[OpenSIPS-Users] Doubt about call center module

Daniel Zanutti daniel.zanutti at gmail.com
Fri Aug 31 09:17:32 EDT 2018


You are correct, sorry.

I'll fix and start testing again.

Thanks

On Fri, Aug 31, 2018 at 10:10 AM Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> As I said, in the cc_flows, you have no value for the "message_queue"
> column - this is a must, it has to be an URL to provide playback for the
> call queuing.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Bootcamp 2018
>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>
> On 08/31/2018 04:06 PM, Daniel Zanutti wrote:
>
> 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  1000 at plat5.domain.com  sip:1000 at plat5.domain.com:5060         1
> suporte       1535650312
>
> Thanks
>
> On Fri, Aug 31, 2018 at 5:02 AM Bogdan-Andrei Iancu <bogdan at 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 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
>>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20180831/dda88ebe/attachment-0001.html>


More information about the Users mailing list