[OpenSIPS-Users] load_balance not releasing resources
Schneur Rosenberg
rosenberg11219 at gmail.com
Tue Nov 15 02:20:59 CET 2011
I might be missing a record route, would that cause this problem?
On Tue, Nov 15, 2011 at 3:16 AM, Schneur Rosenberg
<rosenberg11219 at gmail.com> wrote:
> Here is the code sending the 404 not here, I dont understand why
> if(loose_route()) does not return true, is this the way its supposed
> to be?
>
> if (has_totag())
> {
> # sequential request withing a dialog should
> # take the path determined by record-routing
> if (loose_route())
> {
> if (is_method("BYE"))
> {
> setflag(1); # do accounting ...
> setflag(3); # ... even if the transaction fails
> }
> else if (is_method("INVITE"))
> {
> record_route();
> }
> route(1);
> }
> else
> {
> if ( is_method("ACK") )
> {
> if ( t_check_trans() )
> {
> t_relay();
> exit;
> }
> else
> {
> exit;
> }
> }
> sl_send_reply("404","Not here");
> }
> exit;
> }
>
>
> On Tue, Nov 15, 2011 at 3:00 AM, Schneur Rosenberg
> <rosenberg11219 at gmail.com> wrote:
>> I see asterisk is sending the BYE to the phone, but opensips sends a
>> not here, bellow is the sip strace
>>
>> U 93.172.0.116:1047 -> opensipsip:5060INVITE
>> sip:1917398XXXX at opensipsip SIP/2.0.Via: SIP/2.0/UDP
>> 192.168.1.8:5060;branch=z9hG4bK-b5ec4068.From:
>> <sip:solhome3 at opensipsip>;tag=9c059eac8018b3c8o0.To:
>> <sip:19173985000 at opensipsip>.Remote-Party-ID:
>> <sip:solhome3 at opensipsip>;screen=yes;party=calling.Call-ID:
>> 82537c-a80f0538 at 192.168.1.8.CSeq: 101 INVITE.Max-Forwards: 70.Contact:
>> <sip:solhome3 at 192.168.1.8:5060>.Expires: 240.User-Agent:
>> Linksys/SPA2102-5.2.12.Content-Length: 444.Allow: ACK, BYE, CANCEL,
>> INFO, INVITE, NOTIFY, OPTIONS, REFER.Supported: x-sipura,
>> replaces.Content-Type: application/sdp.
>>
>>
>> U opensipsip:5060 -> 93.172.0.116:1047
>> SIP/2.0 407 Proxy Authentication Required.
>> Via: SIP/2.0/UDP
>> 192.168.1.8:5060;branch=z9hG4bK-b5ec4068;rport=1047;received=93.172.0.116.
>> From: <sip:solhome3 at opensipsip>;tag=9c059eac8018b3c8o0.
>> To: <sip:1917398XXXX at sopensipsip>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.ef95.
>> Call-ID: 82537c-a80f0538 at 192.168.1.8.
>> CSeq: 101 INVITE.
>> Proxy-Authenticate: Digest realm="opensipsip",
>> nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee".
>> Server: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
>> Content-Length: 0.
>>
>>
>> U 93.172.0.116:1047 -> opensipsIP:5060
>> INVITE sip:1917398XXXX at opensipsIP SIP/2.0.
>> Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK-ec946528.
>> From: <sip:solhome3 at opensipsIP>;tag=9c059eac8018b3c8o0.
>> To: <sip:1917398XXXX at opensipsIP>.
>> Remote-Party-ID: <sip:solhome3 at opensipsIP>;screen=yes;party=calling.
>> Call-ID: 82537c-a80f0538 at 192.168.1.8.
>> CSeq: 102 INVITE.
>> Max-Forwards: 70.
>> Proxy-Authorization: Digest
>> username="solhome3",realm="opensipsIP",nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee",uri="sip:1917398XXXX at opensipsIP",algorithm=MD5,response="db2640507b2e9824235649f51629ceee".
>> Contact: <sip:solhome3 at 192.168.1.8:5060>.
>> Expires: 240.
>> User-Agent: Linksys/SPA2102-5.2.12.
>> Content-Length: 444.
>> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
>> Supported: x-sipura, replaces.
>> Content-Type: application/sdp.
>>
>>
>> U opensipsIP:5060 -> 93.172.0.116:1047
>> SIP/2.0 100 Giving a try.
>> Via: SIP/2.0/UDP
>> 192.168.1.8:5060;branch=z9hG4bK-ec946528;rport=1047;received=93.172.0.116.
