIPS-Users] Issue with opensips-cli . In-Reply-To: References: <4f177b0b-8ad8-2400-a1f8-9d726ab1dc9e@opensips.org> <91629096-8de4-9a6e-8458-8855d36a629f@opensips.org> Message-ID: <138054fd-aee3-b2ad-0d53-3afbba9b28c1@opensips.org> Hi, Sasmita, Saurabh! 1. fifo, by its nature, can only be used for localhost queries; if you need to run remote commands, you need to use a remote communication protocol, such as http 2. Most likely you are querying a bad http root path for the `mi_http` module: in your URL it shows you are querying the `json` root, whereas the default mi_http root is `mi`. Can you make sure A. you're not modifying the mi_http root parameter[1], and/or if you do, make sure it corresponds to the path in the URL? [1] https://opensips.org/docs/modules/3.1.x/mi_http.html#param_root Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 4/15/21 12:51 PM, Saurabh Chopra wrote: > Hi Guys, > > Any update would be appreciated regarding the above matter. > > Best Regards > Saurabh Chopra > +918861979979 > > > On Wed, Apr 14, 2021 at 3:33 PM Saurabh Chopra > wrote: > > Hi Opensips Team/ Razvan, > > As per Sasmita's previous mail, we are still stuck in a situation where, > > 1. If we are using "FIFO" as a communication_type we are not getting > data from a remote machine instead getting data from a local host. > 2. If we are using "HTTP" as a communication_type then we are unable > to switch to a remote machine. > > For both scenarios logs have been attached in *pastebin*. > > Kindly advise and provide us with an early resolution. We will be > waiting for your prompt response > > Best Regards > Saurabh Chopra > +918861979979 > > > On Tue, Mar 30, 2021 at 11:31 AM Sasmita Panda > wrote: > > Hi , > > I have changed my instance.cfg like bellow . > > [10.0.0.1] > prompt_name: opensips-cli at 10.0.0.1 > communication_type: http > url: http://10.0.0.1:8888/json > > With this config above I am getting some error . I am not able > to switch to the remote instance as well . > > https://pastebin.com/1fDuHbBk > > > if I am giving below config . > >  [10.0.0.1] > prompt_name: opensips-cli at 10.0.0.1 > communication_type: fifo > fifo_file: /tmp/opensips_fifo > > I am able to switch to the remote instance . After switching > when I am running any command its giving the result for > localhost not  for the remote instance ;. > > > What should I do ? Am I missing any dependency library ? Please > do help . > > > > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Fri, Mar 26, 2021 at 5:58 PM Sasmita Panda > > wrote: > > same result . > > my instance.cfg contains : > cat instance.cfg > [10.0.1.1] > prompt_name: opensips-cli at 10.0.1.1 > > fifo_file: /tmp/opensips_fifo > > https://pastebin.com/FQBipyZJ > This is the result . > > */ > /* > */ > /* > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Fri, Mar 26, 2021 at 5:38 PM Răzvan Crainea > > wrote: > > You are not switching the instance correctly: should > have used `instance > switch 10.0.1.1` > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > > On 3/26/21 1:54 PM, Sasmita Panda wrote: > > I am not getting any error for this . I have pasted > the logs in pastebin . > > > > https://pastebin.com/4tzXzrrG > > > > > > > What needs to be done in the remote machine 10.0.1.1 > . I have allowed > > 8888 tcp port . I saw default http url is like > http://10.0.1.1:8888/mi > > > . > > Do suggest if I am missing anything . > > > > > > */Thanks & Regards/* > > /Sasmita Panda/ > > /Senior Network Testing and Software Engineer/ > > /3CLogic , ph:07827611765/ > > > > > > On Fri, Mar 26, 2021 at 5:00 PM Răzvan Crainea > > > >> wrote: > > > >     Hi, Sasmita! > > > >     Can you post the logs of an execution for the > 10.0.1.1 instance run > >     with > >     debugging (-d parameter)? Also, please post them > on pastebin[1] or in a > >     different place, not in the mail itself. > > > >     [1] https://pastebin.com/ > > > > > >     Best regards, > > > >     Răzvan Crainea > >     OpenSIPS Core Developer > > http://www.opensips-solutions.com > > > > > > >     On 3/26/21 11:16 AM, Sasmita Panda wrote: > >      > Hi All , > >      > > >      > I am using opensips-cli  ( 0.1.0 ) and > opensips-3.1 . > >      > > >      > I am able to get all the data of the local > host through cli but > >     when I > >      > am connecting to a remote host even though > it's getting connected > >     but no > >      > data is visible for the remote instance  . It > only showing the > >     data of > >      > the instance on which I am running the cli . > >      > > >      > > >      >   my opensips-cli.cfg > >      > *[default] > >      > database_modules: All > >      > prompt_intro: Welcome to OpenSIPS > >      > > >      > database_admin_url: > mysql://root:cccl0g1c at localhost > >      > #database_url: >  mysql://opensips:opensipsrw at localhost > >      > database_schema_path: /opt/opensips-3.1/scripts > >      > > >      > [10.0.1.1] > >      > prompt_name: opensips-cli at 10.0.1.1 > > > > >      > >> > >      > prompt_intro: OpenSIPS CLI for 10.0.1.1 instance > >      > communication_type: fifo > >      > url: http://10.0.1.1:8888/mi > > > >      >>* > >      > > >      > Am I doing anything wrong ? Please do help me . > >      > > >      > > >      > */Thanks & Regards/* > >      > /Sasmita Panda/ > >      > /Senior Network Testing and Software Engineer/ > >      > /3CLogic , ph:07827611765/ > >      > > >      > _______________________________________________ > >      > Users mailing list > >      > Users at lists.opensips.org > > > > >      > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >  > > >      > > > > >     _______________________________________________ > >     Users mailing list > > Users at lists.opensips.org > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >  > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From razvan at opensips.org Thu Apr 15 13:37:50 2021 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 15 Apr 2021 16:37:50 +0300 Subject: [OpenSIPS-Users] Issue with opensips-cli . In-Reply-To: References: <4f177b0b-8ad8-2400-a1f8-9d726ab1dc9e@opensips.org> <91629096-8de4-9a6e-8458-8855d36a629f@opensips.org> <138054fd-aee3-b2ad-0d53-3afbba9b28c1@opensips.org> Message-ID: <83ddb502-1f2f-8d16-d506-a8f5156db3bc@opensips.org> Can you make sure the remote opensips is actually listening on that IP and port? From the host you're running opensips-cli, can you run `nc -zv 10.x.x.x 8888`? Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 4/15/21 4:12 PM, Saurabh Chopra wrote: > Hi Razvan, > > Thank you for your response. > > I have changed the parameters but the problem still persists. Below is > the opensips-cli.cfg > > [10.x.x.x] > prompt_name: opensips-cli at 10.x.x.x > prompt_intro: OpenSIPS CLI for 10.x.x.x instance > communication_type: http > url: http://10.x.x.x:8888/mi > > Getting below errors when i tried to switch to instance 10.x.x.x > /(opensips-cli): instance switch 10.x.x.x > OpenSIPS CLI for 10.x.x.x instance > ERROR: communication exception for 'which' returned: > ERROR: Is OpenSIPS running?/ > Please advise what wrong I am doing now. > > > Best Regards > Saurabh Chopra > +918861979979 > > > On Thu, Apr 15, 2021 at 4:11 PM Răzvan Crainea > wrote: > > Hi, Sasmita, Saurabh! > > 1. fifo, by its nature, can only be used for localhost queries; if you > need to run remote commands, you need to use a remote communication > protocol, such as http > 2. Most likely you are querying a bad http root path for the `mi_http` > module: in your URL it shows you are querying the `json` root, whereas > the default mi_http root is `mi`. Can you make sure A. you're not > modifying the mi_http root parameter[1], and/or if you do, make sure it > corresponds to the path in the URL? > > [1] https://opensips.org/docs/modules/3.1.x/mi_http.html#param_root > > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 4/15/21 12:51 PM, Saurabh Chopra wrote: > > Hi Guys, > > > > Any update would be appreciated regarding the above matter. > > > > Best Regards > > Saurabh Chopra > > +918861979979 > > > > > > On Wed, Apr 14, 2021 at 3:33 PM Saurabh Chopra > > > >> wrote: > > > >     Hi Opensips Team/ Razvan, > > > >     As per Sasmita's previous mail, we are still stuck in a > situation where, > > > >     1. If we are using "FIFO" as a communication_type we are not > getting > >     data from a remote machine instead getting data from a > local host. > >     2. If we are using "HTTP" as a communication_type then we are > unable > >     to switch to a remote machine. > > > >     For both scenarios logs have been attached in *pastebin*. > > > >     Kindly advise and provide us with an early resolution. We will be > >     waiting for your prompt response > > > >     Best Regards > >     Saurabh Chopra > >     +918861979979 > > > > > >     On Tue, Mar 30, 2021 at 11:31 AM Sasmita Panda > > >     >> wrote: > > > >         Hi , > > > >         I have changed my instance.cfg like bellow . > > > >         [10.0.0.1] > >         prompt_name: opensips-cli at 10.0.0.1 > > > >         communication_type: http > >         url: http://10.0.0.1:8888/json > > > > > >         With this config above I am getting some error . I am not > able > >         to switch to the remote instance as well . > > > > https://pastebin.com/1fDuHbBk > > > > > > > >         if I am giving below config . > > > >           [10.0.0.1] > >         prompt_name: opensips-cli at 10.0.0.1 > > > >         communication_type: fifo > >         fifo_file: /tmp/opensips_fifo > > > >         I am able to switch to the remote instance . After switching > >         when I am running any command its giving the result for > >         localhost not  for the remote instance ;. > > > > > >         What should I do ? Am I missing any dependency library ? > Please > >         do help . > > > > > > > > > >         */Thanks & Regards/* > >         /Sasmita Panda/ > >         /Senior Network Testing and Software Engineer/ > >         /3CLogic , ph:07827611765/ > > > > > >         On Fri, Mar 26, 2021 at 5:58 PM Sasmita Panda > >          > >> wrote: > > > >             same result . > > > >             my instance.cfg contains : > >             cat instance.cfg > >             [10.0.1.1] > >             prompt_name: opensips-cli at 10.0.1.1 > > >              > > >             fifo_file: /tmp/opensips_fifo > > > > https://pastebin.com/FQBipyZJ > > > >             This is the result . > > > >             */ > >             /* > >             */ > >             /* > >             */Thanks & Regards/* > >             /Sasmita Panda/ > >             /Senior Network Testing and Software Engineer/ > >             /3CLogic , ph:07827611765/ > > > > > >             On Fri, Mar 26, 2021 at 5:38 PM Răzvan Crainea > >              > >> wrote: > > > >                 You are not switching the instance correctly: should > >                 have used `instance > >                 switch 10.0.1.1` > > > >                 Best regards, > > > >                 Răzvan Crainea > >                 OpenSIPS Core Developer > > http://www.opensips-solutions.com > >                  > > > > >                 On 3/26/21 1:54 PM, Sasmita Panda wrote: > >                  > I am not getting any error for this . I have > pasted > >                 the logs in pastebin . > >                  > > >                  > https://pastebin.com/4tzXzrrG > > >                  > > >                  > >                  >> > >                  > > >                  > What needs to be done in the remote machine > 10.0.1.1 > >                 . I have allowed > >                  > 8888 tcp port . I saw default http url is like > > http://10.0.1.1:8888/mi > > > >                  > >> . > >                  > Do suggest if I am missing anything . > >                  > > >                  > > >                  > */Thanks & Regards/* > >                  > /Sasmita Panda/ > >                  > /Senior Network Testing and Software Engineer/ > >                  > /3CLogic , ph:07827611765/ > >                  > > >                  > > >                  > On Fri, Mar 26, 2021 at 5:00 PM Răzvan Crainea > >                  > > > >                  > > >                  >>> wrote: > >                  > > >                  >     Hi, Sasmita! > >                  > > >                  >     Can you post the logs of an execution for the > >                 10.0.1.1 instance run > >                  >     with > >                  >     debugging (-d parameter)? Also, please > post them > >                 on pastebin[1] or in a > >                  >     different place, not in the mail itself. > >                  > > >                  >     [1] https://pastebin.com/ > > > >                  > >> > >                  > > >                  >     Best regards, > >                  > > >                  >     Răzvan Crainea > >                  >     OpenSIPS Core Developer > >                  > http://www.opensips-solutions.com > > >                  > > >                  > >                  >> > >                  > > >                  >     On 3/26/21 11:16 AM, Sasmita Panda wrote: > >                  >      > Hi All , > >                  >      > > >                  >      > I am using opensips-cli  ( 0.1.0 ) and > >                 opensips-3.1 . > >                  >      > > >                  >      > I am able to get all the data of the local > >                 host through cli but > >                  >     when I > >                  >      > am connecting to a remote host even though > >                 it's getting connected > >                  >     but no > >                  >      > data is visible for the remote > instance  . It > >                 only showing the > >                  >     data of > >                  >      > the instance on which I am running the > cli . > >                  >      > > >                  >      > > >                  >      >   my opensips-cli.cfg > >                  >      > *[default] > >                  >      > database_modules: All > >                  >      > prompt_intro: Welcome to OpenSIPS > >                  >      > > >                  >      > database_admin_url: > >                 mysql://root:cccl0g1c at localhost > >                  >      > #database_url: > >                   mysql://opensips:opensipsrw at localhost > >                  >      > database_schema_path: > /opt/opensips-3.1/scripts > >                  >      > > >                  >      > [10.0.1.1] > >                  >      > prompt_name: opensips-cli at 10.0.1.1 > > >                  > > >                  > >                  >> > >                  >      > >                  > > >                  > >                  >>> > >                  >      > prompt_intro: OpenSIPS CLI for 10.0.1.1 > instance > >                  >      > communication_type: fifo > >                  >      > url: http://10.0.1.1:8888/mi > > >                  > > >                 >> > >                  >      > >                  > > >                  >>>* > >                  >      > > >                  >      > Am I doing anything wrong ? Please do > help me . > >                  >      > > >                  >      > > >                  >      > */Thanks & Regards/* > >                  >      > /Sasmita Panda/ > >                  >      > /Senior Network Testing and Software > Engineer/ > >                  >      > /3CLogic , ph:07827611765/ > >                  >      > > >                  >      > > _______________________________________________ > >                  >      > Users mailing list > >                  >      > Users at lists.opensips.org > > >                  > > >                  > >                  >> > >                  >      > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >   > > >                  > > > >   > >> > >                  >      > > >                  > > >                  > >  _______________________________________________ > >                  >     Users mailing list > >                  > Users at lists.opensips.org > > >                  > > >                  > >                  >> > >                  > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >   > > >                  > > > >   > >> > >                  > > >                  > > >                  > _______________________________________________ > >                  > Users mailing list > >                  > Users at lists.opensips.org > > >                  > > >                  > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >   > > >                  > > > > >                 _______________________________________________ > >                 Users mailing list > > Users at lists.opensips.org > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >   > > > > >         _______________________________________________ > >         Users mailing list > > Users at lists.opensips.org > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >          > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > From razvan at opensips.org Thu Apr 15 14:23:26 2021 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 15 Apr 2021 17:23:26 +0300 Subject: [OpenSIPS-Users] Issue with opensips-cli . In-Reply-To: References: <4f177b0b-8ad8-2400-a1f8-9d726ab1dc9e@opensips.org> <91629096-8de4-9a6e-8458-8855d36a629f@opensips.org> <138054fd-aee3-b2ad-0d53-3afbba9b28c1@opensips.org> <83ddb502-1f2f-8d16-d506-a8f5156db3bc@opensips.org> Message-ID: <203720ef-a5cc-f0a0-c31d-fde09dd9d180@opensips.org> Can you also post the configuration of the httpd and mi_http modules in opensips? Have you tried using the default values? If yes, your call should work with the url=http://10.x.x.x:8888/mi Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 4/15/21 5:00 PM, Saurabh Chopra wrote: > Hi Razvan, > > The remote server is listening on HEP ports(9060) > And from opensips-CLI i have got successful connection towards remote > instance 10.x.x.x. > /Connection to 10.x.x.x 8888 port [tcp/*] succeeded!/ > > On Thu, Apr 15, 2021 at 7:26 PM Saurabh Chopra > wrote: > > Hi Razvan, > The remote server is listening on HEP ports(9060) > And from opensips-CLI i have got successful connection towards > remote instance 10.x.x.x. > /Connection to 10.x.x.x 8888 port [tcp/*] succeeded!/ > > Best Regards > Saurabh Chopra > +918861979979 > > > On Thu, Apr 15, 2021 at 7:22 PM Saurabh Chopra > wrote: > > Hi Razvan, > > The remote server is listening on HEP ports(9060) > And from opensips-CLI i have got successful connection towards > remote instance 10.x.x.x. > /Connection to 10.x.x.x 8888 port [tcp/*] succeeded!/ > > Best Regards > Saurabh Chopra > +918861979979 > > > On Thu, Apr 15, 2021 at 7:07 PM Răzvan Crainea > > wrote: > > Can you make sure the remote opensips is actually listening > on that IP > and port? >  From the host you're running opensips-cli, can you run `nc > -zv 10.x.x.x > 8888`? > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > > On 4/15/21 4:12 PM, Saurabh Chopra wrote: > > Hi Razvan, > > > > Thank you for your response. > > > > I have changed the parameters but the problem still > persists. Below is > > the opensips-cli.cfg > > > > [10.x.x.x] > > prompt_name: opensips-cli at 10.x.x.x > > prompt_intro: OpenSIPS CLI for 10.x.x.x instance > > communication_type: http > > url: http://10.x.x.x:8888/mi > > > > > > Getting below errors when i tried to switch to instance > 10.x.x.x > > /(opensips-cli): instance switch 10.x.x.x > > OpenSIPS CLI for 10.x.x.x instance > > ERROR: communication exception for 'which' returned: > > ERROR: Is OpenSIPS running?