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Freepbx 14 (asterisk 14) incoming call from OBI 202 FXO drops

Started by mlaihk, January 11, 2018, 02:28:49 AM

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mlaihk

Something really had me pulling my hair out!  I have a OBI202 with OBiLine as a FXO to my Freepbx 14 (asterisk 14) with PJSIP.  Incoming PSTN calls from the OBiLine gets thru to the Freepbx14 box and as soon as Freepbx 14 rings the local extensions (IP Phones), it tells the OBI202 that it is ringing via a SIP message.

Seems that within seconds that OBI202 receives the SIP Ringing message, it drops the line and tells Freepbx to cancel the connection, which Freepbx than kills the connect.  This results in calls being dropped before they can be answered!!!!!

Is there someway that I can fix this?  The OBI202 used to work with Freepbx 13 and Asterisk 13 without issue.......

Attached is the syslog capture from the OBI202 while this is happening.......


<7>1 2018-01-11T17:55:08.303940+08:00  OBI200 - - - OBI200: [DAA]: FXO ring on
<7>1 2018-01-11T17:55:09.282870+08:00  OBI200 - - - OBI200: ------ caller id (pcm_id: 2) received! ------------
<7>1 2018-01-11T17:55:09.283729+08:00  OBI200 - - - OBI200: [0]DAA CND 01111755,97631267,,,,
<7>1 2018-01-11T17:55:09.311163+08:00  OBI200 - - - OBI200: CCTL:NewCallOn Term 10[0] ->,28907870
<7>1 2018-01-11T17:55:09.316110+08:00  OBI200 - - - OBI200: sendto c0a800d6:5060(844)
<7>1 2018-01-11T17:55:09.318968+08:00  OBI200 - - - OBI200: INVITE sip:28907870@192.168.0.214:5060 SIP/2.0#015#012Call-ID: e904b5b2eb574ff2@192.168.0.203#015#012Content-Length: 304#015#012CSeq: 8001 INVITE#015#012From: <sip:97631267@192.168.0.214>;tag=SP1ceff6c517b4ed083#015#012Max-Forwards: 70#015#012To: <sip:28907870@192.168.0.214>#015#012Via: SIP/2.0/UDP 192.168.0.203:5060;branch=z9hG4bK-bbfe5dc8;rport#015#012User-Agent: OBIHAI/OBi202-3.2.1.5775#015#012Contact: "28907870" <sip:obi28907870@192.168.0.203:5060>#015#012Expires: 60#015#012Supported: replaces#015#012Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE#015#012Content-Type: application/sdp#015#012#015#012v=0#015#012o=- 1506197 1 IN IP4 192.168.0.203#015#012s=-#015#012c=IN IP4 192.168.0.203#015#012t=0 0#015#012m=audio 10008 RTP/AVP 0 8 100 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:100 fax-event/8000#015#012a=fmtp:100 32#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=ptime:20#015#012a=xg726bitorder:big-endian#015
<7>1 2018-01-11T17:55:09.323165+08:00  OBI200 - - - OBI200: RxFrom:c0a800d6:5060
<7>1 2018-01-11T17:55:09.323165+08:00  OBI200 - - - OBI200: SIP/2.0 100 Trying#015#012Via: SIP/2.0/UDP 192.168.0.203:5060;rport=5060;received=192.168.0.203;branch=z9hG4bK-bbfe5dc8#015#012Call-ID: e904b5b2eb574ff2@192.168.0.203#015#012From: <sip:97631267@192.168.0.214>;tag=SP1ceff6c517b4ed083#015#012To: <sip:28907870@192.168.0.214>#015#012CSeq: 8001 INVITE#015#012Server: FPBX-14.0.1.24(14.7.5)#015#012Content-Length:  0#015#012#015
<7>1 2018-01-11T17:55:18.400025+08:00  OBI200 - - - OBI200: RxFrom:c0a800d6:5060
<7>1 2018-01-11T17:55:18.400196+08:00  OBI200 - - - OBI200: SIP/2.0 180 Ringing#015#012Via: SIP/2.0/UDP 192.168.0.203:5060;rport=5060;received=192.168.0.203;branch=z9hG4bK-bbfe5dc8#015#012Call-ID: e904b5b2eb574ff2@192.168.0.203#015#012From: <sip:97631267@192.168.0.214>;tag=SP1ceff6c517b4ed083#015#012To: <sip:28907870@192.168.0.214>;tag=ecf8ea06-41ed-4fe2-b098-fc4db1c2f0c8#015#012CSeq: 8001 INVITE#015#012Server: FPBX-14.0.1.24(14.7.5)#015#012Contact: <sip:192.168.0.214:5060>#015#012Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER#015#012Content-Length:  0#015#012#015
<7>1 2018-01-11T17:55:19.638205+08:00  OBI200 - - - OBI200: RxFrom:c0a800d6:5060
<7>1 2018-01-11T17:55:19.638425+08:00  OBI200 - - - OBI200: SIP/2.0 180 Ringing#015#012Via: SIP/2.0/UDP 192.168.0.203:5060;rport=5060;received=192.168.0.203;branch=z9hG4bK-bbfe5dc8#015#012Call-ID: e904b5b2eb574ff2@192.168.0.203#015#012From: <sip:97631267@192.168.0.214>;tag=SP1ceff6c517b4ed083#015#012To: <sip:28907870@192.168.0.214>;tag=ecf8ea06-41ed-4fe2-b098-fc4db1c2f0c8#015#012CSeq: 8001 INVITE#015#012Server: FPBX-14.0.1.24(14.7.5)#015#012Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER#015#012Contact: <sip:192.168.0.214:5060>#015#012Content-Length:  0#015#012#015
<7>1 2018-01-11T17:55:29.805451+08:00  OBI200 - - - OBI200: [DAA]: FXO ring off
<7>1 2018-01-11T17:55:29.806803+08:00  OBI200 - - - OBI200: sendto c0a800d6:5060(346)
<7>1 2018-01-11T17:55:29.806803+08:00  OBI200 - - - OBI200: CANCEL sip:28907870@192.168.0.214:5060 SIP/2.0#015#012Call-ID: e904b5b2eb574ff2@192.168.0.203#015#012Content-Length: 0#015#012CSeq: 8001 CANCEL#015#012From: <sip:97631267@192.168.0.214>;tag=SP1ceff6c517b4ed083#015#012Max-Forwards: 70#015#012To: <sip:28907870@192.168.0.214>#015#012Via: SIP/2.0/UDP 192.168.0.203:5060;branch=z9hG4bK-bbfe5dc8;rport#015#012User-Agent: OBIHAI/OBi202-3.2.1.5775#015#012#015
<7>1 2018-01-11T17:55:29.807739+08:00  OBI200 - - - OBI200: RTP:Del Channel
<7>1 2018-01-11T17:55:29.823224+08:00  OBI200 - - - OBI200: [DAA]: FXO ONHOOK MONITOR
<7>1 2018-01-11T17:55:29.829586+08:00  OBI200 - - - OBI200: RxFrom:c0a800d6:5060
<7>1 2018-01-11T17:55:29.829763+08:00  OBI200 - - - OBI200: SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP 192.168.0.203:5060;rport=5060;received=192.168.0.203;branch=z9hG4bK-bbfe5dc8#015#012Call-ID: e904b5b2eb574ff2@192.168.0.203#015#012From: <sip:97631267@192.168.0.214>;tag=SP1ceff6c517b4ed083#015#012To: <sip:28907870@192.168.0.214>;tag=ecf8ea06-41ed-4fe2-b098-fc4db1c2f0c8#015#012CSeq: 8001 CANCEL#015#012Server: FPBX-14.0.1.24(14.7.5)#015#012Content-Length:  0#015#012#015
<7>1 2018-01-11T17:55:29.832905+08:00  OBI200 - - - OBI200: RxFrom:c0a800d6:5060
<7>1 2018-01-11T17:55:29.833423+08:00  OBI200 - - - OBI200: SIP/2.0 487 Request Terminated#015#012Via: SIP/2.0/UDP 192.168.0.203:5060;rport=5060;received=192.168.0.203;branch=z9hG4bK-bbfe5dc8#015#012Call-ID: e904b5b2eb574ff2@192.168.0.203#015#012From: <sip:97631267@192.168.0.214>;tag=SP1ceff6c517b4ed083#015#012To: <sip:28907870@192.168.0.214>;tag=ecf8ea06-41ed-4fe2-b098-fc4db1c2f0c8#015#012CSeq: 8001 INVITE#015#012Server: FPBX-14.0.1.24(14.7.5)#015#012Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER#015#012Content-Length:  0#015#012#015
<7>1 2018-01-11T17:55:29.833815+08:00  OBI200 - - - OBI200: sendto c0a800d6:5060(381)
<7>1 2018-01-11T17:55:29.834623+08:00  OBI200 - - - OBI200: ACK sip:28907870@192.168.0.214:5060 SIP/2.0#015#012Call-ID: e904b5b2eb574ff2@192.168.0.203#015#012Content-Length: 0#015#012CSeq: 8001 ACK#015#012From: <sip:97631267@192.168.0.214>;tag=SP1ceff6c517b4ed083#015#012Max-Forwards: 70#015#012To: <sip:28907870@192.168.0.214>;tag=ecf8ea06-41ed-4fe2-b098-fc4db1c2f0c8#015#012Via: SIP/2.0/UDP 192.168.0.203:5060;branch=z9hG4bK-bbfe5dc8;rport#015#012User-Agent: OBIHAI/OBi202-3.2.1.5775#015#012#015
<7>1 2018-01-11T17:55:52.717574+08:00  OBI200 - - - OBI200: RxFrom:c0a800d6:5060