>> From: <sip:solhome3 at opensipsIP>;tag=9c059eac8018b3c8o0.
>> To: <sip:1917398xxxx at opensipsIP>.
>> Call-ID: 82537c-a80f0538 at 192.168.1.8.
>> CSeq: 102 INVITE.
>> Server: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
>> Content-Length: 0.
>>
>> U opensipsIP:5060 -> asteriskIP:5060
>> INVITE sip:1917398XXXX at opensipsIP SIP/2.0.
>> Record-Route: <sip:opensipsIP;lr=on;did=935.e9420777>.
>> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bK9049.19290602.0.
>> Via: SIP/2.0/UDP
>> 192.168.1.8:5060;rport=1047;received=93.172.0.116;branch=z9hG4bK-ec946528.
>> From: <sip:solhome3 at opensipsIP>;tag=9c059eac8018b3c8o0.
>> To: <sip:19173985000 at opensipsIP>.
>> Remote-Party-ID: <sip:solhome3 at opensipsIP>;screen=yes;party=calling.
>> Call-ID: 82537c-a80f0538 at 192.168.1.8.
>> CSeq: 102 INVITE.
>> Max-Forwards: 69.
>> Contact: <sip:solhome3 at 93.172.0.116:1047>.
>> Expires: 240.
>> User-Agent: Linksys/SPA2102-5.2.12.
>> Content-Length: 444.
>> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
>> Supported: x-sipura, replaces.
>> Content-Type: application/sdp.
>>
>> U asteriskIP:5060 -> opensipsIP:5060
>> SIP/2.0 100 Trying.
>> Via: SIP/2.0/UDPopensipsIP;branch=z9hG4bK9049.19290602.0;received=opensipsIP;rport=5060.
>> Via: SIP/2.0/UDP
>> 192.168.1.8:5060;rport=1047;received=93.172.0.116;branch=z9hG4bK-ec946528.
>> Record-Route: <sip:opensipsIP;lr=on;did=935.e9420777>.
>> From: <sip:solhome3 at opensipsIP>;tag=9c059eac8018b3c8o0.
>> To: <sip:1917398xxxx at opensipsIP>.
>> Call-ID: 82537c-a80f0538 at 192.168.1.8.
>> CSeq: 102 INVITE.
>> Server: Asterisk PBX 1.8.7.1.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH.
>> Supported: replaces, timer.
>> Contact: <sip:19173985000 at 64.69.47.109:5060>.
>> Content-Length: 0.
>>
>> U DIDProviderIP:5060 -> opensipsIP:5060
>> INVITE sip:917398xxxx at opensipsIP SIP/2.0.
>> Via: SIP/2.0/UDP DIDProviderIP:5060;branch=z9hG4bK0b523109;rport.
>> Max-Forwards: 70.
>> From: "ROSENBERG S" <sip:9173985xxxx at DIDproviderIP>;tag=as09899a91.
>> To: <sip:917398xxxx at opensipsIP>.
>> Contact: <sip:917398xxxx at DIDProviderip>.
>> Call-ID: 66d0ba94185dba0430f45f195772e31a at DIDProvidorIP.
>> CSeq: 102 INVITE.
>> User-Agent: Linksys/SPA2100-3.3.6(0911s).
>> Remote-Party-ID: "ROSENBERG S"
>> <sip:917398xxxx at DIDProviderIP>;privacy=off;screen=no.
>> Date: Mon, 14 Nov 2011 23:35:28 GMT.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
>> Supported: replaces, timer.
>> Content-Type: application/sdp.
>> Content-Length: 340.
>>
>> U opensipsIP:5060 -> asterisk2ip:5060
>> INVITE sip:did917398xxxx at opensipsIP SIP/2.0.
>> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bKf77f.f5d40393.0.
>> Via: SIP/2.0/UDPDIDProviderIP:5060;received=DIDProviderIP;branch=z9hG4bK0b523109;rport=5060.
>> Max-Forwards: 69.
>> From: "ROSENBERG S" <sip:917398xxxx at DIDProviderIP>;tag=as09899a91.
>> To: <sip:9173985000 at opensipsIP>.
>> Contact: <sip:917398xxxx at DIDProviderIP>.
>> Call-ID: 66d0ba94185dba0430f45f195772e31a at DIDProviderIP.
>> CSeq: 102 INVITE.
>> User-Agent: Linksys/SPA2100-3.3.6(0911s).
