[OpenSIPS-Users] Doubt about call center module
Bogdan-Andrei Iancu
bogdan at opensips.org
Fri Aug 31 09:10:24 EDT 2018
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 <mailto:1000 at plat5.domain.com>
> sip:1000 at plat5.domain.com:5060 <http://sip:1000@plat5.domain.com:5060>
> 1 suporte 1535650312
>
> Thanks
>
> On Fri, Aug 31, 2018 at 5:02 AM Bogdan-Andrei Iancu
> <bogdan at opensips.org <mailto: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 <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
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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