/ > > Please advise what wrong I am doing now. > > > > > > Best Regards > > Saurabh Chopra > > +918861979979 > > > > > > On Thu, Apr 15, 2021 at 4:11 PM Răzvan Crainea > > > >> wrote: > > > >     Hi, Sasmita, Saurabh! > > > >     1. fifo, by its nature, can only be used for > localhost queries; if you > >     need to run remote commands, you need to use a remote > communication > >     protocol, such as http > >     2. Most likely you are querying a bad http root path > for the `mi_http` > >     module: in your URL it shows you are querying the > `json` root, whereas > >     the default mi_http root is `mi`. Can you make sure > A. you're not > >     modifying the mi_http root parameter[1], and/or if > you do, make sure it > >     corresponds to the path in the URL? > > > >     [1] > https://opensips.org/docs/modules/3.1.x/mi_http.html#param_root > > > >  > > > > >     Best regards, > > > >     Răzvan Crainea > >     OpenSIPS Core Developer > > http://www.opensips-solutions.com > > > > > > >     On 4/15/21 12:51 PM, Saurabh Chopra wrote: > >      > Hi Guys, > >      > > >      > Any update would be appreciated regarding the > above matter. > >      > > >      > Best Regards > >      > Saurabh Chopra > >      > +918861979979 > >      > > >      > > >      > On Wed, Apr 14, 2021 at 3:33 PM Saurabh Chopra > >      > > > >      > >>> wrote: > >      > > >      >     Hi Opensips Team/ Razvan, > >      > > >      >     As per Sasmita's previous mail, we are still > stuck in a > >     situation where, > >      > > >      >     1. If we are using "FIFO" as a > communication_type we are not > >     getting > >      >     data from a remote machine instead getting > data from a > >     local host. > >      >     2. If we are using "HTTP" as a > communication_type then we are > >     unable > >      >     to switch to a remote machine. > >      > > >      >     For both scenarios logs have been attached in > *pastebin*. > >      > > >      >     Kindly advise and provide us with an early > resolution. We will be > >      >     waiting for your prompt response > >      > > >      >     Best Regards > >      >     Saurabh Chopra > >      >     +918861979979 > >      > > >      > > >      >     On Tue, Mar 30, 2021 at 11:31 AM Sasmita Panda > >      > > > >      >      >>> wrote: > >      > > >      >         Hi , > >      > > >      >         I have changed my instance.cfg like bellow . > >      > > >      >         [10.0.0.1] > >      >         prompt_name: opensips-cli at 10.0.0.1 > > >      > > > >      >> > >      >         communication_type: http > >      >         url: http://10.0.0.1:8888/json > > >      > > >     >> > >      > > >      >         With this config above I am getting some > error . I am not > >     able > >      >         to switch to the remote instance as well . > >      > > >      > https://pastebin.com/1fDuHbBk > > > > >      > >> > >      > > >      > > >      >         if I am giving below config . > >      > > >      >           [10.0.0.1] > >      >         prompt_name: opensips-cli at 10.0.0.1 > > >      > > > >      >> > >      >         communication_type: fifo > >      >         fifo_file: /tmp/opensips_fifo > >      > > >      >         I am able to switch to the remote instance > . After switching > >      >         when I am running any command its giving > the result for > >      >         localhost not  for the remote instance ;. > >      > > >      > > >      >         What should I do ? Am I missing any > dependency library ? > >     Please > >      >         do help . > >      > > >      > > >      > > >      > > >      >         */Thanks & Regards/* > >      >         /Sasmita Panda/ > >      >         /Senior Network Testing and Software Engineer/ > >      >         /3CLogic , ph:07827611765/ > >      > > >      > > >      >         On Fri, Mar 26, 2021 at 5:58 PM Sasmita Panda > >      >          > > >      >>> wrote: > >      > > >      >             same result . > >      > > >      >             my instance.cfg contains : > >      >             cat instance.cfg > >      >             [10.0.1.1] > >      >             prompt_name: opensips-cli at 10.0.1.1 > > >      > > >      >              > >      >> > >      >             fifo_file: /tmp/opensips_fifo > >      > > >      > https://pastebin.com/FQBipyZJ > > > > >      > >> > >      >             This is the result . > >      > > >      >             */ > >      >             /* > >      >             */ > >      >             /* > >      >             */Thanks & Regards/* > >      >             /Sasmita Panda/ > >      >             /Senior Network Testing and Software > Engineer/ > >      >             /3CLogic , ph:07827611765/ > >      > > >      > > >      >             On Fri, Mar 26, 2021 at 5:38 PM Răzvan > Crainea > >      >              > > >      >>> wrote: > >      > > >      >                 You are not switching the instance > correctly: should > >      >                 have used `instance > >      >                 switch 10.0.1.1` > >      > > >      >                 Best regards, > >      > > >      >                 Răzvan Crainea > >      >                 OpenSIPS Core Developer > >      > http://www.opensips-solutions.com > > > > >      >                  > >      >> > >      > > >      >                 On 3/26/21 1:54 PM, Sasmita Panda > wrote: > >      >                  > I am not getting any error for > this . I have > >     pasted > >      >                 the logs in pastebin . > >      >                  > > >      >                  > https://pastebin.com/4tzXzrrG > > >      > > >      >                  > >      >> > >      >                  > >      > > >      >                  > >      >>> > >      >                  > > >      >                  > What needs to be done in the > remote machine > >     10.0.1.1 > >      >                 . I have allowed > >      >                  > 8888 tcp port . I saw default > http url is like > >      > http://10.0.1.1:8888/mi > > > >      > >> > >      >                  > > >     > > > >     >>> . > >      >                  > Do suggest if I am missing > anything . > >      >                  > > >      >                  > > >      >                  > */Thanks & Regards/* > >      >                  > /Sasmita Panda/ > >      >                  > /Senior Network Testing and > Software Engineer/ > >      >                  > /3CLogic , ph:07827611765/ > >      >                  > > >      >                  > > >      >                  > On Fri, Mar 26, 2021 at 5:00 PM > Răzvan Crainea > >      >                  > > >      >> > >      >                  > > >     > > >      >                  > >      >>>> wrote: > >      >                  > > >      >                  >     Hi, Sasmita! > >      >                  > > >      >                  >     Can you post the logs of an > execution for the > >      >                 10.0.1.1 instance run > >      >                  >     with > >      >                  >     debugging (-d parameter)? > Also, please > >     post them > >      >                 on pastebin[1] or in a > >      >                  >     different place, not in the > mail itself. > >      >                  > > >      >                  >     [1] https://pastebin.com/ > > >     > > > >> > >      >                  > > >      > >>> > >      >                  > > >      >                  >     Best regards, > >      >                  > > >      >                  >     Răzvan Crainea > >      >                  >     OpenSIPS Core Developer > >      >                  > > http://www.opensips-solutions.com > > >      > > >      >                  > >      >> > >      >                  > >      > > >      >                  > >      >>> > >      >                  > > >      >                  >     On 3/26/21 11:16 AM, > Sasmita Panda wrote: > >      >                  >      > Hi All , > >      >                  >      > > >      >                  >      > I am using opensips-cli > ( 0.1.0 ) and > >      >                 opensips-3.1 . > >      >                  >      > > >      >                  >      > I am able to get all the > data of the local > >      >                 host through cli but > >      >                  >     when I > >      >                  >      > am connecting to a > remote host even though > >      >                 it's getting connected > >      >                  >     but no > >      >                  >      > data is visible for the > remote > >     instance  . It > >      >                 only showing the > >      >                  >     data of > >      >                  >      > the instance on which I > am running the > >     cli . > >      >                  >      > > >      >                  >      > > >      >                  >      >   my opensips-cli.cfg > >      >                  >      > *[default] > >      >                  >      > database_modules: All > >      >                  >      > prompt_intro: Welcome to > OpenSIPS > >      >                  >      > > >      >                  >      > database_admin_url: > >      >                 mysql://root:cccl0g1c at localhost > >      >                  >      > #database_url: > >      > >  mysql://opensips:opensipsrw at localhost > >      >                  >      > database_schema_path: > >     /opt/opensips-3.1/scripts > >      >                  >      > > >      >                  >      > [10.0.1.1] > >      >                  >      > prompt_name: > opensips-cli at 10.0.1.1 > >      > > >      >                  > >      >> > >      >                  > >      > > >      >                  > >      >>> > >      >                  > >   > >      > > >      >                  > >      >> > >      >                  > >      > > >      >                  > >      >>>> > >      >                  >      > prompt_intro: OpenSIPS > CLI for 10.0.1.1 > >     instance > >      >                  >      > communication_type: fifo > >      >                  >      > url: > http://10.0.1.1:8888/mi > >     > > >      >                  > >     >> > > >     > > >      >                  >>> > >      >                  >      > >     > > >      >                  > >     >> > > >     > > >      >                  > >     >>>>* > >      >                  >      > > >      >                  >      > Am I doing anything > wrong ? Please do > >     help me . > >      >                  >      > > >      >                  >      > > >      >                  >      > */Thanks & Regards/* > >      >                  >      > /Sasmita Panda/ > >      >                  >      > /Senior Network Testing > and Software > >     Engineer/ > >      >                  >      > /3CLogic , ph:07827611765/ > >      >                  >      > > >      >                  >      > > >     _______________________________________________ > >      >                  >      > Users mailing list > >      >                  >      > Users at lists.opensips.org > > >      > > >      >                  > >      >> > >      >                  > >      > > >      >                  > >      >>> > >      >                  >      > > >      > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >   > > >      > > > >   > > >   >> > >      >                  > > >      > > > >   > > >   > > > >   > > >   >>> > >      >                  >      > > >      >                  > > >      >                  > > >       _______________________________________________ > >      >                  >     Users mailing list > >      >                  > Users at lists.opensips.org > > >      > > >      >                  > >      >> > >      >                  > >      > > >      >                  > >      >>> > >      >                  > > >      > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >   > > >      > > > >   > > >   >> > >      >                  > > >      > > > >   > > >   > > > >   > > >   >>> > >      >                  > > >      >                  > > >      >                  > > _______________________________________________ > >      >                  > Users mailing list > >      >                  > Users at lists.opensips.org > > >      > > >      >                  > >      >> > >      >                  > > >      > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >   > > >      > > > >   > > >   >> > >      >                  > > >      > > >      > >  _______________________________________________ > >      >                 Users mailing list > >      > Users at lists.opensips.org > > > > >      > >> > >      > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >   > > >      > > > >   > > >   >> > >      > > >      > >  _______________________________________________ > >      >         Users mailing list > >      > Users at lists.opensips.org > > > > >      > >> > >      > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >   > > >      > >   > > >   >> > >      > > >      > > >      > _______________________________________________ > >      > Users mailing list > >      > Users at lists.opensips.org > > > > >      > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >   > > >      > > > > >     _______________________________________________ > >     Users mailing list > > Users at lists.opensips.org > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > >   > > > > From bullehs at gmail.com Thu Apr 15 15:00:50 2021 From: bullehs at gmail.com (HS) Date: Thu, 15 Apr 2021 20:00:50 +0500 Subject: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications In-Reply-To: References: <48bbe2f1-0131-03e1-69ca-34fd872d066f@opensips.org> Message-ID: Bogdan-Andrei, Thanks a lot for sharing. Appreciate the help. Do we need to make any changes for the script to work with iOS? Apologies, but really difficult to find help with Opensips where I am located. Much appreciate your time and help. Best wishes. On Thu, Apr 15, 2021 at 11:01 AM Bogdan-Andrei Iancu wrote: > The last link is for OpenSIPS 2.3 and higher. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/14/21 7:33 PM, HS wrote: > > Bogdan-Andrei, > > Thanks a lot for sharing these links. I had actually looked at them > previously, all are for Opensips 3.1 - any possibility of getting PN to > work with Openips 3.0 please? > > Thanks. > > On Wed, Apr 14, 2021 at 3:53 PM Bogdan-Andrei Iancu > wrote: > >> See >> >> >> https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/ >> >> >> https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ >> >> >> https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp 2021 online >> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >> >> On 4/9/21 12:39 PM, HS wrote: >> >> Hello all. >> >> We are using Opensips 3.0 (with rtpproxy) and been looking to implement >> Push Notifications for iOS devices. Did come across quite a few resources, >> but they seem either too old ~2013 time or about V3.1. Has anyone >> implemented something to make calls to devices with the UAC in background >> mode on V3.0 pls? >> >> Or suggestions on how to make this work on v3.0? >> >> http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/ >> >> Appreciate the help. >> >> Thanks. >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From calvin.ellison at voxox.com Thu Apr 15 17:47:24 2021 From: calvin.ellison at voxox.com (Calvin Ellison) Date: Thu, 15 Apr 2021 10:47:24 -0700 Subject: [OpenSIPS-Users] Gracefully handling bad from header Message-ID: I would like to more gracefully handle non-RFC compliant From headers. Is it possible to fix them with regex or a string replacement so I can process them? Most often the misbehavior is a space in the user part: #015#012To: #015#012From: ;tag=sansay2313795380rdb15209#015#012Call-ID: ers-1718949283-0-660497714 at redact#015#012CSeq: 1 INVITE#015#012Contact: #015#012P-Asserted-Identity: #015#012Remote-Party-ID: ;privacy=off;screen=yes#015#012Max-Forwards: 63#015#012Content-Length: 0#015#012#015#012 Regards, *Calvin Ellison* Systems Architect calvin.ellison at voxox.com +1 (213) 285-0555 ----------------------------------------------- *voxox.com * 5825 Oberlin Drive, Suite 5 San Diego, CA 92121 [image: Voxox] -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 16 06:26:33 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 16 Apr 2021 09:26:33 +0300 Subject: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications In-Reply-To: References: <48bbe2f1-0131-03e1-69ca-34fd872d066f@opensips.org> Message-ID: <7c24e2da-7dd0-8ece-9a64-844c7a33109f@opensips.org> Hi, As you see, the script does not have anything special for triggering the PN (which may be OS specific). Normally, just before doing the "wait", you should trigger (rest_client, exec, etc) the sending of the PN. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/15/21 6:00 PM, HS wrote: > Bogdan-Andrei, > > Thanks a lot for sharing. Appreciate the help. Do we need to make any > changes for the script to work with iOS? Apologies, but really > difficult to find help with Opensips where I am located. > > Much appreciate your time and help. > > Best wishes. > > On Thu, Apr 15, 2021 at 11:01 AM Bogdan-Andrei Iancu > > wrote: > > The last link is for OpenSIPS 2.3 and higher. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/14/21 7:33 PM, HS wrote: >> Bogdan-Andrei, >> >> Thanks a lot for sharing these links. I had actually looked at >> them previously, all are for Opensips 3.1 - any possibility of >> getting PN to work with Openips 3.0 please? >> >> Thanks. >> >> On Wed, Apr 14, 2021 at 3:53 PM Bogdan-Andrei Iancu >> > wrote: >> >> See >> >> https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/ >> >> https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ >> >> https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp 2021 online >> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >> >> On 4/9/21 12:39 PM, HS wrote: >>> Hello all. >>> >>> We are using Opensips 3.0 (with rtpproxy) and been looking >>> to implement Push Notifications for iOS devices. Did come >>> across quite a few resources, but they seem either too old >>> ~2013 time or about V3.1. Has anyone implemented something >>> to make calls to devices with the UAC in background mode on >>> V3.0 pls? >>> >>> Or suggestions on how to make this work on v3.0? >>> http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/ >>> >>> Appreciate the help. >>> >>> Thanks. >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bullehs at gmail.com Fri Apr 16 07:36:51 2021 From: bullehs at gmail.com (HS) Date: Fri, 16 Apr 2021 12:36:51 +0500 Subject: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications In-Reply-To: <7c24e2da-7dd0-8ece-9a64-844c7a33109f@opensips.org> References: <48bbe2f1-0131-03e1-69ca-34fd872d066f@opensips.org> <7c24e2da-7dd0-8ece-9a64-844c7a33109f@opensips.org> Message-ID: Bogdan-Andrei, Thanks a lot for the help. Much appreciated. Best wishes. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 16 07:50:44 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 16 Apr 2021 10:50:44 +0300 Subject: [OpenSIPS-Users] Gracefully handling bad from header In-Reply-To: References: Message-ID: <3521d5b7-6190-253c-90a0-ce54211f551c@opensips.org> Hi Calvin, I'm afraid there is not much you can do, especially if (via script and modules) you will need access to the "fixed" FROM  - the msg changes you do via script are not applied on the spot, but only when the msg is sent out -> so you do not see your own changes. But if you do not need to access the FROM hdr, you can do a brutal replacement with remove_hf() and append_hf(), but not sure if you will be able to extract any other parts of the FROM (if you want o preserve them). