SIP call disconnects after 20 seconds

Asterisk Asterisk Support
1

I have a site to site VPN, with a cudatel system on 1 side, and a Asterisk system on the other side, when I make a call the call lasts for less thank 22 seconds and hangs up. any ideas?

Using SIP RTP CoS mark 5
– Called SIP/1518
– SIP/1518-0000d42a is ringing
– SIP/1518-0000d42a answered SIP/BPS-0000d429
– Executing [s@macro-auto-blkvm:1] Set(“SIP/1518-0000d42a”, “__MACRO_RESULT=”) in new stack
– Executing [s@macro-auto-blkvm:2] NoOp(“SIP/1518-0000d42a”, “Deleting: BLKVM/1518/SIP/BPS-0000d429 TRUE”) in new stack
– Executing [h@macro-dial-one:1] Macro(“SIP/BPS-0000d429”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/BPS-0000d429”, “1?endmixmoncheck”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] NoOp(“SIP/BPS-0000d429”, “End of MIXMON check”) in new stack
– Executing [s@macro-hangupcall:10] GotoIf(“SIP/BPS-0000d429”, “1?nomeetmemon”) in new stack
– Goto (macro-hangupcall,s,15)
– Executing [s@macro-hangupcall:15] NoOp(“SIP/BPS-0000d429”, “MEETME_RECORDINGFILE=”) in new stack
– Executing [s@macro-hangupcall:16] GotoIf(“SIP/BPS-0000d429”, “1?noautomon”) in new stack
– Goto (macro-hangupcall,s,18)
– Executing [s@macro-hangupcall:18] NoOp(“SIP/BPS-0000d429”, “TOUCH_MONITOR_OUTPUT=”) in new stack
– Executing [s@macro-hangupcall:19] GotoIf(“SIP/BPS-0000d429”, “1?noautomon2”) in new stack
– Goto (macro-hangupcall,s,25)
– Executing [s@macro-hangupcall:25] NoOp(“SIP/BPS-0000d429”, “MONITOR_FILENAME=”) in new stack
– Executing [s@macro-hangupcall:26] GotoIf(“SIP/BPS-0000d429”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,29)
– Executing [s@macro-hangupcall:29] GotoIf(“SIP/BPS-0000d429”, “0?skipblkvm”) in new stack
– Executing [s@macro-hangupcall:30] NoOp(“SIP/BPS-0000d429”, “Cleaning Up Block VM Flag: BLKVM/1518/SIP/BPS-0000d429”) in new stack
– Executing [s@macro-hangupcall:31] NoOp(“SIP/BPS-0000d429”, "Deleting: BLKVM/1518/SIP/BPS-0000d429 ") in new stack
– Executing [s@macro-hangupcall:32] GotoIf(“SIP/BPS-0000d429”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,34)
– Executing [s@macro-hangupcall:34] Hangup(“SIP/BPS-0000d429”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 34) exited non-zero on ‘SIP/BPS-0000d429’ in macro ‘hangupcall’
== Spawn extension (macro-dial-one, h, 1) exited non-zero on ‘SIP/BPS-0000d429’
== Spawn extension (macro-dial-one, s, 37) exited non-zero on ‘SIP/BPS-0000d429’ in macro ‘dial-one’
== Spawn extension (macro-simple-dial, s, 6) exited non-zero on ‘SIP/BPS-0000d429’ in macro ‘simple-dial’
== Spawn extension (from-trunk, 1518, 16) exited non-zero on ‘SIP/BPS-0000d429’

2

Please supply the dialplan. If it from a third party, in particular FreePBX, please seek support from that third party.

Also, there is an insufficient logging level to see why the call is being dropped, but the names appearing the dialplan make me think the caller may be blacklisted - this is a function of the dialplan, not of Asterisk itself. I’d suggest a standard SIP debugging setup, full log enabled, verbose 5, debug 5 and “sip set debug on”, or the equivalent PJSIP command, issued.

When reporting issues with a timing element, please use the log files, as they contain time stamps, which are not available from screen scrapes.

3

So I was able to receive some more logs from my counterpart. I am failing registration because it appears it is recognizing my registration IP as the router IP, is there a way i can fix this?