>> Remote-Party-ID: "ROSENBERG S"
>> <sip:917398xxxx at DIDProviderIP>;privacy=off;screen=no.
>> Date: Mon, 14 Nov 2011 23:35:28 GMT.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
>> Supported: replaces, timer.
>> Content-Type: application/sdp.
>> Content-Length: 340.
>> P-hint: Unathenticated from outside ie did.
>>
>> U asterisk2IP:5060 -> opensipsIP:5060
>> SIP/2.0 100 Trying
>> Truncated because of length
>>
>> U asterisk2IP:5060 -> opensipsIP:5060
>> INVITE sip:solhome7 at opensipsIP SIP/2.0.
>> Via: SIP/2.0/UDP asterisk2IP:5060;branch=z9hG4bK39459435;rport.
>> Max-Forwards: 70.
>> From: "ROSENBERG S" <sip:917398xxxx at asterisk2IP>;tag=as5ec8d074.
>> To: <sip:solhome5 at opensipsIP>.
>> Contact: <sip:917398xxxx at asterisk2IP:5060>.
>> Call-ID: 73f977bc448143a26b68be5d38de196e at asterisk2IP:5060.
>> CSeq: 102 INVITE.
>> User-Agent: Asterisk PBX 1.8.7.1.
>> Date: Mon, 14 Nov 2011 23:35:19 GMT.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH.
>> Supported: replaces, timer.
>> P-Asserted-Identity: "ROSENBERG S" <sip:917398xxxx at asterisk2IP>.
>> Content-Type: application/sdp.
>> Content-Length: 282.
>>
>> RINGING
>>
>> U 93.172.0.116:5060 -> opensipsIP:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bKa96f.8afc2a77.0.
>> Via: SIP/2.0/UDP
>> asterisk2IP:5060;received=asterisk2IP;branch=z9hG4bK727d493c;rport=5060.
>> From: "ROSENBERG S" <sip:917398xxxx at asterisk2IP>;tag=as605029e0.
>> To: <sip:solhome7 at sopensipsIP>;tag=6A174081-8FE8464C.
>> CSeq: 102 INVITE.
>> Call-ID: 09fdaad65a393c1751acd56e150d50a9 at asterisk2IP:5060.
>> Contact: <sip:solhome7 at 192.168.1.2>.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
>> NOTIFY, PRACK, UPDATE, REFER.
>> User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134.
>> Accept-Language: en.
>> Content-Type: application/sdp.
>> Content-Length: 197.
>>
>> U opensipsIP:5060 -> asterisk2IP:5060
>> SIP/2.0 200 OK.
>>
>> U asterisk2IP:5060 -> opensipsIP:5060
>> ACK sip:solhome7 at 93.172.0.116:5060;nat=yes SIP/2.0.
>>
>> U 93.172.0.116:1047 -> opensipsIP:5060
>> BYE sip:1917398xxxx at asteriskIP:5060;nat=yes SIP/2.0.
>> Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK-5f187bca.
>> From: <sip:solhome3 at opensipsIP>;tag=9c059eac8018b3c8o0.
>> To: <sip:1917398xxxx at opensipsIP>;tag=as5852d19d.
>> Call-ID: 82537c-a80f0538 at 192.168.1.8.
>> CSeq: 103 BYE.
>> Max-Forwards: 70.
>> Route: <sip:opensipsIP;lr=on;did=935.e9420777>.
>> Proxy-Authorization: Digest
>> username="solhome3",realm="opensipsIP",nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee",uri="sip:1917398xxxx at asteriskIP:5060",algorithm=MD5,response="3bc688c27090bca344187bef1a5e4eee".
>> User-Agent: Linksys/SPA2102-5.2.12.
>> Content-Length: 0.
>> .
>>
>>
>> U opensipsIP:5060 -> asteriskIP:5060
>> BYE sip:1917398xxxx at asteriskIP:5060 SIP/2.0.
>> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bKa049.76464162.0.
>> Via: SIP/2.0/UDP
>> 192.168.1.8:5060;rport=1047;received=93.172.0.116;branch=z9hG4bK-5f187bca.
>> From: <sip:solhome3 at opensikpsIP>;tag=9c059eac8018b3c8o0.
>> To: <sip:1917398xxxx at opensipsIP>;tag=as5852d19d.
>> Call-ID: 82537c-a80f0538 at 192.168.1.8.
>> CSeq: 103 BYE.
>> Max-Forwards: 69.
