[OpenSIPS-Users] Doubt about call center module

Bogdan-Andrei Iancu bogdan at opensips.org
Fri Aug 31 04:02:47 EDT 2018


Hi Daniel,

It is not about the B2B scenario, but about how you provisioned the flow 
in DB. Could you simply dump the output of "select * from cc_flows" ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
   http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 08/30/2018 08:34 PM, Daniel Zanutti wrote:
> Hi Bogdan
>
> Yes, It's the same scenario and same message. The call flow is:
>
> Asterisk Dials(port 5070) -> Opensips (port 5060) forward to Queue -> 
> Calls local user
>
> I'm using standard Queue scenario:
> <?xml version="1.0"?>
> <scenario id="call center" name="Call center" param="1" type="script">
>         <init>
>                 <bridge>
>                         <server>
> <id>server1</id>
>                         </server>
>                         <client>
> <id>client1</id>
> <type>message</type>
>                                 <destination>
>                                         <value type="param">1</value>
>                                 </destination>
>                         </client>
>                 </bridge>
>                 <state>1</state>
>         </init>
> </scenario>
>
> And SIP message is the same on all calls, just changed Call-id/tags:
>
> U 10.10.10.10:5070 <http://10.10.10.10:5070> -> 10.10.10.10:5060 
> <http://10.10.10.10:5060>
> INVITE sip:fila-1 at 10.10.10.10:5060 
> <http://sip:fila-1@10.10.10.10:5060> SIP/2.0.
> Via: SIP/2.0/UDP 10.10.10.10:5070;branch=z9hG4bK2abb2acc;rport.
> Max-Forwards: 70.
> From: <sip:551122223333 at 10.10.10.10:5070 
> <http://sip:551122223333@10.10.10.10:5070>>;tag=as6440e239.
> To: <sip:fila-1 at 10.10.10.10:5060 <http://sip:fila-1@10.10.10.10:5060>>.
> Contact: <sip:551122223333 at 10.10.10.10:5070 
> <http://sip:551122223333@10.10.10.10:5070>>.
> Call-ID: 357cf76348e4e68325d065e85282320a at 10.10.10.10:5070 
> <http://357cf76348e4e68325d065e85282320a@10.10.10.10:5070>.
> CSeq: 102 INVITE.
> User-Agent: PBX SIPTEK.
> Date: Thu, 30 Aug 2018 17:30:30 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
> INFO, PUBLISH, MESSAGE.
> Supported: replaces, timer.
> P-Asserted-Identity: "551122223333" <sip:551122223333 at 10.10.10.10 
> <mailto:sip%3A551122223333 at 10.10.10.10>>.
> Content-Type: application/sdp.
> Content-Length: 353.
> [SDP OMMITED]
>
> I updated to latest 2.4.2 GIT version (commit 
> 8b6830cdd96298682fcc298095ad1b718c54c77d), same problem is happening.
>
> Also you can access the server if you want, it's dedicated to this test.
>
> Thanks
>
>
>
>
> On Thu, Aug 30, 2018 at 1:04 PM Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Daniel,
>
>     Are you sure you configured a proper SIP URI as "message_queue" in
>     the flow description ? My impression is you have an empty string
>     there - and OpenSIPS is trying to put the call on the queue (as
>     there is no agent), but the SIP URI is not valid.
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>        http://www.opensips-solutions.com
>     OpenSIPS Bootcamp 2018
>        http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>
>     On 08/29/2018 10:26 PM, Daniel Zanutti wrote:
>>     Got some more info.
>>
>>     *This is the first call that worked fine:*
>>     ......
>>
>>     *This is the second call that had the problem:*
>>     .....
>>     Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>>     DBG:call_center:cc_call_state_machine: selecting QUEUE
>>     Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>>     DBG:call_center:cc_queue_push_call:  QUEUE - adding call
>>     0x7fd8510524a8
>>     Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>>     DBG:call_center:cc_queue_push_call: adding call on pos 0 (already
>>     1 calls), l=(nil) h=(nil)
>>     Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>>     DBG:call_center:w_handle_call: new destination for
>>     call(0x7fd8510524a8) is  (state=2)
>>     .....
>>
>>
>>     On Mon, Aug 27, 2018 at 6:15 PM Daniel Zanutti
>>     <daniel.zanutti at gmail.com <mailto:daniel.zanutti at gmail.com>> wrote:
>>
>>         Trying to configure the call center modules, but found a
>>         problem when there is no agents available.
>>
>>         If there is 1 agent available, call is sent to him with no
>>         problem:
>>
>>         Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Recebida
>>         asterisk - Tentando entrar na fila fila-1
>>         Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Entrou na fila
>>         com sucesso (fila-1)!
>>         Aug 27 18:11:01 plat5 /sbin/opensips[23569]: incoming reply
>>
>>         But when there is no agent available, opensips refuses:
>>         Aug 27 18:11:07 plat5 /sbin/opensips[23569]: Recebida
>>         asterisk - Tentando entrar na fila fila-1
>>         Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>>         ERROR:b2b_logic:b2b_process_scenario_init: Failed to get the
>>         value for the b2b client ruri
>>         Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>>         ERROR:call_center:set_call_leg: failed to init new b2bua call
>>         (empty ID received)
>>         Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>>         ERROR:call_center:w_handle_call: failed to set new
>>         destination for call
>>         Aug 27 18:11:07 plat5 /sbin/opensips[23569]: errnum: -1
>>
>>         Error -1 means flowID is invalid, but I sent the same value
>>         on both calls.
>>
>>         This is the call:
>>
>>         cc_handle_call("$rU")
>>
>>         I'm using Opensips 2.4.2 with Debian 8.11.
>>
>>         Am I missing something or found a bug?
>>
>>         Thanks
>>
>>
>>
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>

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