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/15/21 8:47 PM, Calvin Ellison wrote: > I would like to more gracefully handle non-RFC compliant From headers. > Is it possible to fix them with regex or a string replacement so I can > process them? > > Most often the misbehavior is a space in the user part: > > > Other times there's an unclosed double quote on the friendly-name part > or some kind of non-printing character in the friendly-name or user > part. It's garbage that will ultimately be rejected, but the failure > to parse is preventing me from logging the offending trunk ID in the > 'otg=' parameter of the From header. > > opensips 3.1.1 (x86_64/linux) with commit to remove unnecessary regex > limitation > https://github.com/OpenSIPS/opensips/commit/a8f8aabc245ca86bd40a9eddf9befcd74a087be1 > > Here's the error followed by xlog $mb > > Apr 15 15:26:24  /usr/sbin/opensips[13864]: > ERROR:core:parse_from_header: bad from header > Apr 15 15:26:24  /usr/sbin/opensips[13864]: > ERROR:core:pv_get_from_attr: cannot parse From header > Apr 15 15:26:24  /usr/sbin/opensips[13864]: INVITE > sip:redact at redact:5060 SIP/2.0#015#012Via: SIP/2.0/UDP > redact:5060;branch=z9hG4bK1sansay2313795380rdb15209#015#012Record-Route: > #015#012To: > #015#012From: @redact;otg=redact;qtype=ers>;tag=sansay2313795380rdb15209#015#012Call-ID: > ers-1718949283-0-660497714 at redact#015#012CSeq: 1 > INVITE#015#012Contact: @redact:5060>#015#012P-Asserted-Identity: > #015#012Remote-Party-ID: > ;privacy=off;screen=yes#015#012Max-Forwards: > 63#015#012Content-Length: 0#015#012#015#012 > > > Regards, > > *Calvin Ellison* > Systems Architect > calvin.ellison at voxox.com > +1 (213) 285-0555 > > ----------------------------------------------- > *voxox.com * > 5825 Oberlin Drive, Suite 5 > San Diego, CA 92121 > > Voxox > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From solarmon at one-n.co.uk Fri Apr 16 09:22:08 2021 From: solarmon at one-n.co.uk (solarmon) Date: Fri, 16 Apr 2021 10:22:08 +0100 Subject: [OpenSIPS-Users] t_reply() with Reason header Message-ID: Hi, We have a code that response back with a "503 Service not available" if it has exhausted all the dispatcher endpoints - see end of email. It would be useful if we casn send back the Reason header of the last rejection. Could this be done and what should I look at? Thank you! failure_route[call_failover] { xlog("[$ci] call failed to established with $T_reply_code code\n"); rtpproxy_unforce("$avp(rtpp_set)"); if (t_was_cancelled()) { t_reply("487","Request cancelled"); exit; } # any failure indication ? if ( t_check_status("[56][0-9][0-9]") || (t_check_status("408") && t_local_replied("all")) ) { xlog("[$ci] destination $rd failed with $T_reply_code -> retry\n "); ###ds_mark_dst("p"); if ( ds_next_domain() ) { xlog("[$ci] using new destination <$rd>\n "); # send it out again t_on_failure("call_failover"); t_relay(); exit; } else { xlog("[$ci] no other destination to retry\n "); t_reply("503","Service not available"); exit; } } # if call failure, allow the reply to propagate to caller exit; } -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 16 10:06:53 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 16 Apr 2021 13:06:53 +0300 Subject: [OpenSIPS-Users] t_reply() with Reason header In-Reply-To: References: Message-ID: <686e4059-6a2f-b99a-6305-f852f4ff4b80@opensips.org> Hi, use the append_to_reply() function https://opensips.org/html/docs/modules/3.1.x/sipmsgops.html#func_append_to_reply Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/16/21 12:22 PM, solarmon wrote: > Hi, > > We have a code that response back with a "503 Service not available" > if it has exhausted all the dispatcher endpoints - see end of email. > > It would be useful if we casn send back the Reason header of the last > rejection. Could this be done and what should I look at? > > Thank you! > > failure_route[call_failover] > { >         xlog("[$ci] call failed to established with $T_reply_code > code\n"); > >         rtpproxy_unforce("$avp(rtpp_set)"); > >         if (t_was_cancelled()) { >                 t_reply("487","Request cancelled"); >                 exit; >         } > >         # any failure indication ? >         if ( t_check_status("[56][0-9][0-9]") >         || (t_check_status("408") && t_local_replied("all")) >         ) { >                 xlog("[$ci] destination $rd failed  with $T_reply_code > -> retry\n "); > >                 ###ds_mark_dst("p"); > >                 if ( ds_next_domain() ) { >                         xlog("[$ci] using new destination <$rd>\n "); > >                         # send it out again >                         t_on_failure("call_failover"); >                         t_relay(); >                         exit; >                 } else { >                         xlog("[$ci] no other destination to retry\n "); >                         t_reply("503","Service not available"); >                         exit; >                 } >         } > >         # if call failure, allow the reply to propagate to caller >         exit; > } > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From solarmon at one-n.co.uk Fri Apr 16 10:28:40 2021 From: solarmon at one-n.co.uk (solarmon) Date: Fri, 16 Apr 2021 11:28:40 +0100 Subject: [OpenSIPS-Users] t_reply() with Reason header In-Reply-To: <686e4059-6a2f-b99a-6305-f852f4ff4b80@opensips.org> References: <686e4059-6a2f-b99a-6305-f852f4ff4b80@opensips.org> Message-ID: Hi Bogdan, Thanks for the response amd link to append_to_reply() What is the best function to use to extract the Reason header to feed in to append_to_reply()? Thank you. On Fri, 16 Apr 2021 at 11:07, Bogdan-Andrei Iancu wrote: > Hi, > > use the append_to_reply() function > > > https://opensips.org/html/docs/modules/3.1.x/sipmsgops.html#func_append_to_reply > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/16/21 12:22 PM, solarmon wrote: > > Hi, > > We have a code that response back with a "503 Service not available" if it > has exhausted all the dispatcher endpoints - see end of email. > > It would be useful if we casn send back the Reason header of the last > rejection. Could this be done and what should I look at? > > Thank you! > > failure_route[call_failover] > { > xlog("[$ci] call failed to established with $T_reply_code code\n"); > > rtpproxy_unforce("$avp(rtpp_set)"); > > if (t_was_cancelled()) { > t_reply("487","Request cancelled"); > exit; > } > > # any failure indication ? > if ( t_check_status("[56][0-9][0-9]") > || (t_check_status("408") && t_local_replied("all")) > ) { > xlog("[$ci] destination $rd failed with $T_reply_code -> > retry\n "); > > ###ds_mark_dst("p"); > > if ( ds_next_domain() ) { > xlog("[$ci] using new destination <$rd>\n "); > > # send it out again > t_on_failure("call_failover"); > t_relay(); > exit; > } else { > xlog("[$ci] no other destination to retry\n "); > t_reply("503","Service not available"); > exit; > } > } > > # if call failure, allow the reply to propagate to caller > exit; > } > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 16 11:43:50 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 16 Apr 2021 14:43:50 +0300 Subject: [OpenSIPS-Users] t_reply() with Reason header In-Reply-To: References: <686e4059-6a2f-b99a-6305-f852f4ff4b80@opensips.org> Message-ID: <69effa47-9e90-500d-db10-23a89ae87b5e@opensips.org> Where do you want to extract the Reason from (like sip msg) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/16/21 1:28 PM, solarmon wrote: > Hi Bogdan, > > Thanks for the response amd link to append_to_reply() > > What is the best function to use to extract the Reason header to feed > in to append_to_reply()? > > Thank you. > > On Fri, 16 Apr 2021 at 11:07, Bogdan-Andrei Iancu > wrote: > > Hi, > > use the append_to_reply() function > > https://opensips.org/html/docs/modules/3.1.x/sipmsgops.html#func_append_to_reply > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/16/21 12:22 PM, solarmon wrote: >> Hi, >> >> We have a code that response back with a "503 Service not >> available" if it has exhausted all the dispatcher endpoints - see >> end of email. >> >> It would be useful if we casn send back the Reason header of the >> last rejection. Could this be done and what should I look at? >> >> Thank you! >> >> failure_route[call_failover] >> { >>         xlog("[$ci] call failed to established with $T_reply_code >> code\n"); >> >>         rtpproxy_unforce("$avp(rtpp_set)"); >> >>         if (t_was_cancelled()) { >>                 t_reply("487","Request cancelled"); >>                 exit; >>         } >> >>         # any failure indication ? >>         if ( t_check_status("[56][0-9][0-9]") >>         || (t_check_status("408") && t_local_replied("all")) >>         ) { >>                 xlog("[$ci] destination $rd failed  with >> $T_reply_code -> retry\n "); >> >>                 ###ds_mark_dst("p"); >> >>                 if ( ds_next_domain() ) { >>                         xlog("[$ci] using new destination <$rd>\n "); >> >>                         # send it out again >>                         t_on_failure("call_failover"); >>                         t_relay(); >>                         exit; >>                 } else { >>                         xlog("[$ci] no other destination to >> retry\n "); >>                         t_reply("503","Service not available"); >>                         exit; >>                 } >>         } >> >>         # if call failure, allow the reply to propagate to caller >>         exit; >> } >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From solarmon at one-n.co.uk Fri Apr 16 11:59:53 2021 From: solarmon at one-n.co.uk (solarmon) Date: Fri, 16 Apr 2021 12:59:53 +0100 Subject: [OpenSIPS-Users] t_reply() with Reason header In-Reply-To: <69effa47-9e90-500d-db10-23a89ae87b5e@opensips.org> References: <686e4059-6a2f-b99a-6305-f852f4ff4b80@opensips.org> <69effa47-9e90-500d-db10-23a89ae87b5e@opensips.org> Message-ID: Hi Bogdan, It is in the response coming back to us from the dispatcher endpoint. In this case, it is an 402 Bad Gateway message, which has a Reason header with the cause code in it. On Fri, 16 Apr 2021 at 12:43, Bogdan-Andrei Iancu wrote: > Where do you want to extract the Reason from (like sip msg) ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/16/21 1:28 PM, solarmon wrote: > > Hi Bogdan, > > Thanks for the response amd link to append_to_reply() > > What is the best function to use to extract the Reason header to feed in > to append_to_reply()? > > Thank you. > > On Fri, 16 Apr 2021 at 11:07, Bogdan-Andrei Iancu > wrote: > >> Hi, >> >> use the append_to_reply() function >> >> >> https://opensips.org/html/docs/modules/3.1.x/sipmsgops.html#func_append_to_reply >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp 2021 online >> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >> >> On 4/16/21 12:22 PM, solarmon wrote: >> >> Hi, >> >> We have a code that response back with a "503 Service not available" if >> it has exhausted all the dispatcher endpoints - see end of email. >> >> It would be useful if we casn send back the Reason header of the last >> rejection. Could this be done and what should I look at? >> >> Thank you! >> >> failure_route[call_failover] >> { >> xlog("[$ci] call failed to established with $T_reply_code >> code\n"); >> >> rtpproxy_unforce("$avp(rtpp_set)"); >> >> if (t_was_cancelled()) { >> t_reply("487","Request cancelled"); >> exit; >> } >> >> # any failure indication ? >> if ( t_check_status("[56][0-9][0-9]") >> || (t_check_status("408") && t_local_replied("all")) >> ) { >> xlog("[$ci] destination $rd failed with $T_reply_code -> >> retry\n "); >> >> ###ds_mark_dst("p"); >> >> if ( ds_next_domain() ) { >> xlog("[$ci] using new destination <$rd>\n "); >> >> # send it out again >> t_on_failure("call_failover"); >> t_relay(); >> exit; >> } else { >> xlog("[$ci] no other destination to retry\n "); >> t_reply("503","Service not available"); >> exit; >> } >> } >> >> # if call failure, allow the reply to propagate to caller >> exit; >> } >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 16 12:09:00 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 16 Apr 2021 15:09:00 +0300 Subject: [OpenSIPS-Users] t_reply() with Reason header In-Reply-To: References: <686e4059-6a2f-b99a-6305-f852f4ff4b80@opensips.org> <69effa47-9e90-500d-db10-23a89ae87b5e@opensips.org> Message-ID: <535de831-1e2f-2769-df0b-68f7408f3b37@opensips.org> try via $(hdr(Reason)) in failure route, but be sure that the 402 is the winning reply, triggering the failover route. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/16/21 2:59 PM, solarmon wrote: > Hi Bogdan, > > It is in the response coming back to us from the dispatcher endpoint. > In this case, it is an 402 Bad Gateway message, which has a Reason > header with the cause code in it. > > On Fri, 16 Apr 2021 at 12:43, Bogdan-Andrei Iancu > wrote: > > Where do you want to extract the Reason from (like sip msg) ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/16/21 1:28 PM, solarmon wrote: >> Hi Bogdan, >> >> Thanks for the response amd link to append_to_reply() >> >> What is the best function to use to extract the Reason header to >> feed in to append_to_reply()? >> >> Thank you. >> >> On Fri, 16 Apr 2021 at 11:07, Bogdan-Andrei Iancu >> > wrote: >> >> Hi, >> >> use the append_to_reply() function >> >> https://opensips.org/html/docs/modules/3.1.x/sipmsgops.html#func_append_to_reply >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp 2021 online >> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >> >> On 4/16/21 12:22 PM, solarmon wrote: >>> Hi, >>> >>> We have a code that response back with a "503 Service not >>> available" if it has exhausted all the dispatcher endpoints >>> - see end of email. >>> >>> It would be useful if we casn send back the Reason header of >>> the last rejection. Could this be done and what should I >>> look at? >>> >>> Thank you! >>> >>> failure_route[call_failover] >>> { >>>         xlog("[$ci] call failed to established with >>> $T_reply_code code\n"); >>> >>>         rtpproxy_unforce("$avp(rtpp_set)"); >>> >>>         if (t_was_cancelled()) { >>>                 t_reply("487","Request cancelled"); >>>                 exit; >>>         } >>> >>>         # any failure indication ? >>>         if ( t_check_status("[56][0-9][0-9]") >>>         || (t_check_status("408") && t_local_replied("all")) >>>         ) { >>>                 xlog("[$ci] destination $rd failed  with >>> $T_reply_code -> retry\n "); >>> >>>                 ###ds_mark_dst("p"); >>> >>>                 if ( ds_next_domain() ) { >>>                         xlog("[$ci] using new destination >>> <$rd>\n "); >>> >>>                         # send it out again >>> t_on_failure("call_failover"); >>>                         t_relay(); >>>                         exit; >>>                 } else { >>>                         xlog("[$ci] no other destination to >>> retry\n "); >>>                         t_reply("503","Service not available"); >>>                         exit; >>>                 } >>>         } >>> >>>         # if call failure, allow the reply to propagate to >>> caller >>>         exit; >>> } >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mwbazdarich at yahoo.com Tue Apr 20 08:05:14 2021 From: mwbazdarich at yahoo.com (mwb) Date: Tue, 20 Apr 2021 08:05:14 +0000 (UTC) Subject: [OpenSIPS-Users] Removing INVITE Contact from Reply (2.1.5) References: <1741851750.4222838.1618905914682.ref@mail.yahoo.com> Message-ID: <1741851750.4222838.1618905914682@mail.yahoo.com> Hello,I posed a questions a few months ago and received some helpful advice from a user here but I'm having an issue implementing it properly. We have a redirect server using OpenSIPS 2.1.5 that does LNP dipping with a little bit of routing as well.  There are two scenarios: 1. There is No RN/NPDI in the RURI (so the redirect server sends a LNP request to an external LNP server 2. The INVITE has RN/NPDI in the RURI so no LNP dip is dispatched. In the second scenario we do a few operations and generate URIs to embed in a 302 reply using the send_reply function in a . The issue we are running into is that it appears the $ds variable is being populated with the INVITE Request URI (RN and/or NPDI) info and it is being sent as the "top" Contact header in the 302 response.  I have tried a few different things, including "remove_branch(0)" but they don't seem to be able to clobber the value in the $ds variable. It also seems impossible to use $ds as an LVALUE to try and overwrite it (this is consistent with the documentation which says it is not r/w). Is there a way to: 1. Clobber the $ds variable?2. Preclude the $ds variable from being inserted as a Contact header in the reply?3. Keep the $ds variable from being populated when the initial INVITE is received. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From miha at softnet.si Tue Apr 20 08:15:12 2021 From: miha at softnet.si (Miha) Date: Tue, 20 Apr 2021 10:15:12 +0200 Subject: [OpenSIPS-Users] how can i combine signaling and RTP from rtpproxy Message-ID: Hello due to debugging i would like to combine cap from opensips and also cap from rtpproxy (they are on different servers) so that I can check if RTP is missing for certain call. Can you help me with solving this issue :) thank you miha -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Apr 20 11:59:53 2021 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 20 Apr 2021 14:59:53 +0300 Subject: [OpenSIPS-Users] =?utf-8?b?W0Jsb2ddIE9wZW5TSVBJdCDigJkwMSDigJMg?= =?utf-8?q?Testing_the_Trending_SIP_Security_Enhancements?= Message-ID: Hi, folks! Last week, there was quite a bit of buzz around the first official OpenSIPIt event.  Check out the below blog post to see how things went and what the status quo around the new SIP "Security RFCs" looks like. https://blog.opensips.org/2021/04/20/opensipit-01-testing-the-trending-sip-security-enhancements/ Enjoy! -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS eBootcamp 2021 | www.opensips.org/training From apsaras at microbase.gr Tue Apr 20 13:55:18 2021 From: apsaras at microbase.gr (Antonis Psaras) Date: Tue, 20 Apr 2021 16:55:18 +0300 Subject: [OpenSIPS-Users] Generate CANCEL on 180 Message-ID: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> Dear all I am trying to create a service which will generate missed calls. In order to be more accurate, I want to CANCEL the request when 180 is received. The scenario is the following Asterisk Invite -> OpenSIPs -> Carrier Carrier 183 -> OpenSIPs -> Asterisk Carrier 180 -> OpenSIPs OpenSIPs Cancel -> Carrier Is that possible to be done from script without external app? