[OpenSIPS-Users] Doubt about call center module

Daniel Zanutti daniel.zanutti at gmail.com
Thu Aug 30 13:34:44 EDT 2018


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 -> 10.10.10.10:5060
INVITE sip:fila-1 at 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>;tag=as6440e239.
To: <sip:fila-1 at 10.10.10.10:5060>.
Contact: <sip:551122223333 at 10.10.10.10:5070>.
Call-ID: 357cf76348e4e68325d065e85282320a at 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>.
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>
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>
> 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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