  • Executing [1518@from-sip-external:1] NoOp(“SIP/10.0.0.2-000103ca”, “Received incoming SIP connection from unknown peer to 1518”) in new stack
    – Executing [1518@from-sip-external:2] Set(“SIP/10.0.0.2-000103ca”, “DID=1518”) in new stack
    – Executing [1518@from-sip-external:3] Goto(“SIP/10.0.0.2-000103ca”, “s,1”) in new stack
    – Goto (from-sip-external,s,1)
    – Executing [s@from-sip-external:1] GotoIf(“SIP/10.0.0.2-000103ca”, “0?checklang:noanonymous”) in new stack
    – Goto (from-sip-external,s,5)
    – Executing [s@from-sip-external:5] Set(“SIP/10.0.0.2-000103ca”, “TIMEOUT(absolute)=15”) in new stack
    Channel will hangup at 2016-01-08 13:59:59.232 CST.
    – Executing [s@from-sip-external:6] Answer(“SIP/10.0.0.2-000103ca”, “”) in new stack
    – Executing [s@from-sip-external:7] Wait(“SIP/10.0.0.2-000103ca”, “2”) in new stack
    – Executing [s@from-sip-external:8] Playback(“SIP/10.0.0.2-000103ca”, “ss-noservice”) in new stack
    – <SIP/10.0.0.2-000103ca> Playing ‘ss-noservice.gsm’ (language ‘en’)
    == Spawn extension (from-sip-external, s, 8) exited non-zero on ‘SIP/10.0.0.2-000103ca’
    – Executing [h@from-sip-external:1] Hangup(“SIP/10.0.0.2-000103ca”, “”) in new stack
    == Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/10.0.0.2-000103ca’

Jan 8 14:19:44] NOTICE[3402] chan_sip.c: Registration from ‘sip:test@10.0.0.2:5060’ failed for ‘10.0.0.1:65476’ - No matching peer found
[Jan 8 14:19:46] NOTICE[3402] chan_sip.c: Registration from ‘sip:test@10.0.0.2’ failed for ‘10.0.0.1:65476’ - No matching peer found
[Jan 8 14:19:46] NOTICE[3402] chan_sip.c: Registration from ‘sip:test@10.0.0.2’ failed for ‘10.0.0.1:65476’ - No matching peer found
[Jan 8 14:19:47] NOTICE[3402] chan_sip.c: Registration from ‘sip:test@10.0.0.2’ failed for ‘10.0.0.1:65476’ - No matching peer found
[Jan 8 14:19:47] NOTICE[3402] chan_sip.c: Registration from ‘sip:test@10.0.0.2:5060’ failed for ‘10.0.0.1:65476’ - No matching peer found
[Jan 8 14:19:48] NOTICE[3402] chan_sip.c: Registration from ‘sip:test@10.0.0.2’ failed for ‘10.0.0.1:65476’ - No matching peer found
[Jan 8 14:19:49] NOTICE[3402] chan_sip.c: Registration from ‘sip:test@10.0.0.2’ failed for ‘10.0.0.1:65476’ - No matching peer found
[Jan 8 14:19:50] VERBOSE[9031] res_musiconhold.c: – Started music on hold, class ‘default’, on SIP/5511661500-000103d5
[Jan 8 14:19:50] WARNING[9031] mp3/interface.c: Junk at the beginning of frame 49443303
[Jan 8 14:19:50] NOTICE[3402] chan_sip.c: Registration from ‘sip:test@10.0.0.2’ failed for ‘10.0.0.1:65476’ - No matching peer found

4

On Asterisk the “externally visible” address and port number to be used when talking
to a host outside the NAT. This information is derived by one of the
following (mutually exclusive) config file parameters:

a. "externaddr = hostname[:port]" specifies a static address[:port] to
 be used in SIP and SDP messages.
5

Thanks resolved my issues I was having.

Another question, is there a way to get the Asterisk to transmit the 4 digit ext over the SIP trunk, for when they call me from the Asterisk system, it shows the username that is calling me, but the caller number is showing unknown, and the same thing when I call them from my system.

6

If you are simple phone, delete the fromuser from the sip.conf entry for the destination. I seem to remember this is a known bug in some versions of FreePBX, but FreePBX is not supported here.

If you really need fromuser, investigate whether the sendrpid parameter is relevant to your situation.

The default behaviour of Asterisk is to forward the incoming caller ID to the outgoing side.