You are correct, sorry. I'll fix and start testing again.
Thanks On Fri, Aug 31, 2018 at 10:10 AM Bogdan-Andrei Iancu <bog...@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 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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