Calls originating EXTERNAL to my network drop after a number of seconds

BostonDan

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I am having an issue where outgoing calls that originate from my softphone (Zoiper) EXTERNAL to my home network drop after a range of 5-30 seconds. If I am INTERNAL to my home network (wifi), calls stay connected. Also, incoming calls to my Zoiper when I am EXTERNAL to my home network work fine.

Using asterisk -rvvvvv I see:

== Spawn extension (macro-dialout-trunk, s, 31) exited non-zero on 'SIP/1001-00000000' in macro 'dialout-trunk'
== Spawn extension (from-internal, 5865715086, 5) exited non-zero on 'SIP/1001-00000000'
-- Executing [h@from-internal:1] Macro("SIP/1001-00000000", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/1001-00000000", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/1001-00000000", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/1001-00000000", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/1001-00000000' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-00000000'

so somehow dialout-trunk is exiting with a non-zero response that immediately hangs up the call.

Does anybody have any ideas?

Cheers,
B.D.
 

atsak

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session-timers=refuse in the SIP settings

See if that does anything.
 

BostonDan

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Nope. Didn't work, but thanks for the suggestion. I will have to wireshark the traffic this weekend when I have time and try to figure out what is not being sent.

Also, the actual command is session-times=Reject. The three possibilities provided for session-times are Accept, Originate, and Reject.

Again, I do appreciate the suggestion, and figured something else is going on.


Cheers,
B.D.
 

BostonDan

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Ok, the strange thing is that if I just use CHAN_SIP or CHAN_PJSIP without TLS, the calls are fine. If I use TLS the calls drop after some time (anywhere from 8 seconds to just over a minute). Really strange. My firewalls seem to be working as I have 2-way voice, and calls go through fine.

So frustrating. If anybody has any ideas, I would be willing to try anything at this point for secure calls!

Cheers,
B.D.
 

BostonDan

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Ok. After much searching I discovered that when a non-standard port is used then external devices sometimes have issues. To combat this, I edited sip.conf and added the line

externtlsport=<external TLS Port to use>.

Now my calls don't drop.

One final problem to figure out now is packet loss. Sometimes the line is ok (clear voice) and sometimes it has significant drop. But I almost have this working. Woot woot.

Cheers,
B.D.
 

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