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From kingsley at dns99.co.uk Tue Apr 20 16:21:10 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Tue, 20 Apr 2021 17:21:10 +0100 Subject: [OpenSIPS-Users] dialplan module - dpid - could it be a string? Message-ID: <67751808587688838f3a12f60fc9c86bf1a1d95c.camel@dns99.co.uk> I love the dialplan module and how the rules can be reloaded from a DB. However, is there any reason why the dpid can't be a string instead of an integer? (ie the value that's stored in the dialplan.dpid field). I think it would be handy to be able to apply names to dialplan rulesets as it would be more intuitive. If dpid could be a string, then I could for example create a ruleset that I could refer to as "uk_local_to_e164" instead of "7". Could this be a thing? -- Cheers, Kingsley. From alain.bieuzent at free.fr Tue Apr 20 16:43:22 2021 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Tue, 20 Apr 2021 18:43:22 +0200 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> Message-ID: <01AE0372-D02E-4339-BACE-6017A56ABA4C@free.fr> Hi Antonis, it seems possible ...: https://opensips.org/html/docs/modules/3.1.x/tm.html#func_t_cancel_branch. Regards De : Users au nom de Antonis Psaras Organisation : Microbase Répondre à : , OpenSIPS users mailling list Date : mardi 20 avril 2021 à 15:58 À : Objet : [OpenSIPS-Users] Generate CANCEL on 180 Dear all I am trying to create a service which will generate missed calls. In order to be more accurate, I want to CANCEL the request when 180 is received. The scenario is the following Asterisk Invite -> OpenSIPs -> Carrier Carrier 183 -> OpenSIPs -> Asterisk Carrier 180 -> OpenSIPs OpenSIPs Cancel -> Carrier Is that possible to be done from script without external app? Regards _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From kingsley at dns99.co.uk Tue Apr 20 17:09:30 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Tue, 20 Apr 2021 18:09:30 +0100 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> Message-ID: <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> Firstly, I'm new to OpenSIPS so treat my comments accordingly. But, can you do something in an onreply route? eg, in a test setup I have, when I get an INVITE I do this: create_dialog("pPB"); t_on_reply("doodle"); (I can't remember whether the dialog is needed for this) and then I have this: onreply_route[doodle] { # expect $T_reply_code to likely first be 100 # then 180 or 183 for a progressing call # 200 when call is answered # or failure code (eg 4xx) or whatever if (t_check_status("^1[0-9][0-9]$")) { switch ($T_reply_code) { case 180: $acc_extra(t_ringing) = $Ts; break; case 183: $acc_extra(t_progress) = $Ts; break; } } else if (t_check_status("^2[0-9][0-9]$")) { $acc_extra(t_answer) = $Ts; } else { xlog("Something else\n"); } } so when a 180 is received, it calls the above route function. Could you send a CANCEL from there? Cheers, Kingsley. On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: > Dear all > > I am trying to create a service which will generate missed calls. In > order to be more accurate, I want to CANCEL the request when 180 is > received. > > The scenario is the following > > Asterisk Invite -> OpenSIPs -> Carrier > > Carrier 183 -> OpenSIPs -> Asterisk > > Carrier 180 -> OpenSIPs > > OpenSIPs Cancel -> Carrier > > > Is that possible to be done from script without external app? > > Regards > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From kingsley at dns99.co.uk Tue Apr 20 17:16:38 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Tue, 20 Apr 2021 18:16:38 +0100 Subject: [OpenSIPS-Users] dialplan module - ensuring a substring exists in URI Message-ID: <980007766d975b094a3052582ad1eea5dec480fb.camel@dns99.co.uk> Hi, I want a DP ruleset to ensure that ";user=phone" is in the SIP URI, ie if it is not there, add it, but if it is already there, do nothing. I've managed to do this with two lines, but is there a more efficient way? This is what I have in the dialplan table in this ruleset? +----+----------+-------------+-------------+-----------+---------------+ | pr | match_op | match_exp | match_flags | subst_exp | repl_exp | +----+----------+-------------+-------------+-----------+---------------+ | 1 | 1 | ;user=phone | 1 | (.+) | \1 | | 2 | 1 | . | 1 | (.+) | \1;user=phone | +----+----------+-------------+-------------+-----------+---------------+ -- Cheers, Kingsley. From Ben.Newlin at genesys.com Tue Apr 20 18:05:04 2021 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 20 Apr 2021 18:05:04 +0000 Subject: [OpenSIPS-Users] dialplan module - dpid - could it be a string? In-Reply-To: <67751808587688838f3a12f60fc9c86bf1a1d95c.camel@dns99.co.uk> References: <67751808587688838f3a12f60fc9c86bf1a1d95c.camel@dns99.co.uk> Message-ID: This sounds like it might be a good feature request [1] and I agree it sounds nice. I can’t offer any assistance in that direction at this time, but I wanted to say that we use M4 in our config scripts to allow us to define string constants like this and make the script more readable, among other reasons. Could be a good short-term workaround. We specifically use it to define string-based dialplan IDs for use in the script, just like this. [1] https://github.com/OpenSIPS/opensips/issues/new?assignees=&labels=&template=feature_request.md&title=%5BFEATURE%5D Ben Newlin From: Users on behalf of Kingsley Tart Date: Tuesday, April 20, 2021 at 12:22 PM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] dialplan module - dpid - could it be a string? I love the dialplan module and how the rules can be reloaded from a DB. However, is there any reason why the dpid can't be a string instead of an integer? (ie the value that's stored in the dialplan.dpid field). I think it would be handy to be able to apply names to dialplan rulesets as it would be more intuitive. If dpid could be a string, then I could for example create a ruleset that I could refer to as "uk_local_to_e164" instead of "7". Could this be a thing? -- Cheers, Kingsley. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From mwbazdarich at yahoo.com Tue Apr 20 18:15:02 2021 From: mwbazdarich at yahoo.com (mwb) Date: Tue, 20 Apr 2021 18:15:02 +0000 (UTC) Subject: [OpenSIPS-Users] Removing INVITE Contact from Reply (2.1.5) In-Reply-To: <1741851750.4222838.1618905914682@mail.yahoo.com> References: <1741851750.4222838.1618905914682.ref@mail.yahoo.com> <1741851750.4222838.1618905914682@mail.yahoo.com> Message-ID: <1947798476.4451574.1618942502082@mail.yahoo.com> Solved / Follow-up:We found that $ds was being populated in a previous route by way of an $rU assignment. Once we moved that assignment to an area specifically for "scenario #1" (sending the call on to an LNP provider) we found that $ds no longer had the INVITE RURI info populated and thus we could respond only with the URIs that we added in via branch adds. On Tuesday, April 20, 2021, 01:05:14 AM PDT, mwb wrote: Hello,I posed a questions a few months ago and received some helpful advice from a user here but I'm having an issue implementing it properly. We have a redirect server using OpenSIPS 2.1.5 that does LNP dipping with a little bit of routing as well.  There are two scenarios: 1. There is No RN/NPDI in the RURI (so the redirect server sends a LNP request to an external LNP server 2. The INVITE has RN/NPDI in the RURI so no LNP dip is dispatched. In the second scenario we do a few operations and generate URIs to embed in a 302 reply using the send_reply function in a . The issue we are running into is that it appears the $ds variable is being populated with the INVITE Request URI (RN and/or NPDI) info and it is being sent as the "top" Contact header in the 302 response.  I have tried a few different things, including "remove_branch(0)" but they don't seem to be able to clobber the value in the $ds variable. It also seems impossible to use $ds as an LVALUE to try and overwrite it (this is consistent with the documentation which says it is not r/w). Is there a way to: 1. Clobber the $ds variable?2. Preclude the $ds variable from being inserted as a Contact header in the reply?3. Keep the $ds variable from being populated when the initial INVITE is received. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From apsaras at microbase.gr Tue Apr 20 20:13:11 2021 From: apsaras at microbase.gr (Antonis Psaras) Date: Tue, 20 Apr 2021 23:13:11 +0300 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> Message-ID: <1e9501d73621$96ae3960$c40aac20$@microbase.gr> I did the following if (t_check_status("180")) { t_cancel_branch(); drop; } But there is an issue. When 180 is followed by 200 instantly, the CANCEL is not working as expected. When I add a delay on Answer ie 1sec then CANCEL works. Any suggestion? Antonis Psaras -----Original Message----- From: Users On Behalf Of Kingsley Tart Sent: Τρίτη, 20 Απριλίου 2021 20:10 To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 Firstly, I'm new to OpenSIPS so treat my comments accordingly. But, can you do something in an onreply route? eg, in a test setup I have, when I get an INVITE I do this: create_dialog("pPB"); t_on_reply("doodle"); (I can't remember whether the dialog is needed for this) and then I have this: onreply_route[doodle] { # expect $T_reply_code to likely first be 100 # then 180 or 183 for a progressing call # 200 when call is answered # or failure code (eg 4xx) or whatever if (t_check_status("^1[0-9][0-9]$")) { switch ($T_reply_code) { case 180: $acc_extra(t_ringing) = $Ts; break; case 183: $acc_extra(t_progress) = $Ts; break; } } else if (t_check_status("^2[0-9][0-9]$")) { $acc_extra(t_answer) = $Ts; } else { xlog("Something else\n"); } } so when a 180 is received, it calls the above route function. Could you send a CANCEL from there? Cheers, Kingsley. On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: > Dear all > > I am trying to create a service which will generate missed calls. In > order to be more accurate, I want to CANCEL the request when 180 is > received. > > The scenario is the following > > Asterisk Invite -> OpenSIPs -> Carrier > > Carrier 183 -> OpenSIPs -> Asterisk > > Carrier 180 -> OpenSIPs > > OpenSIPs Cancel -> Carrier > > > Is that possible to be done from script without external app? > > Regards > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From calvin.ellison at voxox.com Tue Apr 20 20:58:19 2021 From: calvin.ellison at voxox.com (Calvin Ellison) Date: Tue, 20 Apr 2021 13:58:19 -0700 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <1e9501d73621$96ae3960$c40aac20$@microbase.gr> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> <1e9501d73621$96ae3960$c40aac20$@microbase.gr> Message-ID: Once you receive 200 it is too late to CANCEL. You will need to ACK the 200 and then BYE the call. Regards, *Calvin Ellison* Systems Architect calvin.ellison at voxox.com +1 (213) 285-0555 ----------------------------------------------- *voxox.com * 5825 Oberlin Drive, Suite 5 San Diego, CA 92121 [image: Voxox] On Tue, Apr 20, 2021 at 1:14 PM Antonis Psaras wrote: > I did the following > > if (t_check_status("180")) > { > t_cancel_branch(); > drop; > } > > But there is an issue. > > When 180 is followed by 200 instantly, the CANCEL is not working as > expected. > > When I add a delay on Answer ie 1sec then CANCEL works. > > Any suggestion? > > Antonis Psaras > > -----Original Message----- > From: Users On Behalf Of Kingsley Tart > Sent: Τρίτη, 20 Απριλίου 2021 20:10 > To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 > > Firstly, I'm new to OpenSIPS so treat my comments accordingly. > > But, can you do something in an onreply route? > > eg, in a test setup I have, when I get an INVITE I do this: > > create_dialog("pPB"); > t_on_reply("doodle"); > > (I can't remember whether the dialog is needed for this) > > and then I have this: > > onreply_route[doodle] { > # expect $T_reply_code to likely first be 100 > # then 180 or 183 for a progressing call > # 200 when call is answered > # or failure code (eg 4xx) or whatever > if (t_check_status("^1[0-9][0-9]$")) { > switch ($T_reply_code) { > case 180: $acc_extra(t_ringing) = $Ts; break; > case 183: $acc_extra(t_progress) = $Ts; break; > } > } else if (t_check_status("^2[0-9][0-9]$")) { > $acc_extra(t_answer) = $Ts; > } else { > xlog("Something else\n"); > } > } > > so when a 180 is received, it calls the above route function. Could you > send a CANCEL from there? > > Cheers, > Kingsley. > > On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: > > Dear all > > > > I am trying to create a service which will generate missed calls. In > > order to be more accurate, I want to CANCEL the request when 180 is > > received. > > > > The scenario is the following > > > > Asterisk Invite -> OpenSIPs -> Carrier > > > > Carrier 183 -> OpenSIPs -> Asterisk > > > > Carrier 180 -> OpenSIPs > > > > OpenSIPs Cancel -> Carrier > > > > > > Is that possible to be done from script without external app? > > > > Regards > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Wed Apr 21 07:16:46 2021 From: Johan at democon.be (Johan De Clercq) Date: Wed, 21 Apr 2021 09:16:46 +0200 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> <1e9501d73621$96ae3960$c40aac20$@microbase.gr> Message-ID: Certain providers do this all the time : they bill faulty numbers So they give 180 followed by instant 200 and then they play : "the number that you tried to reach ....". Actually, Antonis, you should try a few of those numbers manually, I bet you that they are all faulty. Op di 20 apr. 2021 om 23:02 schreef Calvin Ellison : > Once you receive 200 it is too late to CANCEL. You will need to ACK the > 200 and then BYE the call. > > > Regards, > > *Calvin Ellison* > Systems Architect > calvin.ellison at voxox.com > +1 (213) 285-0555 > > ----------------------------------------------- > *voxox.com * > 5825 Oberlin Drive, Suite 5 > San Diego, CA 92121 > [image: Voxox] > > > On Tue, Apr 20, 2021 at 1:14 PM Antonis Psaras > wrote: > >> I did the following >> >> if (t_check_status("180")) >> { >> t_cancel_branch(); >> drop; >> } >> >> But there is an issue. >> >> When 180 is followed by 200 instantly, the CANCEL is not working as >> expected. >> >> When I add a delay on Answer ie 1sec then CANCEL works. >> >> Any suggestion? >> >> Antonis Psaras >> >> -----Original Message----- >> From: Users On Behalf Of Kingsley Tart >> Sent: Τρίτη, 20 Απριλίου 2021 20:10 >> To: OpenSIPS users mailling list >> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 >> >> Firstly, I'm new to OpenSIPS so treat my comments accordingly. >> >> But, can you do something in an onreply route? >> >> eg, in a test setup I have, when I get an INVITE I do this: >> >> create_dialog("pPB"); >> t_on_reply("doodle"); >> >> (I can't remember whether the dialog is needed for this) >> >> and then I have this: >> >> onreply_route[doodle] { >> # expect $T_reply_code to likely first be 100 >> # then 180 or 183 for a progressing call >> # 200 when call is answered >> # or failure code (eg 4xx) or whatever >> if (t_check_status("^1[0-9][0-9]$")) { >> switch ($T_reply_code) { >> case 180: $acc_extra(t_ringing) = $Ts; break; >> case 183: $acc_extra(t_progress) = $Ts; break; >> } >> } else if (t_check_status("^2[0-9][0-9]$")) { >> $acc_extra(t_answer) = $Ts; >> } else { >> xlog("Something else\n"); >> } >> } >> >> so when a 180 is received, it calls the above route function. Could you >> send a CANCEL from there? >> >> Cheers, >> Kingsley. >> >> On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: >> > Dear all >> > >> > I am trying to create a service which will generate missed calls. In >> > order to be more accurate, I want to CANCEL the request when 180 is >> > received. >> > >> > The scenario is the following >> > >> > Asterisk Invite -> OpenSIPs -> Carrier >> > >> > Carrier 183 -> OpenSIPs -> Asterisk >> > >> > Carrier 180 -> OpenSIPs >> > >> > OpenSIPs Cancel -> Carrier >> > >> > >> > Is that possible to be done from script without external app? >> > >> > Regards >> > >> > _______________________________________________ >> > Users mailing list >> > Users at lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 21 07:58:11 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 21 Apr 2021 10:58:11 +0300 Subject: [OpenSIPS-Users] dialplan module - dpid - could it be a string? In-Reply-To: References: <67751808587688838f3a12f60fc9c86bf1a1d95c.camel@dns99.co.uk> Message-ID: <3aebd202-e926-c87c-eabc-7ca7b535e037@opensips.org> Indeed, it may be an interesting one. And as Ben said, we are also using M4 for the scripting, nevertheless, if you want to integrate with OpenSIPS Control Panel, it may become tricky ;) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/20/21 9:05 PM, Ben Newlin wrote: > > This sounds like it might be a good feature request [1] and I agree it > sounds nice. I can’t offer any assistance in that direction at this > time, but I wanted to say that we use M4 in our config scripts to > allow us to define string constants like this and make the script more > readable, among other reasons. Could be a good short-term workaround. > We specifically use it to define string-based dialplan IDs for use in > the script, just like this. > > [1] > https://github.com/OpenSIPS/opensips/issues/new?assignees=&labels=&template=feature_request.md&title=%5BFEATURE%5D > > Ben Newlin > > *From: *Users on behalf of Kingsley > Tart > *Date: *Tuesday, April 20, 2021 at 12:22 PM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] dialplan module - dpid - could it be a string? > > I love the dialplan module and how the rules can be reloaded from a DB. > > However, is there any reason why the dpid can't be a string instead of > an integer? (ie the value that's stored in the dialplan.dpid field). > > I think it would be handy to be able to apply names to dialplan > rulesets as it would be more intuitive. If dpid could be a string, then > I could for example create a ruleset that I could refer to as > "uk_local_to_e164" instead of "7". > > Could this be a thing? > > -- > Cheers, > Kingsley. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From kingsley at dns99.co.uk Wed Apr 21 09:07:40 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Wed, 21 Apr 2021 10:07:40 +0100 Subject: [OpenSIPS-Users] dialplan module - dpid - could it be a string? In-Reply-To: <3aebd202-e926-c87c-eabc-7ca7b535e037@opensips.org> References: <67751808587688838f3a12f60fc9c86bf1a1d95c.camel@dns99.co.uk> <3aebd202-e926-c87c-eabc-7ca7b535e037@opensips.org> Message-ID: I've added it as a feature request: https://github.com/OpenSIPS/opensips/issues/2477 Cheers, Kingsley. On Wed, 2021-04-21 at 10:58 +0300, Bogdan-Andrei Iancu wrote: > Indeed, it may be an interesting one. And as Ben said, we are also > using M4 for the scripting, nevertheless, if you want to integrate > with OpenSIPS Control Panel, it may become tricky ;) > > Regards, > On 4/20/21 9:05 PM, Ben Newlin wrote: > > This sounds like it might be a good feature request [1] and I agree > > it sounds nice. I can’t offer any assistance in that direction at > > this time, but I wanted to say that we use M4 in our config scripts > > to allow us to define string constants like this and make the > > script more readable, among other reasons. Could be a good short- > > term workaround. We specifically use it to define string-based > > dialplan IDs for use in the script, just like this. > > > > [1] > > https://github.com/OpenSIPS/opensips/issues/new?assignees=&labels=&template=feature_request.md&title=%5BFEATURE%5D > > > > Ben Newlin > > > > From: Users on behalf of > > Kingsley Tart > > Date: Tuesday, April 20, 2021 at 12:22 PM > > To: OpenSIPS users mailling list > > Subject: [OpenSIPS-Users] dialplan module - dpid - could it be a > > string? > > > > I love the dialplan module and how the rules can be reloaded from a > > DB. > > > > However, is there any reason why the dpid can't be a string instead > > of > > an integer? (ie the value that's stored in the dialplan.dpid > > field). > > > > I think it would be handy to be able to apply names to dialplan > > rulesets as it would be more intuitive. If dpid could be a string, > > then > > I could for example create a ruleset that I could refer to as > > "uk_local_to_e164" instead of "7". > > > > Could this be a thing? > > > > -- > > Cheers, > > Kingsley. > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Wed Apr 21 09:29:37 2021 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 21 Apr 2021 12:29:37 +0300 Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 3.1.2 and 2.4.10 minor releases planning Message-ID: Hi, Everyone! I am glad to announce you that we have scheduled two new minor releases of our OpenSIPS stable branches. Therefore, on 6th of May 2021 we will release OpenSIPS 3.1.2 and 2.4.10! These new releases contain the latest bug fixes, and are fully compatible with previous versions. In order to increase releases' stability, we decided to freeze the code for two weeks - during this period the community is encouraged to fully test the upcoming releases and report any urgent matters, that we can eventually get fixed by the time of the release. The freeze starts tomorrow, 22nd of April 2021 and will last till the time of release, 6th of May 2021. Happy hacking! -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From bogdan at opensips.org Wed Apr 21 09:31:39 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 21 Apr 2021 12:31:39 +0300 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <1e9501d73621$96ae3960$c40aac20$@microbase.gr> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> <1e9501d73621$96ae3960$c40aac20$@microbase.gr> Message-ID: Hi Antonis, What exactly does not work ? sending the CANCEL out? or the callee "refuses" to cancel and sends a 200 OK ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/20/21 11:13 PM, Antonis Psaras wrote: > I did the following > > if (t_check_status("180")) > { > t_cancel_branch(); > drop; > } > > But