@stevet wrote:
New installation of of the FreePBX distro on our own in-house hardware. Migrated from an Elastix VPS in the cloud, partly because I thought it might solve this problem.
Starting last week, we began to have issues with outbound calls. A call is passed to our provider (Vitelity), and as soon as it's answered, it drops. This is intermittent, but seems to happen in clumps, or everything is fine for a while, and then we can't make calls for a while, rinse, repeat.
I'd recently done an major update on the Elastix system, and figured that had caused some sort of issue. Since I was planning on moving away from that system anyway, I got FreePBX set up at the end of the week and over the weekend, and we were up and running on Monday morning, but the problem persists. Different server, different network, different distro, even different phones in a couple of cases. Figured it was Vitelity, so I opened a ticket with them and after digging into their logs, they informed me that the BYE signal was coming from my end...
I've searched everything I can think of to find people who've had the same issue, and most of what I found was from years ago and is unrelated. The most likely thing I can come up with is issues with RTP/firewall, but disabling the firewall doesn't solve the problem.
I captured the sip debug output from one of these calls, which is below. Has anyone seen this recently? I've been running various * distros for almost a decade and haven't run into anything like this before that couldn't be solved by flushing the firewall rules. Even * doesn't know why it hung up, according to the debug log.
TIA
X.X.X.X = OUR PBX IP (PUBLIC, NOT BEHIND NAT) <--- SIP read from UDP:64.2.142.216:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK32d78e67;received=X.X.X.X From: <sip:myusername@X.X.X.X:5160>;tag=as68734862 To: <sip:15033730000@outbound.vitelity.net> Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160 CSeq: 103 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:15033730000@64.2.142.216> Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:64.2.142.216:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK32d78e67;received=X.X.X.X From: <sip:myusername@X.X.X.X:5160>;tag=as68734862 To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561 Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160 CSeq: 103 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 334 v=0 o=root 18246 18246 IN IP4 64.2.142.216 s=session c=IN IP4 64.2.142.216 t=0 0 m=audio 19544 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (11 headers 16 lines) --- sip_route_dump: no route/path Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|g726|g723|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk), peer - audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 64.2.142.216:19544 -- SIP/vitel-outbound-0000000d is making progress passing it to PJSIP/102-00000019 <--- SIP read from UDP:64.2.142.216:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK32d78e67;received=X.X.X.X From: <sip:myusername@X.X.X.X:5160>;tag=as68734862 To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561 Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160 CSeq: 103 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 334 v=0 o=root 18246 18247 IN IP4 64.2.142.216 s=session c=IN IP4 64.2.142.216 t=0 0 m=audio 19544 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (11 headers 16 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|g726|g723|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk), peer - audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 64.2.142.216:19544 sip_route_dump: no route/path [2016-12-21 10:22:52] WARNING[2431][C-0000000e]: chan_sip.c:16592 __set_address_from_contact: Invalid contact uri (missing sip: or sips:), attempting to use anyway [2016-12-21 10:22:52] WARNING[2431][C-0000000e]: chan_sip.c:16605 __set_address_from_contact: Invalid URI: parse_uri failed to acquire hostport Transmitting (no NAT) to 64.2.142.216:5060: ACK sip:15033730000@outbound.vitelity.net SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK774eb655 Max-Forwards: 70 From: <sip:myusername@X.X.X.X:5160>;tag=as68734862 To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561 Contact: <sip:myusername@X.X.X.X:5160> Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160 CSeq: 103 ACK User-Agent: FPBX-13.0.190.7(13.13.1) Content-Length: 0 --- Reliably Transmitting (no NAT) to 64.2.142.216:5060: BYE sip:15033730000@outbound.vitelity.net SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK07263c34 Max-Forwards: 70 From: <sip:myusername@X.X.X.X:5160>;tag=as68734862 To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561 Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160 CSeq: 104 BYE User-Agent: FPBX-13.0.190.7(13.13.1) Proxy-Authorization: Digest username="myusername", realm="asterisk", algorithm=MD5, uri="sip:15033730000@outbound.vitelity.net", nonce="61f2cbea", response="78bab559a93fe55939070a02f664509a" X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- Scheduling destruction of SIP dialog '3e78dad92b6d54a7613617515445382a@X.X.X.X:5160' in 32000 ms (Method: INVITE) -- SIP/vitel-outbound-0000000d answered PJSIP/102-00000019 -- Channel SIP/vitel-outbound-0000000d joined 'simple_bridge' basic-bridge <d87be888-6633-4be3-9af7-0ed72301cf3f> -- Channel PJSIP/102-00000019 joined 'simple_bridge' basic-bridge <d87be888-6633-4be3-9af7-0ed72301cf3f> -- Channel SIP/vitel-outbound-0000000d left 'simple_bridge' basic-bridge <d87be888-6633-4be3-9af7-0ed72301cf3f> -- Channel PJSIP/102-00000019 left 'simple_bridge' basic-bridge <d87be888-6633-4be3-9af7-0ed72301cf3f> == Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on 'PJSIP/102-00000019' in macro 'dialout-trunk' == Spawn extension (from-internal, 5033730000, 6) exited non-zero on 'PJSIP/102-00000019' -- Executing [h@from-internal:1] Macro("PJSIP/102-00000019", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/102-00000019", "1?theend") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/102-00000019", "0?Set(CDR(recordingfile)=)") in new stack -- Executing [s@macro-hangupcall:4] Hangup("PJSIP/102-00000019", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/102-00000019' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/102-00000019' == MixMonitor close filestream (mixed) == End MixMonitor Recording PJSIP/102-00000019 Scheduling destruction of SIP dialog '3e78dad92b6d54a7613617515445382a@X.X.X.X:5160' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 64.2.142.216:5060: BYE sip:15033730000@outbound.vitelity.net SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK21083f8c Max-Forwards: 70 From: <sip:myusername@X.X.X.X:5160>;tag=as68734862 To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561 Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160 CSeq: 105 BYE User-Agent: FPBX-13.0.190.7(13.13.1) Proxy-Authorization: Digest username="myusername", realm="asterisk", algorithm=MD5, uri="sip:15033730000@outbound.vitelity.net", nonce="61f2cbea", response="78bab559a93fe55939070a02f664509a" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:64.2.142.216:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK07263c34;received=X.X.X.X From: <sip:myusername@X.X.X.X:5160>;tag=as68734862 To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561 Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160 CSeq: 104 BYE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:64.2.142.216:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK21083f8c;received=X.X.X.X From: <sip:myusername@X.X.X.X:5160>;tag=as68734862 To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561 Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160 CSeq: 105 BYE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '3e78dad92b6d54a7613617515445382a@X.X.X.X:5160' Method: INVITE
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