>> Proxy-Authorization: Digest
>> username="solhome3",realm="opensipsIP",nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee",uri="sip:1917398xxxx at asteriskIP:5060",algorithm=MD5,response="3bc688c27090bca344187bef1a5e49d8".
>> User-Agent: Linksys/SPA2102-5.2.12.
>> Content-Length: 0.
>>
>> U asteriskIP:5060 -> opensipsIP:5060
>> SIP/2.0 200 OK.
>>
>> U opensipsIP:5060 -> 93.172.0.116:1047
>> SIP/2.0 200 OK.
>>
>> U asterisk2IP:5060 -> opensipsIP:5060
>> BYE sip:solhome7 at 93.172.0.116:5060;nat=yes SIP/2.0.
>>
>> .
>> U opensipsIP:5060 -> asteriskIP:5060
>> SIP/2.0 404 Not here.
>>
>>
>>
>>
>> On Tue, Nov 15, 2011 at 2:19 AM, <duane.larson at gmail.com> wrote:
>>> Could you provide a sip trace of a call from INVITE to BYE? Also in your
>>> opensips config look and see where you have "404 Not here" configured.
>>>
>>>
>>>
>>> On , Schneur Rosenberg <rosenberg11219 at gmail.com> wrote:
>>>> In my case this is not relevant, because I'm calling the other phone
>>>>
>>>> through a DID and the did needs to go to asterisk to decide what to do
>>>>
>>>> with it, it can send it to a IVR which can later send it to Opensips
>>>>
>>>> etc. in any case I need to know why asterisk is not sending the BYE to
>>>>
>>>> the phone, and why opensips sends a not here when the BYE comes from a
>>>>
>>>> phone not on the system, in that case asterisk sends the BYE to
>>>>
>>>> opensips which sends a not here instead of sending it to the phone
>>>>
>>>>
>>>>
>>>> On Tue, Nov 15, 2011 at 2:06 AM, duane.larson at gmail.com> wrote:
>>>>
>>>> > If you want VM then you send to Asterks when the call times out (AKA the
>>>>
>>>> > callee doesn't pick up). We weren't talking about VM here. If you want
>>>> > MOH
>>>>
>>>> > then that is a totally different beast. You would always have to send
>>>> > the
>>>>
>>>> > calls to Asterisk and Asterisk would stay in the flow of the call. From
>>>> > what
>>>>
>>>> > I read above it sounded like the following
>>>>
>>>> >
>>>>
>>>> > When I call from one phone on the system to another phone on the
>>>>
>>>> > same opensips, the phone sends a BYE to opensips which sends it to the
>>>>
>>>> > asterisk but the BYE never gets sent to the called phone.
>>>>
>>>> >
>>>>
>>>> > Sounds like Asterisk is not sending the BYE back to OpenSIPS because its
>>>>
>>>> > stated " opensips which sends it to the asterisk but the BYE never gets
>>>> > sent
>>>>
>>>> > to the called phone."
>>>>
>>>> >
>>>>
>>>> >
>>>>
>>>> >
>>>>
>>>> >
>>>>
>>>> > On , Nick Khamis symack at gmail.com> wrote:
>>>>
>>>> >> On Mon, Nov 14, 2011 at 6:50 PM, duane.larson at gmail.com> wrote:
>>>>
>>>> >>
>>>>
>>>> >> > If two phones are registered with OpenSIPS and they call each other
>>>> >> > why
>>>>
>>>> >>
>>>>
>>>> >> > would you send the SIP messages to Asterisk?
>>>>
>>>> >>
>>>>
>>>> >>
>>>>
>>>> >>
>>>>
>>>> >> Because "http://www.opensips.org/Resources/DocsTutAsterisk" said so! ;)
>>>>
>>>> >>
>>>>
>>>> >>
>>>>
>>>> >>
>>>>
>>>> >> > You need to set up route logic so that if two local users call each
>>>>
>>>> >> > other then
>>>>
>>>> >>
>>>>
>>>> >> > the asterisk boxes are kept out of the equation.
>>>>
>>>> >>
>>>>
>>>> >>
>>>>
>>>> >>
>>>>
>>>> >> Amazing idea! But what would happen to MOH, and VM?
>>>>
>>>> >>
>>>>
>>>> >>
>>>>
>>>> >>
>>>>
>>>> >> Nick.
>>>>
>>>> >>
>>>>
>>>> >>
>>>>
>>>> >>
>>>>
>>>> >> _______________________________________________
>>>>
>>>> >>
>>>>
>>>> >> 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
>>>
>>>
>>
>
More information about the Users
mailing list