there is an issue. > > When 180 is followed by 200 instantly, the CANCEL is not working as expected. > > When I add a delay on Answer ie 1sec then CANCEL works. > > Any suggestion? > > Antonis Psaras > > -----Original Message----- > From: Users On Behalf Of Kingsley Tart > Sent: Τρίτη, 20 Απριλίου 2021 20:10 > To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 > > Firstly, I'm new to OpenSIPS so treat my comments accordingly. > > But, can you do something in an onreply route? > > eg, in a test setup I have, when I get an INVITE I do this: > > create_dialog("pPB"); > t_on_reply("doodle"); > > (I can't remember whether the dialog is needed for this) > > and then I have this: > > onreply_route[doodle] { > # expect $T_reply_code to likely first be 100 > # then 180 or 183 for a progressing call > # 200 when call is answered > # or failure code (eg 4xx) or whatever > if (t_check_status("^1[0-9][0-9]$")) { > switch ($T_reply_code) { > case 180: $acc_extra(t_ringing) = $Ts; break; > case 183: $acc_extra(t_progress) = $Ts; break; > } > } else if (t_check_status("^2[0-9][0-9]$")) { > $acc_extra(t_answer) = $Ts; > } else { > xlog("Something else\n"); > } > } > > so when a 180 is received, it calls the above route function. Could you send a CANCEL from there? > > Cheers, > Kingsley. > > On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: >> Dear all >> >> I am trying to create a service which will generate missed calls. In >> order to be more accurate, I want to CANCEL the request when 180 is >> received. >> >> The scenario is the following >> >> Asterisk Invite -> OpenSIPs -> Carrier >> >> Carrier 183 -> OpenSIPs -> Asterisk >> >> Carrier 180 -> OpenSIPs >> >> OpenSIPs Cancel -> Carrier >> >> >> Is that possible to be done from script without external app? >> >> Regards >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From gmaruzz at gmail.com Wed Apr 21 09:39:30 2021 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 21 Apr 2021 11:39:30 +0200 Subject: [OpenSIPS-Users] [ANNOUNCE]: sipnagios, a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc) Message-ID: Hello fellow VoIPers and RTCers, on GitHub there is an early release of sipnagios, opensource. check it out: https://github.com/gmaruzz/sipnagios sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc) sipnagios implements the Nagios plugin API for monitoring and performance data. sipnagios.c is a modification of the original siprtp.c sample in pjproject distribution. Supposedly, it works on Linux, Windows, and anywhere you can compile pjproject on. It makes a call, checks all the various resulting values (mos, rtt, pdd, tta, jitter, packet loss, bytes and packets transferred, and so on). It verifies these values are included into acceptable, warning, or critical ranges. If the call has gone well, sipnagios print performance data for Nagios graphs, and returns 0. If the call fails, or if its measured values are not inside acceptable ranges, it exits with Nagios conventional WARNING or CRITICAL values. mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on GitHub too) I can't even understand :) (merci Julien!) Enjoy! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From apsaras at microbase.gr Wed Apr 21 09:46:12 2021 From: apsaras at microbase.gr (Antonis Psaras) Date: Wed, 21 Apr 2021 12:46:12 +0300 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> <1e9501d73621$96ae3960$c40aac20$@microbase.gr> Message-ID: <220801d73693$2a4a3e00$7edeba00$@microbase.gr> Hello Bogdan The flow is the following INVITE Trying (instantly) (after 2sec) 183 (after 3sec) 180 (here I do the process but during that I receive) 200 (instantly) (here the CANCEL is sent) CANCEL And the call is keep going, hence the CANCEL was ignored by the carrier by the carrier because the call was answered. Regards Antonis Psaras -----Original Message----- From: Bogdan-Andrei Iancu Sent: Τετάρτη, 21 Απριλίου 2021 12:32 To: apsaras at microbase.gr; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 Hi Antonis, What exactly does not work ? sending the CANCEL out? or the callee "refuses" to cancel and sends a 200 OK ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/20/21 11:13 PM, Antonis Psaras wrote: > I did the following > > if (t_check_status("180")) > { > t_cancel_branch(); > drop; > } > > But there is an issue. > > When 180 is followed by 200 instantly, the CANCEL is not working as expected. > > When I add a delay on Answer ie 1sec then CANCEL works. > > Any suggestion? > > Antonis Psaras > > -----Original Message----- > From: Users On Behalf Of Kingsley > Tart > Sent: Τρίτη, 20 Απριλίου 2021 20:10 > To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 > > Firstly, I'm new to OpenSIPS so treat my comments accordingly. > > But, can you do something in an onreply route? > > eg, in a test setup I have, when I get an INVITE I do this: > > create_dialog("pPB"); > t_on_reply("doodle"); > > (I can't remember whether the dialog is needed for this) > > and then I have this: > > onreply_route[doodle] { > # expect $T_reply_code to likely first be 100 > # then 180 or 183 for a progressing call > # 200 when call is answered > # or failure code (eg 4xx) or whatever > if (t_check_status("^1[0-9][0-9]$")) { > switch ($T_reply_code) { > case 180: $acc_extra(t_ringing) = $Ts; break; > case 183: $acc_extra(t_progress) = $Ts; break; > } > } else if (t_check_status("^2[0-9][0-9]$")) { > $acc_extra(t_answer) = $Ts; > } else { > xlog("Something else\n"); > } > } > > so when a 180 is received, it calls the above route function. Could you send a CANCEL from there? > > Cheers, > Kingsley. > > On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: >> Dear all >> >> I am trying to create a service which will generate missed calls. In >> order to be more accurate, I want to CANCEL the request when 180 is >> received. >> >> The scenario is the following >> >> Asterisk Invite -> OpenSIPs -> Carrier >> >> Carrier 183 -> OpenSIPs -> Asterisk >> >> Carrier 180 -> OpenSIPs >> >> OpenSIPs Cancel -> Carrier >> >> >> Is that possible to be done from script without external app? >> >> Regards >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Wed Apr 21 09:47:04 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 21 Apr 2021 12:47:04 +0300 Subject: [OpenSIPS-Users] dialplan module - ensuring a substring exists in URI In-Reply-To: <980007766d975b094a3052582ad1eea5dec480fb.camel@dns99.co.uk> References: <980007766d975b094a3052582ad1eea5dec480fb.camel@dns99.co.uk> Message-ID: <5877f030-c2a1-9e5b-827e-e33b7907b2b4@opensips.org> Hi Kingsley, You could try with a single rule, like optional matching the param (like "(;user=phone)?") and in subst + repl part do not include the param inside the \1 parenthesis, having it added all the time in the repl part. Still you need a more complex matching if you have multiple params, not only the "user" one. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/20/21 8:16 PM, Kingsley Tart wrote: > Hi, > > I want a DP ruleset to ensure that ";user=phone" is in the SIP URI, ie > if it is not there, add it, but if it is already there, do nothing. > > I've managed to do this with two lines, but is there a more efficient > way? > > This is what I have in the dialplan table in this ruleset? > > +----+----------+-------------+-------------+-----------+---------------+ > | pr | match_op | match_exp | match_flags | subst_exp | repl_exp | > +----+----------+-------------+-------------+-----------+---------------+ > | 1 | 1 | ;user=phone | 1 | (.+) | \1 | > | 2 | 1 | . | 1 | (.+) | \1;user=phone | > +----+----------+-------------+-------------+-----------+---------------+ > From bogdan at opensips.org Wed Apr 21 09:54:55 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 21 Apr 2021 12:54:55 +0300 Subject: [OpenSIPS-Users] how can i combine signaling and RTP from rtpproxy In-Reply-To: References: Message-ID: <0f713188-9ec9-c899-d39f-7ae953f5f55d@opensips.org> https://www.wireshark.org/docs/wsug_html_chunked/ChIOMergeSection.html ?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/20/21 11:15 AM, Miha via Users wrote: > Hello > > due to debugging i would like to combine cap from opensips and also > cap from rtpproxy (they are on different servers) so that I can check > if RTP is missing for certain call. > > > Can you help me with solving this issue :) > > thank you > miha > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 21 10:03:56 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 21 Apr 2021 13:03:56 +0300 Subject: [OpenSIPS-Users] [ANNOUNCE]: sipnagios, a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc) In-Reply-To: References: Message-ID: Really nice Giovanni, thanks for the info ! Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/21/21 12:39 PM, Giovanni Maruzzelli wrote: > Hello fellow VoIPers and RTCers, > > on GitHub there is an early release of sipnagios, opensource. > > check it out: https://github.com/gmaruzz/sipnagios > > sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP > (compatible with checkmk, etc) > > sipnagios implements the Nagios plugin API for monitoring and > performance data. > > sipnagios.c is a modification of the original siprtp.c sample in > pjproject distribution. Supposedly, it works on Linux, Windows, and > anywhere you can compile pjproject on. > > It makes a call, checks all the various resulting values (mos, rtt, > pdd, tta, jitter, packet loss, bytes and packets transferred, and so > on). It verifies these values are included into acceptable, warning, > or critical ranges. > > If the call has gone well, sipnagios print performance data for Nagios > graphs, and returns 0. > > If the call fails, or if its measured values are not inside acceptable > ranges, it exits with Nagios conventional WARNING or CRITICAL values. > > mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on > GitHub too) I can't even understand :) (merci Julien!) > > Enjoy! > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 21 10:20:03 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 21 Apr 2021 13:20:03 +0300 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <220801d73693$2a4a3e00$7edeba00$@microbase.gr> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> <1e9501d73621$96ae3960$c40aac20$@microbase.gr> <220801d73693$2a4a3e00$7edeba00$@microbase.gr> Message-ID: <5e296435-5341-10fa-cb88-dcf102b2dbe4@opensips.org> So there is a race between the CANCEL you send and the incoming 200 OK, right ? And does the carrier even reply to that CANCEL? the RFC3261 says it is a must to provide a reply to any incoming request. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/21/21 12:46 PM, Antonis Psaras wrote: > Hello Bogdan > > The flow is the following > > INVITE > Trying (instantly) > (after 2sec) > 183 > (after 3sec) > 180 > (here I do the process but during that I receive) > 200 (instantly) > (here the CANCEL is sent) > CANCEL > > And the call is keep going, hence the CANCEL was ignored by the carrier by the carrier because the call was answered. > > Regards > > > Antonis Psaras > > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: Τετάρτη, 21 Απριλίου 2021 12:32 > To: apsaras at microbase.gr; OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 > > Hi Antonis, > > What exactly does not work ? sending the CANCEL out? or the callee "refuses" to cancel and sends a 200 OK ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/20/21 11:13 PM, Antonis Psaras wrote: >> I did the following >> >> if (t_check_status("180")) >> { >> t_cancel_branch(); >> drop; >> } >> >> But there is an issue. >> >> When 180 is followed by 200 instantly, the CANCEL is not working as expected. >> >> When I add a delay on Answer ie 1sec then CANCEL works. >> >> Any suggestion? >> >> Antonis Psaras >> >> -----Original Message----- >> From: Users On Behalf Of Kingsley >> Tart >> Sent: Τρίτη, 20 Απριλίου 2021 20:10 >> To: OpenSIPS users mailling list >> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 >> >> Firstly, I'm new to OpenSIPS so treat my comments accordingly. >> >> But, can you do something in an onreply route? >> >> eg, in a test setup I have, when I get an INVITE I do this: >> >> create_dialog("pPB"); >> t_on_reply("doodle"); >> >> (I can't remember whether the dialog is needed for this) >> >> and then I have this: >> >> onreply_route[doodle] { >> # expect $T_reply_code to likely first be 100 >> # then 180 or 183 for a progressing call >> # 200 when call is answered >> # or failure code (eg 4xx) or whatever >> if (t_check_status("^1[0-9][0-9]$")) { >> switch ($T_reply_code) { >> case 180: $acc_extra(t_ringing) = $Ts; break; >> case 183: $acc_extra(t_progress) = $Ts; break; >> } >> } else if (t_check_status("^2[0-9][0-9]$")) { >> $acc_extra(t_answer) = $Ts; >> } else { >> xlog("Something else\n"); >> } >> } >> >> so when a 180 is received, it calls the above route function. Could you send a CANCEL from there? >> >> Cheers, >> Kingsley. >> >> On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: >>> Dear all >>> >>> I am trying to create a service which will generate missed calls. In >>> order to be more accurate, I want to CANCEL the request when 180 is >>> received. >>> >>> The scenario is the following >>> >>> Asterisk Invite -> OpenSIPs -> Carrier >>> >>> Carrier 183 -> OpenSIPs -> Asterisk >>> >>> Carrier 180 -> OpenSIPs >>> >>> OpenSIPs Cancel -> Carrier >>> >>> >>> Is that possible to be done from script without external app? >>> >>> Regards >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From apsaras at microbase.gr Wed Apr 21 10:47:03 2021 From: apsaras at microbase.gr (Antonis Psaras) Date: Wed, 21 Apr 2021 13:47:03 +0300 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <5e296435-5341-10fa-cb88-dcf102b2dbe4@opensips.org> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> <1e9501d73621$96ae3960$c40aac20$@microbase.gr> <220801d73693$2a4a3e00$7edeba00$@microbase.gr> <5e296435-5341-10fa-cb88-dcf102b2dbe4@opensips.org> Message-ID: <22bb01d7369b$aaf9d710$00ed8530$@microbase.gr> No. there is no reply. Just ignored. Antonis Psaras / Managing Director -----Original Message----- From: Bogdan-Andrei Iancu Sent: Τετάρτη, 21 Απριλίου 2021 13:20 To: apsaras at microbase.gr; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 So there is a race between the CANCEL you send and the incoming 200 OK, right ? And does the carrier even reply to that CANCEL? the RFC3261 says it is a must to provide a reply to any incoming request. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/21/21 12:46 PM, Antonis Psaras wrote: > Hello Bogdan > > The flow is the following > > INVITE > Trying (instantly) > (after 2sec) > 183 > (after 3sec) > 180 > (here I do the process but during that I receive) > 200 (instantly) > (here the CANCEL is sent) > CANCEL > > And the call is keep going, hence the CANCEL was ignored by the carrier by the carrier because the call was answered. > > Regards > > > Antonis Psaras > > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: Τετάρτη, 21 Απριλίου 2021 12:32 > To: apsaras at microbase.gr; OpenSIPS users mailling list > > Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 > > Hi Antonis, > > What exactly does not work ? sending the CANCEL out? or the callee "refuses" to cancel and sends a 200 OK ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/20/21 11:13 PM, Antonis Psaras wrote: >> I did the following >> >> if (t_check_status("180")) >> { >> t_cancel_branch(); >> drop; >> } >> >> But there is an issue. >> >> When 180 is followed by 200 instantly, the CANCEL is not working as expected. >> >> When I add a delay on Answer ie 1sec then CANCEL works. >> >> Any suggestion? >> >> Antonis Psaras >> >> -----Original Message----- >> From: Users On Behalf Of Kingsley >> Tart >> Sent: Τρίτη, 20 Απριλίου 2021 20:10 >> To: OpenSIPS users mailling list >> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 >> >> Firstly, I'm new to OpenSIPS so treat my comments accordingly. >> >> But, can you do something in an onreply route? >> >> eg, in a test setup I have, when I get an INVITE I do this: >> >> create_dialog("pPB"); >> t_on_reply("doodle"); >> >> (I can't remember whether the dialog is needed for this) >> >> and then I have this: >> >> onreply_route[doodle] { >> # expect $T_reply_code to likely first be 100 >> # then 180 or 183 for a progressing call >> # 200 when call is answered >> # or failure code (eg 4xx) or whatever >> if (t_check_status("^1[0-9][0-9]$")) { >> switch ($T_reply_code) { >> case 180: $acc_extra(t_ringing) = $Ts; break; >> case 183: $acc_extra(t_progress) = $Ts; break; >> } >> } else if (t_check_status("^2[0-9][0-9]$")) { >> $acc_extra(t_answer) = $Ts; >> } else { >> xlog("Something else\n"); >> } >> } >> >> so when a 180 is received, it calls the above route function. Could you send a CANCEL from there? >> >> Cheers, >> Kingsley. >> >> On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: >>> Dear all >>> >>> I am trying to create a service which will generate missed calls. In >>> order to be more accurate, I want to CANCEL the request when 180 is >>> received. >>> >>> The scenario is the following >>> >>> Asterisk Invite -> OpenSIPs -> Carrier >>> >>> Carrier 183 -> OpenSIPs -> Asterisk >>> >>> Carrier 180 -> OpenSIPs >>> >>> OpenSIPs Cancel -> Carrier >>> >>> >>> Is that possible to be done from script without external app? >>> >>> Regards >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From bogdan at opensips.org Wed Apr 21 10:49:31 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 21 Apr 2021 13:49:31 +0300 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <22bb01d7369b$aaf9d710$00ed8530$@microbase.gr> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> <1e9501d73621$96ae3960$c40aac20$@microbase.gr> <220801d73693$2a4a3e00$7edeba00$@microbase.gr> <5e296435-5341-10fa-cb88-dcf102b2dbe4@opensips.org> <22bb01d7369b$aaf9d710$00ed8530$@microbase.gr> Message-ID: <67c263cd-e4b0-3e43-4492-b5d71dcc58ab@opensips.org> Well, you should bring that to the carrier :( Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/21/21 1:47 PM, Antonis Psaras wrote: > No. there is no reply. Just ignored. > > > Antonis Psaras / Managing Director > > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: Τετάρτη, 21 Απριλίου 2021 13:20 > To: apsaras at microbase.gr; 'OpenSIPS users mailling list' > Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 > > So there is a race between the CANCEL you send and the incoming 200 OK, right ? And does the carrier even reply to that CANCEL? the RFC3261 says it is a must to provide a reply to any incoming request. