Hi Bogdan

Here it is table cc_flows:
    id  flowid  priority  skill    prependcid  message_welcome
message_queue
------  ------  --------  -------  ----------  ---------------
---------------
     1  fila-1       256  suporte  fila-1


Also table agents:
    id  agentid                 location                         logstate
skills   last_call_end
------  ----------------------  -------------------------------  --------
-------  ---------------
     1  1...@plat5.domain.com  sip:1...@plat5.domain.com:5060         1
suporte       1535650312

Thanks

On Fri, Aug 31, 2018 at 5:02 AM Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:

> 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 -> 10.10.10.10:5060
> INVITE 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@10.10.10.10:5070>;tag=as6440e239.
> To: <sip:fila-1@10.10.10.10:5060>.
> Contact: <sip:551122223333@10.10.10.10:5070>.
> Call-ID: 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@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 <bog...@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.zanu...@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
>>>
>>
>>
>> _______________________________________________
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
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