Please help!

azrobert

I'm running Freepbx 14/Asterisk 13 and had a different extension problem. Changing the extension definition from PJSIP to Chan_SIP fixed the problem. It can't hurt to try.

mlaihk

I had no problems with OBI202 and PJSIP as FXS to my fax machine at all.

Just that the OBI202 with OBI Line is giving me grief.  Upon digging deeper, it appears that OBI202/OBI Line gives up as it detects a FXO Ring Drop event while the internet phones were still ringing, which causes the OBI202 to cancel the Invite to Asterisk.  That's the problem why calls don't get thru.

As a test, I had the incoming call routed to an Echo Test/Time test feature codes, which asterisk picks up right away without ringing.  And incoming calls get to the Echo Test and Time Test just fine.

It is that before anybody can pick up the phones, OBI Line detects a FXO Ring Drop event and cancels the line!

This points to either a hardware problem with the OBI Line or a config issue on the OBI202.....

drgeoff

Something strange happening.  You seem to think the problem is the 202 or OBiLINE but both worked OK until you changed the Asterisk version.

Have you done the obvious test of directing the incoming call on OBiLINE to a phone port on the 202?  Does it keep ringing that or does it drop after the same 10 seconds or so?

mlaihk

Will try that once I get a wired handset.....

Actually, I just tried without a phone but pointing Line to ring PH1.  And it DOES DROP THE LINE after couple of rings.....

mlaihk

Nevermind.....  Turns out that it was a problem with the PSTN line.  The line was previously configured as a hunting line, and when all the other lines were cancelled, the telco mis-configured this line and screwed it up......

Basically, the line was still hunting when no answer after 5 rings and it moved the call to a non-existing number......