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/21/21 12:46 PM, Antonis Psaras wrote: >> Hello Bogdan >> >> The flow is the following >> >> INVITE >> Trying (instantly) >> (after 2sec) >> 183 >> (after 3sec) >> 180 >> (here I do the process but during that I receive) >> 200 (instantly) >> (here the CANCEL is sent) >> CANCEL >> >> And the call is keep going, hence the CANCEL was ignored by the carrier by the carrier because the call was answered. >> >> Regards >> >> >> Antonis Psaras >> >> -----Original Message----- >> From: Bogdan-Andrei Iancu >> Sent: Τετάρτη, 21 Απριλίου 2021 12:32 >> To: apsaras at microbase.gr; OpenSIPS users mailling list >> >> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 >> >> Hi Antonis, >> >> What exactly does not work ? sending the CANCEL out? or the callee "refuses" to cancel and sends a 200 OK ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp 2021 online >> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >> >> On 4/20/21 11:13 PM, Antonis Psaras wrote: >>> I did the following >>> >>> if (t_check_status("180")) >>> { >>> t_cancel_branch(); >>> drop; >>> } >>> >>> But there is an issue. >>> >>> When 180 is followed by 200 instantly, the CANCEL is not working as expected. >>> >>> When I add a delay on Answer ie 1sec then CANCEL works. >>> >>> Any suggestion? >>> >>> Antonis Psaras >>> >>> -----Original Message----- >>> From: Users On Behalf Of Kingsley >>> Tart >>> Sent: Τρίτη, 20 Απριλίου 2021 20:10 >>> To: OpenSIPS users mailling list >>> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 >>> >>> Firstly, I'm new to OpenSIPS so treat my comments accordingly. >>> >>> But, can you do something in an onreply route? >>> >>> eg, in a test setup I have, when I get an INVITE I do this: >>> >>> create_dialog("pPB"); >>> t_on_reply("doodle"); >>> >>> (I can't remember whether the dialog is needed for this) >>> >>> and then I have this: >>> >>> onreply_route[doodle] { >>> # expect $T_reply_code to likely first be 100 >>> # then 180 or 183 for a progressing call >>> # 200 when call is answered >>> # or failure code (eg 4xx) or whatever >>> if (t_check_status("^1[0-9][0-9]$")) { >>> switch ($T_reply_code) { >>> case 180: $acc_extra(t_ringing) = $Ts; break; >>> case 183: $acc_extra(t_progress) = $Ts; break; >>> } >>> } else if (t_check_status("^2[0-9][0-9]$")) { >>> $acc_extra(t_answer) = $Ts; >>> } else { >>> xlog("Something else\n"); >>> } >>> } >>> >>> so when a 180 is received, it calls the above route function. Could you send a CANCEL from there? >>> >>> Cheers, >>> Kingsley. >>> >>> On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: >>>> Dear all >>>> >>>> I am trying to create a service which will generate missed calls. In >>>> order to be more accurate, I want to CANCEL the request when 180 is >>>> received. >>>> >>>> The scenario is the following >>>> >>>> Asterisk Invite -> OpenSIPs -> Carrier >>>> >>>> Carrier 183 -> OpenSIPs -> Asterisk >>>> >>>> Carrier 180 -> OpenSIPs >>>> >>>> OpenSIPs Cancel -> Carrier >>>> >>>> >>>> Is that possible to be done from script without external app? >>>> >>>> Regards >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From apsaras at microbase.gr Wed Apr 21 10:54:07 2021 From: apsaras at microbase.gr (Antonis Psaras) Date: Wed, 21 Apr 2021 13:54:07 +0300 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <67c263cd-e4b0-3e43-4492-b5d71dcc58ab@opensips.org> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> <1e9501d73621$96ae3960$c40aac20$@microbase.gr> <220801d73693$2a4a3e00$7edeba00$@microbase.gr> <5e296435-5341-10fa-cb88-dcf102b2dbe4@opensips.org> <22bb01d7369b$aaf9d710$00ed8530$@microbase.gr> <67c263cd-e4b0-3e43-4492-b5d71dcc58ab@opensips.org> Message-ID: <22da01d7369c$a6e8c630$f4ba5290$@microbase.gr> Thank you. I will. Thank you for your support Antonis Psaras -----Original Message----- From: Bogdan-Andrei Iancu Sent: Τετάρτη, 21 Απριλίου 2021 13:50 To: apsaras at microbase.gr; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 Well, you should bring that to the carrier :( Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/21/21 1:47 PM, Antonis Psaras wrote: > No. there is no reply. Just ignored. > > > Antonis Psaras / Managing Director > > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: Τετάρτη, 21 Απριλίου 2021 13:20 > To: apsaras at microbase.gr; 'OpenSIPS users mailling list' > > Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 > > So there is a race between the CANCEL you send and the incoming 200 OK, right ? And does the carrier even reply to that CANCEL? the RFC3261 says it is a must to provide a reply to any incoming request. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/21/21 12:46 PM, Antonis Psaras wrote: >> Hello Bogdan >> >> The flow is the following >> >> INVITE >> Trying (instantly) >> (after 2sec) >> 183 >> (after 3sec) >> 180 >> (here I do the process but during that I receive) >> 200 (instantly) >> (here the CANCEL is sent) >> CANCEL >> >> And the call is keep going, hence the CANCEL was ignored by the carrier by the carrier because the call was answered. >> >> Regards >> >> >> Antonis Psaras >> >> -----Original Message----- >> From: Bogdan-Andrei Iancu >> Sent: Τετάρτη, 21 Απριλίου 2021 12:32 >> To: apsaras at microbase.gr; OpenSIPS users mailling list >> >> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 >> >> Hi Antonis, >> >> What exactly does not work ? sending the CANCEL out? or the callee "refuses" to cancel and sends a 200 OK ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp 2021 online >> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >> >> On 4/20/21 11:13 PM, Antonis Psaras wrote: >>> I did the following >>> >>> if (t_check_status("180")) >>> { >>> t_cancel_branch(); >>> drop; >>> } >>> >>> But there is an issue. >>> >>> When 180 is followed by 200 instantly, the CANCEL is not working as expected. >>> >>> When I add a delay on Answer ie 1sec then CANCEL works. >>> >>> Any suggestion? >>> >>> Antonis Psaras >>> >>> -----Original Message----- >>> From: Users On Behalf Of Kingsley >>> Tart >>> Sent: Τρίτη, 20 Απριλίου 2021 20:10 >>> To: OpenSIPS users mailling list >>> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 >>> >>> Firstly, I'm new to OpenSIPS so treat my comments accordingly. >>> >>> But, can you do something in an onreply route? >>> >>> eg, in a test setup I have, when I get an INVITE I do this: >>> >>> create_dialog("pPB"); >>> t_on_reply("doodle"); >>> >>> (I can't remember whether the dialog is needed for this) >>> >>> and then I have this: >>> >>> onreply_route[doodle] { >>> # expect $T_reply_code to likely first be 100 >>> # then 180 or 183 for a progressing call >>> # 200 when call is answered >>> # or failure code (eg 4xx) or whatever >>> if (t_check_status("^1[0-9][0-9]$")) { >>> switch ($T_reply_code) { >>> case 180: $acc_extra(t_ringing) = $Ts; break; >>> case 183: $acc_extra(t_progress) = $Ts; break; >>> } >>> } else if (t_check_status("^2[0-9][0-9]$")) { >>> $acc_extra(t_answer) = $Ts; >>> } else { >>> xlog("Something else\n"); >>> } >>> } >>> >>> so when a 180 is received, it calls the above route function. Could you send a CANCEL from there? >>> >>> Cheers, >>> Kingsley. >>> >>> On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: >>>> Dear all >>>> >>>> I am trying to create a service which will generate missed calls. >>>> In order to be more accurate, I want to CANCEL the request when 180 >>>> is received. >>>> >>>> The scenario is the following >>>> >>>> Asterisk Invite -> OpenSIPs -> Carrier >>>> >>>> Carrier 183 -> OpenSIPs -> Asterisk >>>> >>>> Carrier 180 -> OpenSIPs >>>> >>>> OpenSIPs Cancel -> Carrier >>>> >>>> >>>> Is that possible to be done from script without external app? >>>> >>>> Regards >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From kingsley at dns99.co.uk Wed Apr 21 11:41:28 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Wed, 21 Apr 2021 12:41:28 +0100 Subject: [OpenSIPS-Users] dialplan module - ensuring a substring exists in URI In-Reply-To: <5877f030-c2a1-9e5b-827e-e33b7907b2b4@opensips.org> References: <980007766d975b094a3052582ad1eea5dec480fb.camel@dns99.co.uk> <5877f030-c2a1-9e5b-827e-e33b7907b2b4@opensips.org> Message-ID: Hi, Thanks but I tried that and whatever I tried I could not get it to work. I don't know whether I have misunderstood what you meant? With this setup: +------+----+----------+----------------+-------------+-------------+-------------+ | dpid | pr | match_op | match_exp | match_flags | subst_exp | repl_exp | +------+----+----------+----------------+-------------+-------------+-------------+ | 100 | 1 | 1 | (;user=phone)? | 1 | ;user=phone | ;user=phone | +------+----+----------+----------------+-------------+-------------+-------------+ I got this: # opensips-cli -x mi dp_reload "OK" # opensips-cli -x mi dp_translate 100 "+12345 at sip.com;food=beans" ERROR: command 'dp_translate' returned: 404: No translation # opensips-cli -x mi dp_translate 100 "+12345 at sip.com;user=phone" { "Output": ";user=phone", "ATTRIBUTES": "" } and with this setup: +------+----+----------+----------------+-------------+----------------+-------------+ | dpid | pr | match_op | match_exp | match_flags | subst_exp | repl_exp | +------+----+----------+----------------+-------------+----------------+-------------+ | 100 | 1 | 1 | (;user=phone)? | 1 | (;user=phone)? | ;user=phone | +------+----+----------+----------------+-------------+----------------+-------------+ I got this: # opensips-cli -x mi dp_reload "OK" # opensips-cli -x mi dp_translate 100 "+12345 at sip.com;food=beans" { "Output": ";user=phone", "ATTRIBUTES": "" } # opensips-cli -x mi dp_translate 100 "+12345 at sip.com;user=phone" { "Output": ";user=phone", "ATTRIBUTES": "" } but also realised that if it had matched, I would have no control over where in the URI it would have added the extra text, so I'm wondering that my two rule approach is perhaps the best way for what I'm doing anyway. Cheers, Kingsley. On Wed, 2021-04-21 at 12:47 +0300, Bogdan-Andrei Iancu wrote: > Hi Kingsley, > > You could try with a single rule, like optional matching the param (like > "(;user=phone)?") and in subst + repl part do not include the param > inside the \1 parenthesis, having it added all the time in the repl part. > > Still you need a more complex matching if you have multiple params, not > only the "user" one. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/20/21 8:16 PM, Kingsley Tart wrote: > > Hi, > > > > I want a DP ruleset to ensure that ";user=phone" is in the SIP URI, > > ie > > if it is not there, add it, but if it is already there, do nothing. > > > > I've managed to do this with two lines, but is there a more > > efficient > > way? > > > > This is what I have in the dialplan table in this ruleset? > > > > +----+----------+-------------+-------------+-----------+-------- > > -------+ > > > pr | match_op | match_exp | match_flags | subst_exp | > > > repl_exp | > > > > +----+----------+-------------+-------------+-----------+-------- > > -------+ > > > 1 | 1 | ;user=phone | 1 | (.+) | > > > \1 | > > > 2 | 1 | . | 1 | (.+) | > > > \1;user=phone | > > > > +----+----------+-------------+-------------+-----------+-------- > > -------+ > > > > From vladpaiu at opensips.org Wed Apr 21 12:45:06 2021 From: vladpaiu at opensips.org (Vlad Paiu) Date: Wed, 21 Apr 2021 15:45:06 +0300 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <220801d73693$2a4a3e00$7edeba00$@microbase.gr> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> <1e9501d73621$96ae3960$c40aac20$@microbase.gr> <220801d73693$2a4a3e00$7edeba00$@microbase.gr> Message-ID: <6ccbf1bf-b180-2eb3-6eb1-dacb1f7479aa@opensips.org> Hello, Try using the race_condition_timeout dialog param along with the 'E' flag when creating the dialog. https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout Best Regards, Vlad On 21.04.2021 12:46, Antonis Psaras wrote: > Hello Bogdan > > The flow is the following > > INVITE > Trying (instantly) > (after 2sec) > 183 > (after 3sec) > 180 > (here I do the process but during that I receive) > 200 (instantly) > (here the CANCEL is sent) > CANCEL > > And the call is keep going, hence the CANCEL was ignored by the carrier by the carrier because the call was answered. > > Regards > > > Antonis Psaras > > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: Τετάρτη, 21 Απριλίου 2021 12:32 > To: apsaras at microbase.gr; OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 > > Hi Antonis, > > What exactly does not work ? sending the CANCEL out? or the callee "refuses" to cancel and sends a 200 OK ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/20/21 11:13 PM, Antonis Psaras wrote: >> I did the following >> >> if (t_check_status("180")) >> { >> t_cancel_branch(); >> drop; >> } >> >> But there is an issue. >> >> When 180 is followed by 200 instantly, the CANCEL is not working as expected. >> >> When I add a delay on Answer ie 1sec then CANCEL works. >> >> Any suggestion? >> >> Antonis Psaras >> >> -----Original Message----- >> From: Users On Behalf Of Kingsley >> Tart >> Sent: Τρίτη, 20 Απριλίου 2021 20:10 >> To: OpenSIPS users mailling list >> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 >> >> Firstly, I'm new to OpenSIPS so treat my comments accordingly. >> >> But, can you do something in an onreply route? >> >> eg, in a test setup I have, when I get an INVITE I do this: >> >> create_dialog("pPB"); >> t_on_reply("doodle"); >> >> (I can't remember whether the dialog is needed for this) >> >> and then I have this: >> >> onreply_route[doodle] { >> # expect $T_reply_code to likely first be 100 >> # then 180 or 183 for a progressing call >> # 200 when call is answered >> # or failure code (eg 4xx) or whatever >> if (t_check_status("^1[0-9][0-9]$")) { >> switch ($T_reply_code) { >> case 180: $acc_extra(t_ringing) = $Ts; break; >> case 183: $acc_extra(t_progress) = $Ts; break; >> } >> } else if (t_check_status("^2[0-9][0-9]$")) { >> $acc_extra(t_answer) = $Ts; >> } else { >> xlog("Something else\n"); >> } >> } >> >> so when a 180 is received, it calls the above route function. Could you send a CANCEL from there? >> >> Cheers, >> Kingsley. >> >> On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: >>> Dear all >>> >>> I am trying to create a service which will generate missed calls. In >>> order to be more accurate, I want to CANCEL the request when 180 is >>> received. >>> >>> The scenario is the following >>> >>> Asterisk Invite -> OpenSIPs -> Carrier >>> >>> Carrier 183 -> OpenSIPs -> Asterisk >>> >>> Carrier 180 -> OpenSIPs >>> >>> OpenSIPs Cancel -> Carrier >>> >>> >>> Is that possible to be done from script without external app? >>> >>> Regards >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From apsaras at microbase.gr Wed Apr 21 12:52:15 2021 From: apsaras at microbase.gr (Antonis Psaras) Date: Wed, 21 Apr 2021 15:52:15 +0300 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <6ccbf1bf-b180-2eb3-6eb1-dacb1f7479aa@opensips.org> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> <1e9501d73621$96ae3960$c40aac20$@microbase.gr> <220801d73693$2a4a3e00$7edeba00$@microbase.gr> <6ccbf1bf-b180-2eb3-6eb1-dacb1f7479aa@opensips.org> Message-ID: <246101d736ad$27c2f810$7748e830$@microbase.gr> Hello Vlad I am using 2.4 so race_condition_timeout is not available but I can upgrade if required. Actually is not very clear to me what that parameter does. Can you explain a bit more? Regards Antonis Psaras / Managing Director -----Original Message----- From: Users On Behalf Of Vlad Paiu Sent: Τετάρτη, 21 Απριλίου 2021 15:45 To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 Hello, Try using the race_condition_timeout dialog param along with the 'E' flag when creating the dialog. https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout Best Regards, Vlad On 21.04.2021 12:46, Antonis Psaras wrote: > Hello Bogdan > > The flow is the following > > INVITE > Trying (instantly) > (after 2sec) > 183 > (after 3sec) > 180 > (here I do the process but during that I receive) > 200 (instantly) > (here the CANCEL is sent) > CANCEL > > And the call is keep going, hence the CANCEL was ignored by the carrier by the carrier because the call was answered. > > Regards > > > Antonis Psaras > > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: Τετάρτη, 21 Απριλίου 2021 12:32 > To: apsaras at microbase.gr; OpenSIPS users mailling list > > Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 > > Hi Antonis, > > What exactly does not work ? sending the CANCEL out? or the callee "refuses" to cancel and sends a 200 OK ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/20/21 11:13 PM, Antonis Psaras wrote: >> I did the following >> >> if (t_check_status("180")) >> { >> t_cancel_branch(); >> drop; >> } >> >> But there is an issue. >> >> When 180 is followed by 200 instantly, the CANCEL is not working as expected. >> >> When I add a delay on Answer ie 1sec then CANCEL works. >> >> Any suggestion? >> >> Antonis Psaras >> >> -----Original Message----- >> From: Users On Behalf Of Kingsley >> Tart >> Sent: Τρίτη, 20 Απριλίου 2021 20:10 >> To: OpenSIPS users mailling list >> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 >> >> Firstly, I'm new to OpenSIPS so treat my comments accordingly. >> >> But, can you do something in an onreply route? >> >> eg, in a test setup I have, when I get an INVITE I do this: >> >> create_dialog("pPB"); >> t_on_reply("doodle"); >> >> (I can't remember whether the dialog is needed for this) >> >> and then I have this: >> >> onreply_route[doodle] { >> # expect $T_reply_code to likely first be 100 >> # then 180 or 183 for a progressing call >> # 200 when call is answered >> # or failure code (eg 4xx) or whatever >> if (t_check_status("^1[0-9][0-9]$")) { >> switch ($T_reply_code) { >> case 180: $acc_extra(t_ringing) = $Ts; break; >> case 183: $acc_extra(t_progress) = $Ts; break; >> } >> } else if (t_check_status("^2[0-9][0-9]$")) { >> $acc_extra(t_answer) = $Ts; >> } else { >> xlog("Something else\n"); >> } >> } >> >> so when a 180 is received, it calls the above route function. Could you send a CANCEL from there? >> >> Cheers, >> Kingsley. >> >> On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: >>> Dear all >>> >>> I am trying to create a service which will generate missed calls. In >>> order to be more accurate, I want to CANCEL the request when 180 is >>> received. >>> >>> The scenario is the following >>> >>> Asterisk Invite -> OpenSIPs -> Carrier >>> >>> Carrier 183 -> OpenSIPs -> Asterisk >>> >>> Carrier 180 -> OpenSIPs >>> >>> OpenSIPs Cancel -> Carrier >>> >>> >>> Is that possible to be done from script without external app? >>> >>> Regards >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From vladpaiu at opensips.org Wed Apr 21 13:44:15 2021 From: vladpaiu at opensips.org (Vlad Paiu) Date: Wed, 21 Apr 2021 16:44:15 +0300 Subject: [OpenSIPS-Users] Generate CANCEL on 180 In-Reply-To: <246101d736ad$27c2f810$7748e830$@microbase.gr> References: <1cd001d735ec$cc3eca20$64bc5e60$@microbase.gr> <4d1dfc411a7f1881f3048792707f4e6088bb06e3.camel@dns99.co.uk> <1e9501d73621$96ae3960$c40aac20$@microbase.gr> <220801d73693$2a4a3e00$7edeba00$@microbase.gr> <6ccbf1bf-b180-2eb3-6eb1-dacb1f7479aa@opensips.org> <246101d736ad$27c2f810$7748e830$@microbase.gr> Message-ID: <3e5f7462-3832-411b-0a1b-835cc7af889e@opensips.org> Hello, In 3.1, create_dialog [1] supports passing it the 'E' flag which will lead to the call termination if a dialog race condition occurs. The dialog will be ended after race_condition_timeout [2] seconds. Some of the most frequent race conditions are documented in RFC 5407 [3], and the dialog module currently supports terminating the call in case races 3.1.2 and 3.1.3 from the RFC occur. [1] https://opensips.org/html/docs/modules/3.1.x/dialog.html#func_create_dialog [2] https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout [3] https://tools.ietf.org/html/rfc5407 On 21.04.2021 15:52, Antonis Psaras wrote: > Hello Vlad > > I am using 2.4 so race_condition_timeout is not available but I can upgrade if required. > > Actually is not very clear to me what that parameter does. Can you explain a bit more? > > Regards > > > Antonis Psaras / Managing Director > > > -----Original Message----- > From: Users On Behalf Of Vlad Paiu > Sent: Τετάρτη, 21 Απριλίου 2021 15:45 > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 > > Hello, > > Try using the race_condition_timeout dialog param along with the 'E' > flag when creating the dialog. > > https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout > > > Best Regards, > Vlad > On 21.04.2021 12:46, Antonis Psaras wrote: >> Hello Bogdan >> >> The flow is the following >> >> INVITE >> Trying (instantly) >> (after 2sec) >> 183 >> (after 3sec) >> 180 >> (here I do the process but during that I receive) >> 200 (instantly) >> (here the CANCEL is sent) >> CANCEL >> >> And the call is keep going, hence the CANCEL was ignored by the carrier by the carrier because the call was answered. >> >> Regards >> >> >> Antonis Psaras >> >> -----Original Message----- >> From: Bogdan-Andrei Iancu >> Sent: Τετάρτη, 21 Απριλίου 2021 12:32 >> To: apsaras at microbase.gr; OpenSIPS users mailling list >> >> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 >> >> Hi Antonis, >> >> What exactly does not work ? sending the CANCEL out? or the callee "refuses" to cancel and sends a 200 OK ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp 2021 online >> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >> >> On 4/20/21 11:13 PM, Antonis Psaras wrote: >>> I did the following >>> >>> if (t_check_status("180")) >>> { >>> t_cancel_branch(); >>> drop; >>> } >>> >>> But there is an issue. >>> >>> When 180 is followed by 200 instantly, the CANCEL is not working as expected. >>> >>> When I add a delay on Answer ie 1sec then CANCEL works. >>> >>> Any suggestion? >>> >>> Antonis Psaras >>> >>> -----Original Message----- >>> From: Users On Behalf Of Kingsley >>> Tart >>> Sent: Τρίτη, 20 Απριλίου 2021 20:10 >>> To: OpenSIPS users mailling list >>> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180 >>> >>> Firstly, I'm new to OpenSIPS so treat my comments accordingly. >>> >>> But, can you do something in an onreply route? >>> >>> eg, in a test setup I have, when I get an INVITE I do this: >>> >>> create_dialog("pPB"); >>> t_on_reply("doodle"); >>> >>> (I can't remember whether the dialog is needed for this) >>> >>> and then I have this: >>> >>> onreply_route[doodle] { >>> # expect $T_reply_code to likely first be 100 >>> # then 180 or 183 for a progressing call >>> # 200 when call is answered >>> # or failure code (eg 4xx) or whatever >>> if (t_check_status("^1[0-9][0-9]$")) { >>> switch ($T_reply_code) { >>> case 180: $acc_extra(t_ringing) = $Ts; break; >>> case 183: $acc_extra(t_progress) = $Ts; break; >>> } >>> } else if (t_check_status("^2[0-9][0-9]$")) { >>> $acc_extra(t_answer) = $Ts; >>> } else { >>> xlog("Something else\n"); >>> } >>> } >>> >>> so when a 180 is received, it calls the above route function. Could you send a CANCEL from there? >>> >>> Cheers, >>> Kingsley. >>> >>> On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote: >>>> Dear all >>>> >>>> I am trying to create a service which will generate missed calls. In >>>> order to be more accurate, I want to CANCEL the request when 180 is >>>> received. >>>> >>>> The scenario is the following >>>> >>>> Asterisk Invite -> OpenSIPs -> Carrier >>>> >>>> Carrier 183 -> OpenSIPs -> Asterisk >>>> >>>> Carrier 180 -> OpenSIPs >>>> >>>> OpenSIPs Cancel -> Carrier >>>> >>>> >>>> Is that possible to be done from script without external app? >>>> >>>> Regards >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From vladp at opensips.org Thu Apr 22 12:21:45 2021 From: vladp at opensips.org (Vlad Patrascu) Date: Thu, 22 Apr 2021 15:21:45 +0300 Subject: [OpenSIPS-Users] [Blog] Clustering Engine Improvements in OpenSIPS 3.2 Message-ID: Hi all, Take a look at the latest blog post [1] that presents the new Clustering support improvements brought by the upcoming OpenSIPS 3.2 release. [1] https://blog.opensips.org/2021/04/22/clustering-engine-improvements-in-opensips-3-2/ Enjoy! -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com From ag at ag-projects.com Thu Apr 22 14:15:23 2021 From: ag at ag-projects.com (Adrian Georgescu) Date: Thu, 22 Apr 2021 11:15:23 -0300 Subject: [OpenSIPS-Users] Usage of TLS require_cert and verify_cert settings Message-ID: <5D29DEF8-AFCE-419D-A84A-338FF45B23BD@ag-projects.com> Hello, I have a question. I have the following TLS scenarios: 1) A local user for a domain I own, connects to my server using TLS. If the domain is local, I will authenticate the user against my database and I do not care if the user has a certificate 2) A remote server, connects to my server using TLS and pretends that is domainX.com. In such case, the only way to verify that this is true is by requiring a certificate and verify it So there is a logic split between when to require and how to verify a certificate depending on the fact that we deal with a local user or a foreign domain. I would like to know if is possible to set require_cert and verify_cert on the fly, while routing packets, instead of configuring them statically per domain. Regards, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: From kingsley at dns99.co.uk Fri Apr 23 09:31:13 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Fri, 23 Apr 2021 10:31:13 +0100 Subject: [OpenSIPS-Users] redirect an already-established call to another host? Message-ID: <5c73685c6f5d155e4194b4b4f066caf9fbbaa2e5.camel@dns99.co.uk> Hi, Is it possible to redirect a call to another endpoint if the call and media path has already been established? We have a number of Asterisk boxes and have OpenSIPS forwarding calls to these using load balancing logic. One of the services these Asterisk boxes perform is conference calls, so all callers to a particular conference need to go to the same Asterisk box. The issue is that we cannot determine the conference by the number called. The user calls, gets connected to Asterisk and then uses DTMF to enter a conference number. It is only then that we know which conference it is. Currently, the Asterisk box then has to check a central database to see whether that conference has any other participants and if so which Asterisk box is hosting it. If on a different box, Asterisk has to make a SIP call to the box where the conference is, so the user's call is going through two Asterisk boxes by this point. It would be nice if there was a clean way for the Asterisk box to tell OpenSIPS which asterisk box to go to, and then have OpenSIPS forward the call there instead. Is this possible? Cheers, Kingsley. From bogdan at opensips.org Fri Apr 23 11:58:27 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 23 Apr 2021 14:58:27 +0300 Subject: [OpenSIPS-Users] redirect an already-established call to another host? In-Reply-To: <5c73685c6f5d155e4194b4b4f066caf9fbbaa2e5.camel@dns99.co.uk> References: <5c73685c6f5d155e4194b4b4f066caf9fbbaa2e5.camel@dns99.co.uk> Message-ID: Hi, An established call cannot be redirect, but it can be transferred to another SIP endpoint. See the SIP attended and unattended (blind) transfer. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/23/21 12:31 PM, Kingsley Tart wrote: > Hi, > > Is it possible to redirect a call to another endpoint if the call and > media path has already been established? > > We have a number of Asterisk boxes and have OpenSIPS forwarding calls > to these using load balancing logic. > > One of the services these Asterisk boxes perform is conference calls, > so all callers to a particular conference need to go to the same > Asterisk box. > > The issue is that we cannot determine the conference by the number > called. The user calls, gets connected to Asterisk and then uses DTMF > to enter a conference number. It is only then that we know which > conference it is. > > Currently, the Asterisk box then has to check a central database to see > whether that conference has any other participants and if so which > Asterisk box is hosting it. If on a different box, Asterisk has to make > a SIP call to the box where the conference is, so the user's call is > going through two Asterisk boxes by this point. > > It would be nice if there was a clean way for the Asterisk box to tell > OpenSIPS which asterisk box to go to, and then have OpenSIPS forward > the call there instead. > > Is this possible? > > Cheers, > Kingsley. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From kingsley at dns99.co.uk Fri Apr 23 12:16:03 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Fri, 23 Apr 2021 13:16:03 +0100 Subject: [OpenSIPS-Users] redirect an already-established call to another host? In-Reply-To: References: <5c73685c6f5d155e4194b4b4f066caf9fbbaa2e5.camel@dns99.co.uk> Message-ID: <19a1643b26899998b78410a0dc6bc75d9369719c.camel@dns99.co.uk> Thank you Bogdan. Will look at that :) Cheers, Kingsley. On Fri, 2021-04-23 at 14:58 +0300, Bogdan-Andrei Iancu wrote: > Hi, > > An established call cannot be redirect, but it can be transferred to > another SIP endpoint. See the SIP attended and unattended (blind) > transfer. > > Regards, > > Bogdan-Andrei Iancu From rob.dyck at telus.net Mon Apr 26 20:43:05 2021 From: rob.dyck at telus.net (Rob Dyck) Date: Mon, 26 Apr 2021 13:43:05 -0700 Subject: [OpenSIPS-Users] MQTT support for 3.2 Message-ID: <6009887.DvuYhMxLoT@blacky.mylan> i see that the roadmap shows MQTT support. I do not see it in the modules list. Has this been dropped? Rob From kingsley at dns99.co.uk Tue Apr 27 10:24:37 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Tue, 27 Apr 2021 11:24:37 +0100 Subject: [OpenSIPS-Users] xlog - whether to \n Message-ID: <404b29dae8e6811708931c76ed12336dee949ded.camel@dns99.co.uk> Hi, If I use xlog() to log a single line, it does what I expect even if I don't end the log string with \n, so what is the reason I see people putting \n at the end of xlog strings? -- Cheers, Kingsley. From bogdan at opensips.org Tue Apr 27 11:00:14 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 27 Apr 2021 14:00:14 +0300 Subject: [OpenSIPS-Users] xlog - whether to \n In-Reply-To: <404b29dae8e6811708931c76ed12336dee949ded.camel@dns99.co.uk> References: <404b29dae8e6811708931c76ed12336dee949ded.camel@dns99.co.uk> Message-ID: <53668e63-41c1-1498-eb39-c28aee283dc2@opensips.org> The '\n' at the end of the xlog lines is a must. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/27/21 1:24 PM, Kingsley Tart wrote: > Hi, > > If I use xlog() to log a single line, it does what I expect even if I > don't end the log string with \n, so what is the reason I see people > putting \n at the end of xlog strings? > From bogdan at opensips.org Tue Apr 27 11:02:07 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 27 Apr 2021 14:02:07 +0300 Subject: [OpenSIPS-Users] MQTT support for 3.2 In-Reply-To: <6009887.DvuYhMxLoT@blacky.mylan> References: <6009887.DvuYhMxLoT@blacky.mylan> Message-ID: <7bca989e-431e-a92d-1b98-992c8085c765@opensips.org> Rob, The module is still on the "TODO" list, not 100% sure it will make it for the 3.2 release as we have several other items (with higher priority) under work; see https://www.opensips.org/Development/Opensips-3-2-Planning Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2021 online https://opensips.org/training/OpenSIPS_eBootcamp_2021/ On 4/26/21 11:43 PM, Rob Dyck wrote: > i see that the roadmap shows MQTT support. I do not see it in the modules > list. Has this been dropped? > > Rob > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From kingsley at dns99.co.uk Tue Apr 27 12:03:53 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Tue, 27 Apr 2021 13:03:53 +0100 Subject: [OpenSIPS-Users] xlog - whether to \n In-Reply-To: <53668e63-41c1-1498-eb39-c28aee283dc2@opensips.org> References: <404b29dae8e6811708931c76ed12336dee949ded.camel@dns99.co.uk> <53668e63-41c1-1498-eb39-c28aee283dc2@opensips.org> Message-ID: <4e99740757bdd2983daf03927d0ec32b2d38d39e.camel@dns99.co.uk> OK, thanks. It was working without it but I've added \n to my xlog lines now. Cheers, Kingsley. On Tue, 2021-04-27 at 14:00 +0300, Bogdan-Andrei Iancu wrote: > The '\n' at the end of the xlog lines is a must. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/27/21 1:24 PM, Kingsley Tart wrote: > > Hi, > > > > If I use xlog() to log a single line, it does what I expect even if > > I > > don't end the log string with \n, so what is the reason I see > > people > > putting \n at the end of xlog strings? > > > > From kingsley at dns99.co.uk Tue Apr 27 12:42:04 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Tue, 27 Apr 2021 13:42:04 +0100 Subject: [OpenSIPS-Users] a comma ending a parameter list string? Message-ID: Hi, firstly apologies if this is a dumb question; I'm still pretty new to OpenSIPS and some aspects of SIP itself. Is a comma supposed to mark the end of a parameter list? Eg, if I do this, the param transformation is treating the first comma as the end of the string: $var(paramstr) = "h=BT1;dir=in;dprulesRX=1,2,3;dprulesTX=4,100"; xlog("PARAMSTR='$var(paramstr)'\n"); xlog("PARAMCOUNT='$(var(paramstr){param.count})'\n"); xlog("h='$(var(paramstr){param.value,h})'\n"); xlog("dir='$(var(paramstr){param.value,dir})'\n"); xlog("dprulesRX='$(var(paramstr){param.value,dprulesRX})'\n"); xlog("dprulesTX='$(var(paramstr){param.value,dprulesTX})'\n"); then this is what gets logged: PARAMSTR='h=BT1;dir=in;dprulesRX=1,2,3;dprulesTX=4,100' PARAMCOUNT='3' h='BT1' dir='in' dprulesRX='1' dprulesTX='' Is this deliberate behaviour? OpenSIPS 3.1, FWIW. -- Cheers, Kingsley. From kingsley at dns99.co.uk Tue Apr 27 13:01:29 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Tue, 27 Apr 2021 14:01:29 +0100 Subject: [OpenSIPS-Users] a comma ending a parameter list string? In-Reply-To: References: Message-ID: <46c8cfe7013331999ed334239a2d1d67ac5e619e.camel@dns99.co.uk> BTW, I noticed it's OK if I wrap double quotes around the CSV, eg: $var(paramstr) = "h=BT1;dir=in;dprulesRX=\"1,2,3\";dprulesTX=\"4,100\""; (getting a bit embarrassed with the number of emails I'm submitting to this list, sorry). Cheers, Kingsley. On Tue, 2021-04-27 at 13:42 +0100, Kingsley Tart wrote: > Hi, > > firstly apologies if this is a dumb question; I'm still pretty new to > OpenSIPS and some aspects of SIP itself. > > Is a comma supposed to mark the end of a parameter list? > > Eg, if I do this, the param transformation is treating the first > comma > as the end of the string: > > $var(paramstr) = > "h=BT1;dir=in;dprulesRX=1,2,3;dprulesTX=4,100"; > xlog("PARAMSTR='$var(paramstr)'\n"); > xlog("PARAMCOUNT='$(var(paramstr){param.count})'\n"); > > xlog("h='$(var(paramstr){param.value,h})'\n"); > xlog("dir='$(var(paramstr){param.value,dir})'\n"); > xlog("dprulesRX='$(var(paramstr){param.value,dprulesRX})'\n"); > xlog("dprulesTX='$(var(paramstr){param.value,dprulesTX})'\n"); > > then this is what gets logged: > > PARAMSTR='h=BT1;dir=in;dprulesRX=1,2,3;dprulesTX=4,100' > PARAMCOUNT='3' > h='BT1' > dir='in' > dprulesRX='1' > dprulesTX='' > > Is this deliberate behaviour? OpenSIPS 3.1, FWIW. > From liviu at opensips.org Tue Apr 27 13:15:16 2021 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 27 Apr 2021 16:15:16 +0300 Subject: [OpenSIPS-Users] a comma ending a parameter list string? In-Reply-To: References: Message-ID: On 27.04.2021 15:42, Kingsley Tart wrote: > $var(paramstr) = "h=BT1;dir=in;dprulesRX=1,2,3;dprulesTX=4,100"; > xlog("PARAMSTR='$var(paramstr)'\n"); > xlog("PARAMCOUNT='$(var(paramstr){param.count})'\n"); > ... > > Is this deliberate behaviour? OpenSIPS 3.1, FWIW. Hi, Yes, that is RFC 3261 behavior.  You are not allowed to put "," in either a SIP URI parameter name or its value. However, since you have invoked this part of the RFC: "A string of text is parsed as a single word if it is quoted using    double-quote marks.  In quoted strings, quotation marks (") and    backslashes (\) need to be escaped." ... OpenSIPS kindly obliges and parses the entire parameter value, with whatever content it may have. Regards, -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS eBootcamp 2021 | www.opensips.org/training From kingsley at dns99.co.uk Tue Apr 27 13:36:04 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Tue, 27 Apr 2021 14:36:04 +0100 Subject: [OpenSIPS-Users] a comma ending a parameter list string? In-Reply-To: References: Message-ID: <5d7c27307bc8bc5754f29b9994176877c694cbb5.camel@dns99.co.uk> Thanks. I actually wondered about that behaviour after I sent my initial email, tested it and found that it did that, but it's good to know that it's deliberate behaviour. Sorry to spam the list, I'm getting quite embarrassed about all of my emails :( Cheers, Kingsley. On Tue, 2021-04-27 at 16:15 +0300, Liviu Chircu wrote: > On 27.04.2021 15:42, Kingsley Tart wrote: > > $var(paramstr) = > > "h=BT1;dir=in;dprulesRX=1,2,3;dprulesTX=4,100"; > > xlog("PARAMSTR='$var(paramstr)'\n"); > > xlog("PARAMCOUNT='$(var(paramstr){param.count})'\n"); > > ... > > > > Is this deliberate behaviour? OpenSIPS 3.1, FWIW. > > Hi, > > Yes, that is RFC 3261 behavior. You are not allowed to put "," in > either a SIP URI parameter name or its value. > > However, since you have invoked this part of the RFC: > > "A string of text is parsed as a single word if it is quoted using > double-quote marks. In quoted strings, quotation marks (") and > backslashes (\) need to be escaped." > > ... OpenSIPS kindly obliges and parses the entire parameter value, > with > whatever content it may have. > > Regards, > From xaled at web.de Tue Apr 27 20:28:16 2021 From: xaled at web.de (xaled at web.de) Date: Tue, 27 Apr 2021 22:28:16 +0200 Subject: [OpenSIPS-Users] $b2b_logic.entity(id) in [reply_route] B2B module opensips 3.2 Message-ID: <0b0801d73ba3$dae18600$90a49200$@web.de> Hi, I'm not getting the $b2b_logic.entity(id) in B2B [reply_route]. The log shows Is this variable not available in the B2B [reply_route]? route[b2b_announce_response] { if ($rs != 500) { xlog("B2B REPLY: [$rs $rm] from entity: $b2b_logic.entity(id)\n"); b2b_handle_reply(); exit; } B2B REPLY: [183 INVITE] from entity: Thanks, Xaled From kingsley at dns99.co.uk Wed Apr 28 12:19:01 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Wed, 28 Apr 2021 13:19:01 +0100 Subject: [OpenSIPS-Users] routing function variable scope Message-ID: <927c304f4e603c4330f5329b00e66a4493dc0f2b.camel@dns99.co.uk> Hi, (sorry!) Is there a way in routing functions to create variables local to just that function? I had a problem earlier where I had a function call another, and they both used the same var name (intending them to be local to the function), but the sub-function ended up overwriting the value of the same-named var in the calling function. I find myself using AVPs and then unsetting them afterwards in order to not tread on vars that other functions might use but with all of the tidyup code things get messy. Foe example, this would loop forever, but using an AVP as the counter variable would then need another loop to remove all of the added values (but not all values in that AVP, in case that deletes AVPs set somewhere else): route(test_outer); route[test_outer] { $var(counter) = 0; while ($var(counter) < 10) { route(show_multiples, $var(counter)); $var(counter) = $var(counter) + 1; } } route[show_multiples] { $var(counter) = 1; while ($var(counter) < 5) { $var(multiple) = $param(1) * $var(counter); xlog("multiple=$var(multiple)\n"); $var(counter) = $var(counter) + 1; } } Is there a better way, apart from trying to make sure that variables in each function don't share names with variables in other functions? -- Cheers, Kingsley. From liviu at opensips.org Wed Apr 28 12:27:40 2021 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 28 Apr 2021 15:27:40 +0300 Subject: [OpenSIPS-Users] routing function variable scope In-Reply-To: <927c304f4e603c4330f5329b00e66a4493dc0f2b.camel@dns99.co.uk> References: <927c304f4e603c4330f5329b00e66a4493dc0f2b.camel@dns99.co.uk> Message-ID: <341ea76e-10c6-40f0-1cc2-6c7da67635e4@opensips.org> On 28.04.2021 15:19, Kingsley Tart wrote: > Is there a better way, apart from trying to make sure that variables in > each function don't share names with variables in other functions? No, you have to manage those iterator names and make sure the logic still holds during nested calls. Best, -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS eBootcamp 2021 | www.opensips.org/training From kingsley at dns99.co.uk Wed Apr 28 12:33:29 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Wed, 28 Apr 2021 13:33:29 +0100 Subject: [OpenSIPS-Users] routing function variable scope In-Reply-To: <341ea76e-10c6-40f0-1cc2-6c7da67635e4@opensips.org> References: <927c304f4e603c4330f5329b00e66a4493dc0f2b.camel@dns99.co.uk> <341ea76e-10c6-40f0-1cc2-6c7da67635e4@opensips.org> Message-ID: <2342ba7439d124d3957a3990c017b2a0e740892c.camel@dns99.co.uk> On Wed, 2021-04-28 at 15:27 +0300, Liviu Chircu wrote: > On 28.04.2021 15:19, Kingsley Tart wrote: > > Is there a better way, apart from trying to make sure that variables in > > each function don't share names with variables in other functions? > > No, you have to manage those iterator names and make sure the logic > still holds during nested calls. Right OK. Is it a reasonable feature request to have a variable type called $local that