TRY THIS internal calling broken chan_sip

Bill Dengler

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Hello,
Running Asterisk 13.6.0, Freepbx 12 on Incredible PBX ISO.
I have a few phones registered to the system, which is running on a VPS.
Using Pjsip, the phones can register, and internal calls pass successfully without issue.
Using chan_sip, phones appear to register, and outgoing calls work over the Vitelity pjsip trunk. The system answers incoming calls, but attempts to call an extension go straight to voicemail. Attempting to call from one extension to another results in music for a few seconds on the originating extension, then the call terminates, printing the following to the console:
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Extension Changed 702[ext-local] new state Ringing for Notify User
== Extension Changed 702[ext-local] new state Ringing for Notify User
-- Called SIP/702
-- Started music on hold, class 'default', on channel 'SIP/701-00000182'
-- Connected line update to SIP/701-00000182 prevented.
> 0xb310a0d8 -- Probation passed - setting RTP source address to my.local.ipv4.address:5004
[2015-11-21 22:36:06] WARNING[2745]: chan_sip.c:4009 retrans_pkt: Retransmission timeout reached on transmission [email protected]4.address:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2015-11-21 22:36:06] WARNING[2745]: chan_sip.c:4038 retrans_pkt: Hanging up call [email protected]4.address:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Extension Changed 702[ext-local] new state Idle for Notify User
== Extension Changed 702[ext-local] new state Idle for Notify User
== Everyone is busy/congested at this time (1:0/0/1)
-- Stopped music on hold on SIP/701-00000182
-- Executing [[email protected]:45] ExecIf("SIP/701-00000182", "0?MacroExit()") in new stack
-- Executing [[email protected]:46] ExecIf("SIP/701-00000182", "0?Set(DIALSTATUS=)") in new stack
-- Executing [[email protected]:47] GosubIf("SIP/701-00000182", "0?s-CHANUNAVAIL,1()") in new stack
-- Executing [[email protected]:48] MacroExit("SIP/701-00000182", "") in new stack
-- Executing [[email protected]:17] Set("SIP/701-00000182", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [[email protected]:18] GosubIf("SIP/701-00000182", "0?docfu,1()") in new stack
-- Executing [[email protected]:19] GosubIf("SIP/701-00000182", "0?docfb,1()") in new stack
-- Executing [[email protected]:20] Set("SIP/701-00000182", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [[email protected]:21] ExecIf("SIP/701-00000182", "0?MacroExit()") in new stack
-- Executing [[email protected]:22] GotoIf("SIP/701-00000182", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [[email protected]:1] GotoIf("SIP/701-00000182", "0?exit,1") in new stack
-- Executing [[email protected]:2] PlayTones("SIP/701-00000182", "congestion") in new stack
-- Executing [[email protected]:3] Congestion("SIP/701-00000182", "10") in new stack
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/701-00000182' in macro 'exten-vm'
== Spawn extension (from-internal, 702, 2) exited non-zero on 'SIP/701-00000182'
-- Executing [[email protected]:1] Hangup("SIP/701-00000182", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/701-00000182'
== Extension Changed 701[ext-local] new state Idle for Notify User
== Extension Changed 701[ext-local] new state Idle for Notify User
== Extension Changed 701[ext-local] new state Idle for Notify User 702

Any ideas?
 

wardmundy

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Code:
cd /usr/src/asterisk-13*
make
make install
amportal restart
 

Bill Dengler

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Code:
cd /usr/src/asterisk-13*
make
make install
amportal restart
Attempted to use the script at ~/upgrade-asterisk-to-current and a manual recompile (make clean/make/make install) but the same error still occurs.
According to the wikipage referenced in the console output, this appears to be a NAT issue (but if that were the case why does pjsip work)?
 

Bill Dengler

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Oct 4, 2014
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Attempted to use the script at ~/upgrade-asterisk-to-current and a manual recompile (make clean/make/make install) but the same error still occurs.
According to the wikipage referenced in the console output, this appears to be a NAT issue (but if that were the case why does pjsip work)?
Fixed with new install.
 

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