is only visible within the scope that created it? Or at least, not outside of the function, so you could use $local(myname) just like you would use $var(myname). Other functions could then share the same name and it would be their own local copy. If you think that's reasonable then I'll submit it. I'm not sure how much work it would be to implement, however! -- Cheers, Kingsley. From vladp at opensips.org Wed Apr 28 16:06:20 2021 From: vladp at opensips.org (Vlad Patrascu) Date: Wed, 28 Apr 2021 19:06:20 +0300 Subject: [OpenSIPS-Users] $b2b_logic.entity(id) in [reply_route] B2B module opensips 3.2 In-Reply-To: <0b0801d73ba3$dae18600$90a49200$@web.de> References: <0b0801d73ba3$dae18600$90a49200$@web.de> Message-ID: <50f14621-a284-5205-bb2d-1329ab480dc8@opensips.org> HI Xaled, The variable should normally be available in b2b_logic's reply route. Do you get the same result for the 200 OK reply or other replies? Also, is the other information about the entity available ($b2b_logic.entity(callid), $b2b_logic.entity(key)) ? Regards, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 27.04.2021 23:28, xaled at web.de wrote: > Hi, > > I'm not getting the $b2b_logic.entity(id) in B2B [reply_route]. The log shows > Is this variable not available in the B2B [reply_route]? > > route[b2b_announce_response] { > > if ($rs != 500) > { > xlog("B2B REPLY: [$rs $rm] from entity: $b2b_logic.entity(id)\n"); > b2b_handle_reply(); > exit; > } > > B2B REPLY: [183 INVITE] from entity: > > Thanks, > Xaled > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From vladp at opensips.org Wed Apr 28 16:25:47 2021 From: vladp at opensips.org (Vlad Patrascu) Date: Wed, 28 Apr 2021 19:25:47 +0300 Subject: [OpenSIPS-Users] Usage of TLS require_cert and verify_cert settings In-Reply-To: <5D29DEF8-AFCE-419D-A84A-338FF45B23BD@ag-projects.com> References: <5D29DEF8-AFCE-419D-A84A-338FF45B23BD@ag-projects.com> Message-ID: <5f3faa7b-0601-7b82-f2b9-03e640da09c4@opensips.org> Hi Adrian, It is not possible to change those parameters on the fly. But I think you can implement your scenarios by setting a domain with require_cert=0, verify_cert=0 and using the is_peer_verified() script function to check the connections from the remote server. Regards, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 22.04.2021 17:15, Adrian Georgescu wrote: > Hello, > > I have a question. > > I have the following TLS scenarios: > > 1) A local user for a domain I own, connects to my server using TLS. > If the domain is local, I will authenticate the user against my > database and I do not care if the user has a certificate > 2) A remote server, connects to my server using TLS and pretends that > is domainX.com . In such case, the only way to > verify that this is true is by requiring a certificate and verify it > > So there is a logic split between when to require and how to verify a > certificate depending on the fact that we deal with a local user or a > foreign domain. > > I would like to know if is possible to set require_cert and > verify_cert on the fly, while routing packets, instead of configuring > them statically per domain. > > Regards, > Adrian > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 29 08:43:15 2021 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 29 Apr 2021 11:43:15 +0300 Subject: [OpenSIPS-Users] OpenSIPS Summit Distributed 2021- in the cloud Message-ID: OpenSIPS Summit Distributed 2021 Sep 5th-10th, 2021 online *Bridging people, bridging technologies, bridging experiences * OpenSIPS is one of the most used Open Source SIP Servers in the world. It's routing millions of calls across the globe each day. It has been 100% Open Source and has been backed by a robust community for over 10 years.. Join the online OpenSIPS Summit to engage and learn about all the newest developments in OpenSIPS. You will surely walk away having captured important knowledge and insight into how OpenSIPS can be, and is being, used in some of the top companies in UCaaS and RTC. Experiencing first hand presentations on complex end-user deployments, high throughput infrastructure components, and the latest improvements to OpenSIPS is undoubtedly a "can't miss" experience for employees of carriers, telcos or ITSPs. *Some Great Reasons to Attend* * Access the latest news, knowledge and experience in the VoIP & RTC world * Learn about upcoming 3.2 OpenSIPS release and how you can leverage it * Attend unique presentations and interactive technical workshops * Meet FOSS developers and community to share experience and comments * Become an Expert attending the OpenSIPS Advanced Training *Summit Agenda* * Five full days of presentations given by key speakers * One full day of Interactive Demos and Showcases * One full day OpenSIPS Training (limited seats!) * Five days of networking with key people from OpenSIPS and other OSS projects * The mighty raffle *Be part of it* Be a part of both OpenSIPS and the Open Source community, be part of the OpenSIPS Summit 2021. *Speak to share* - the Call for Papers is already on, so you can share your wisdom and experience with the world. *Attend to learn* - the registration process will be open in the following days, stay tuned. Interested? Please contact our team or email us! ** -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From kingsley at dns99.co.uk Thu Apr 29 13:21:51 2021 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Thu, 29 Apr 2021 14:21:51 +0100 Subject: [OpenSIPS-Users] is this a bug? acc module setting empty string for unset fields Message-ID: <8ff40ca4ba26e9380e32d64bd73d6b1ce2c6bbea.camel@dns99.co.uk> Hi, This is with OpenSIPS 3.1. I have extra DB fields defined with "extra_fields", and this was working fine until I tried to turn on multi-leg accounting so that I could log details of failures, eg endpoint address, SIP response etc. I have a failure route set up to try the next endpoint, eg when I get an INVITE: t_on_failure("next_host"); and: failure_route[next_host] { # dialog flag 1 means a CANCEL has been received if (!is_dlg_flag_set(1)) { $avp(endpoint) = NULL; if ($avp(endpoint)) { t_on_failure("next_host"); acc_new_leg(); sethost($avp(endpoint)); route(numheaders_set_for_endpoint); route(relay); exit; } send_reply(503, "Service unavailable"); exit; } } The problems started when I added acc_new_leg() into the failure route. I've set this up so that the first few endpoints fail so that I can see what is happening. The problem is that when the BYE is received, I get the following errors: CRITICAL:db_mysql:wrapper_single_mysql_real_query: driver error (1366): Incorrect integer value: '' for column `opensips`.`acc`.`t_ringing` at row 1 ERROR:core:db_do_insert: error while submitting query ERROR:acc:acc_db_request: failed to insert into acc table The problem here is that t_ringing is an integer column (which defaults to NULL) so it can't take a string, and OpenSIPS is trying to set the value to an empty string. I tried specifically setting $acc_extra(t_ringing) = NULL but this made no difference. I tried moving t_ringing into "leg_fields" (needed to do that anyway) and then setting $acc_leg(t_ringing) in the script when needed, but the same problem persisted. These are the DB actions that OpenSIPS performs on the acc and missed_calls tables: Prepare insert into missed_calls (method,from_tag,to_tag,callid,sip_code,sip_reason,time,srcAddr,numFrom,numTo,numPAI,numPPI,hdrPrivacy,numRPID,hdrRPIDprivacy,hdrRPIDscreen,t_start,t_ringing,t_progress,t_answer,t_clear,dstAddr,setuptime,created ) values (?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?) Execute insert into missed_calls (method,from_tag,to_tag,callid,sip_code,sip_reason,time,srcAddr,numFrom,numTo,numPAI,numPPI,hdrPrivacy,numRPID,hdrRPIDprivacy,hdrRPIDscreen,t_start,t_ringing,t_progress,t_answer,t_clear,dstAddr,setuptime,created ) values ('INVITE','2007660754','z9hG4bK048.2ac41536.0','1619701113-752375-609080253','401','Unauthorized',TIMESTAMP'2021-04-29 13:58:33','88.151.41.21','00353*9051','01743850850','0018001231234','118118','none','003539994444',NULL,NULL,'1619701113',NULL,NULL,NULL,'1619701113','88.151.41.23',0,TIMESTAMP'2021-04-29 13:58:33') Prepare insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,srcAddr,numFrom,numTo,numPAI,numPPI,hdrPrivacy,numRPID,hdrRPIDprivacy,hdrRPIDscreen,t_start,t_ringing,t_progress,t_answer,t_clear,dstAddr,setuptime,created,duration,ms_duration ) values (?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?) Execute insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,srcAddr,numFrom,numTo,numPAI,numPPI,hdrPrivacy,numRPID,hdrRPIDprivacy,hdrRPIDscreen,t_start,t_ringing,t_progress,t_answer,t_clear,dstAddr,setuptime,created,duration,ms_duration ) values ('INVITE','2007660754',NULL,'1619701113-752375-609080253','503','Service Unavailable',TIMESTAMP'2021-04-29 13:58:33','88.151.41.21','00353*9051','01743850850','0018001231234','118118','none','003539994444',NULL,NULL,'1619701113',NULL,NULL,NULL,'1619701113','88.151.41.23',0,NULL,0,0) Query insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,srcAddr,numFrom,numTo,numPAI,numPPI,hdrPrivacy,numRPID,hdrRPIDprivacy,hdrRPIDscreen,t_start,t_ringing,t_progress,t_answer,t_clear,dstAddr,setuptime,created,duration,ms_duration ) values ('INVITE','2007660754','','1619701113-752375-609080253','503','Service Unavailable','2021-04-29 13:58:33','88.151.41.21','00353*9051','01743850850','0018001231234','118118','none','003539994444','','','1619701113','','','','1619701113','88.151.41.24',0,NULL,0,0) Is this a bug? I would prefer any fields not set in my script to be either set as NULL, or perhaps even better not even mentione in the query at all. -- Cheers, Kingsley. From social at bohboh.info Fri Apr 30 13:12:14 2021 From: social at bohboh.info (Social Boh) Date: Fri, 30 Apr 2021 08:12:14 -0500 Subject: [OpenSIPS-Users] OpenSIPs HA with KeepAlived Message-ID: Hello list, maybe someone might be interested: "OpenSIPs 3.1 HA with KeepAlilved": https://www.voztovoice.org/?q=node/2879 Spanish Article -- --- I'm SoCIaL, MayBe -------------- next part -------------- An HTML attachment was scrubbed... URL: From fedorna at anura.com.ar Fri Apr 30 16:34:46 2021 From: fedorna at anura.com.ar (Federico Edorna) Date: Fri, 30 Apr 2021 13:34:46 -0300 Subject: [OpenSIPS-Users] pkmem statistics for version 3.1 Message-ID: Hello! moving from 2.4 to 3.1 I've found something strange in pkmem stats. MI is showing zero values for the first 7 processes *opensips-cli -x mi get_statistics pkmem: -o output_type=lines*pkmem:1-used_size: 0 pkmem:1-real_used_size: 0 pkmem:1-max_used_size: 0 pkmem:1-free_size: 0 pkmem:1-fragments: 0 pkmem:2-used_size: 0 pkmem:2-real_used_size: 0 pkmem:2-max_used_size: 0 pkmem:2-free_size: 0 pkmem:2-fragments: 0 pkmem:3-used_size: 0 pkmem:3-real_used_size: 0 pkmem:3-max_used_size: 0 pkmem:3-free_size: 0 pkmem:3-fragments: 0 pkmem:4-used_size: 0 pkmem:4-real_used_size: 0 pkmem:4-max_used_size: 0 pkmem:4-free_size: 0 pkmem:4-fragments: 0 pkmem:5-used_size: 0 pkmem:5-real_used_size: 0 pkmem:5-max_used_size: 0 pkmem:5-free_size: 0 pkmem:5-fragments: 0 pkmem:6-used_size: 0 pkmem:6-real_used_size: 0 pkmem:6-max_used_size: 0 pkmem:6-free_size: 0 pkmem:6-fragments: 0 pkmem:7-used_size: 0 pkmem:7-real_used_size: 0 pkmem:7-max_used_size: 0 pkmem:7-free_size: 0 pkmem:7-fragments: 0 pkmem:8-used_size: 3167224 pkmem:8-real_used_size: 3387464 pkmem:8-max_used_size: 3396024 pkmem:8-free_size: 13389752 pkmem:8-fragments: 3561 pkmem:9-used_size: 3164712 pkmem:9-real_used_size: 3385048 pkmem:9-max_used_size: 3393616 pkmem:9-free_size: 13392168 pkmem:9-fragments: 3559 pkmem:10-used_size: 3164736 pkmem:10-real_used_size: 3385360 pkmem:10-max_used_size: 3393904 pkmem:10-free_size: 13391856 pkmem:10-fragments: 3559 pkmem:11-used_size: 3164688 pkmem:11-real_used_size: 3385840 pkmem:11-max_used_size: 3394384 pkmem:11-free_size: 13391376 pkmem:11-fragments: 3558 pkmem:12-used_size: 3163936 pkmem:12-real_used_size: 3379712 pkmem:12-max_used_size: 3379896 pkmem:12-free_size: 13397504 pkmem:12-fragments: 3547 pkmem:13-used_size: 3163936 pkmem:13-real_used_size: 3379712 pkmem:13-max_used_size: 3379896 pkmem:13-free_size: 13397504 pkmem:13-fragments: 3547 pkmem:14-used_size: 3163936 pkmem:14-real_used_size: 3379664 pkmem:14-max_used_size: 3379728 pkmem:14-free_size: 13397552 pkmem:14-fragments: 3547 pkmem:15-used_size: 3163936 pkmem:15-real_used_size: 3379712 pkmem:15-max_used_size: 3379896 pkmem:15-free_size: 13397504 pkmem:15-fragments: 3547 pkmem:16-used_size: 3163936 pkmem:16-real_used_size: 3379664 pkmem:16-max_used_size: 3379728 pkmem:16-free_size: 13397552 pkmem:16-fragments: 3547 pkmem:17-used_size: 3163936 pkmem:17-real_used_size: 3379664 pkmem:17-max_used_size: 3379728 pkmem:17-free_size: 13397552 pkmem:17-fragments: 3547 pkmem:18-used_size: 3163936 pkmem:18-real_used_size: 3379664 pkmem:18-max_used_size: 3379728 pkmem:18-free_size: 13397552 pkmem:18-fragments: 3547 pkmem:19-used_size: 3163936 pkmem:19-real_used_size: 3379664 pkmem:19-max_used_size: 3379728 pkmem:19-free_size: 13397552 pkmem:19-fragments: 3547 pkmem:20-used_size: 3163936 pkmem:20-real_used_size: 3379664 pkmem:20-max_used_size: 3379728 pkmem:20-free_size: 13397552 pkmem:20-fragments: 3547 pkmem:21-used_size: 3163936 pkmem:21-real_used_size: 3379664 pkmem:21-max_used_size: 3379728 pkmem:21-free_size: 13397552 pkmem:21-fragments: 3547 pkmem:22-used_size: 3163936 pkmem:22-real_used_size: 3379712 pkmem:22-max_used_size: 3379896 pkmem:22-free_size: 13397504 pkmem:22-fragments: 3547 pkmem:23-used_size: 3163936 pkmem:23-real_used_size: 3379664 pkmem:23-max_used_size: 3379728 pkmem:23-free_size: 13397552 pkmem:23-fragments: 3547 pkmem:24-used_size: 3163936 pkmem:24-real_used_size: 3379664 pkmem:24-max_used_size: 3379728 pkmem:24-free_size: 13397552 pkmem:24-fragments: 3547 pkmem:25-used_size: 3163936 pkmem:25-real_used_size: 3379664 pkmem:25-max_used_size: 3379728 pkmem:25-free_size: 13397552 pkmem:25-fragments: 3547 pkmem:26-used_size: 3163936 pkmem:26-real_used_size: 3379712 pkmem:26-max_used_size: 3379896 pkmem:26-free_size: 13397504 pkmem:26-fragments: 3547 pkmem:27-used_size: 3163936 pkmem:27-real_used_size: 3379664 pkmem:27-max_used_size: 3379728 pkmem:27-free_size: 13397552 pkmem:27-fragments: 3547 pkmem:28-used_size: 3164016 pkmem:28-real_used_size: 3380848 pkmem:28-max_used_size: 3385616 pkmem:28-free_size: 13396368 pkmem:28-fragments: 3549 pkmem:29-used_size: 3164016 pkmem:29-real_used_size: 3380848 pkmem:29-max_used_size: 3385616 pkmem:29-free_size: 13396368 pkmem:29-fragments: 3549 pkmem:30-used_size: 3164016 pkmem:30-real_used_size: 3380848 pkmem:30-max_used_size: 3385616 pkmem:30-free_size: 13396368 pkmem:30-fragments: 3549 pkmem:31-used_size: 3164016 pkmem:31-real_used_size: 3380848 pkmem:31-max_used_size: 3385616 pkmem:31-free_size: 13396368 pkmem:31-fragments: 3549 pkmem:32-used_size: 3167120 pkmem:32-real_used_size: 3386496 pkmem:32-max_used_size: 3395216 pkmem:32-free_size: 13390720 pkmem:32-fragments: 3556 pkmem:33-used_size: 3166664 pkmem:33-real_used_size: 3386040 pkmem:33-max_used_size: 3451584 pkmem:33-free_size: 13391176 pkmem:33-fragments: 3554 pkmem:34-used_size: 3164064 pkmem:34-real_used_size: 3381952 pkmem:34-max_used_size: 3389440 pkmem:34-free_size: 13395264 pkmem:34-fragments: 3550 pkmem:35-used_size: 3163936 pkmem:35-real_used_size: 3379664 pkmem:35-max_used_size: 3379728 pkmem:35-free_size: 13397552 pkmem:35-fragments: 3547 pkmem:36-used_size: 3163936 pkmem:36-real_used_size: 3379664 pkmem:36-max_used_size: 3379728 pkmem:36-free_size: 13397552 pkmem:36-fragments: 3547 pkmem:37-used_size: 3163936 pkmem:37-real_used_size: 3379664 pkmem:37-max_used_size: 3379728 pkmem:37-free_size: 13397552 pkmem:37-fragments: 3547 pkmem:38-used_size: 3163936 pkmem:38-real_used_size: 3379664 pkmem:38-max_used_size: 3379728 pkmem:38-free_size: 13397552 pkmem:38-fragments: 3547 pkmem:39-used_size: 3166392 pkmem:39-real_used_size: 3382936 pkmem:39-max_used_size: 3384192 pkmem:39-free_size: 13394280 pkmem:39-fragments: 3548 pkmem:40-used_size: 3163936 pkmem:40-real_used_size: 3379712 pkmem:40-max_used_size: 3379896 pkmem:40-free_size: 13397504 pkmem:40-fragments: 3547 pkmem:41-used_size: 3166456 pkmem:41-real_used_size: 3383048 pkmem:41-max_used_size: 3384440 pkmem:41-free_size: 13394168 pkmem:41-fragments: 3549 pkmem:42-used_size: 3166392 pkmem:42-real_used_size: 3382936 pkmem:42-max_used_size: 3384240 pkmem:42-free_size: 13394280 pkmem:42-fragments: 3548 pkmem:43-used_size: 3163936 pkmem:43-real_used_size: 3379712 pkmem:43-max_used_size: 3379896 pkmem:43-free_size: 13397504 pkmem:43-fragments: 3547 pkmem:44-used_size: 3163936 pkmem:44-real_used_size: 3379664 pkmem:44-max_used_size: 3379728 pkmem:44-free_size: 13397552 pkmem:44-fragments: 3547 pkmem:45-used_size: 3166392 pkmem:45-real_used_size: 3382936 pkmem:45-max_used_size: 3384192 pkmem:45-free_size: 13394280 pkmem:45-fragments: 3548 pkmem:46-used_size: 3166392 pkmem:46-real_used_size: 3382936 pkmem:46-max_used_size: 3384240 pkmem:46-free_size: 13394280 pkmem:46-fragments: 3548 pkmem:47-used_size: 3166392 pkmem:47-real_used_size: 3382936 pkmem:47-max_used_size: 3384240 pkmem:47-free_size: 13394280 pkmem:47-fragments: 3548 pkmem:48-used_size: 3166392 pkmem:48-real_used_size: 3382936 pkmem:48-max_used_size: 3384264 pkmem:48-free_size: 13394280 pkmem:48-fragments: 3548 pkmem:49-used_size: 2960896 pkmem:49-real_used_size: 3175664 pkmem:49-max_used_size: 3175664 pkmem:49-free_size: 13601552 pkmem:49-fragments: 2071 I've checked the log, in the initialization the process count seems to be fine, it finds 50 processes: *opensips[4644]: DBG:core:init_pkg_stats: setting stats for 50 processes* This is the output for *opensips-cli ps:* opensips-cli -x mi ps -o output_type=lines Processes: ID: 0 PID: 4644 Type: attendant ID: 1 PID: 4645 Type: HTTPD 10.10.10.184:9999 ID: 2 PID: 4646 Type: MI Datagram ID: 3 PID: 4647 Type: RabbitMQ sender ID: 4 PID: 4648 Type: Stun loop ID: 5 PID: 4649 Type: MI FIFO ID: 6 PID: 4650 Type: time_keeper ID: 7 PID: 4651 Type: timer ID: 8 PID: 4652 Type: SIP receiver udp:XXXX:YYY ID: 9 PID: 4653 Type: SIP receiver udp:XXXX:YYY ID: 10 PID: 4654 Type: SIP receiver udp:XXXX:YYYY ID: 11 PID: 4655 Type: SIP receiver udp:XXXX:YYYY ID: 12 PID: 4656 Type: SIP receiver udp:XXXX:YYYY ID: 13 PID: 4657 Type: SIP receiver udp:XXXX:YYYY ID: 14 PID: 4658 Type: SIP receiver udp:XXXX:YYYY ID: 15 PID: 4659 Type: SIP receiver udp:XXXX:YYYY ID: 16 PID: 4660 Type: SIP receiver udp:XXXX:YYYY ID: 17 PID: 4661 Type: SIP receiver udp:XXXX:YYYY ID: 18 PID: 4662 Type: SIP receiver udp:XXXX:YYYY ID: 19 PID: 4663 Type: SIP receiver udp:XXXX:YYYY ID: 20 PID: 4664 Type: SIP receiver udp:XXXX:YYYY ID: 21 PID: 4665 Type: SIP receiver udp:XXXX:YYYY ID: 22 PID: 4666 Type: SIP receiver udp:XXXX:YYYY ID: 23 PID: 4667 Type: SIP receiver udp:XXXX:YYYY ID: 24 PID: 4668 Type: SIP receiver udp:XXXX:YYYY ID: 25 PID: 4669 Type: SIP receiver udp:XXXX:YYYY ID: 26 PID: 4670 Type: SIP receiver udp:XXXX:YYYY ID: 27 PID: 4671 Type: SIP receiver udp:XXXX:YYYY ID: 28 PID: 4672 Type: SIP receiver udp:XXXX:YYYY ID: 29 PID: 4673 Type: SIP receiver udp:XXXX:YYYY ID: 30 PID: 4674 Type: SIP receiver udp:XXXX:YYYY ID: 31 PID: 4675 Type: SIP receiver udp:XXXX:YYYY ID: 32 PID: 4676 Type: TCP receiver ID: 33 PID: 4677 Type: TCP receiver ID: 34 PID: 4678 Type: TCP receiver ID: 35 PID: 4679 Type: TCP receiver ID: 36 PID: 4680 Type: TCP receiver ID: 37 PID: 4681 Type: TCP receiver ID: 38 PID: 4682 Type: TCP receiver ID: 39 PID: 4683 Type: TCP receiver ID: 40 PID: 4684 Type: TCP receiver ID: 41 PID: 4685 Type: TCP receiver ID: 42 PID: 4686 Type: TCP receiver ID: 43 PID: 4687 Type: TCP receiver ID: 44 PID: 4688 Type: TCP receiver ID: 45 PID: 4689 Type: TCP receiver ID: 46 PID: 4690 Type: TCP receiver ID: 47 PID: 4691 Type: TCP receiver ID: 48 PID: 4692 Type: Timer handler ID: 49 PID: 4693 Type: TCP main Is this an expected behaviour? Just wondering if pkmem stats are available only for workers..(?) Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: