Cannot make intra-extension calls

Cannot make calls between extensions on freepbx. Here is my environment:
freepbx and asterisk running in a docker, at a simple ubuntu machine on azure.

5060, 5160 and 10000-20000 ports are forwarded from the public ip to vm, and from vm to docker.

Here is the log of a call:

-- Executing [[email protected]:51] NoOp("SIP/400-000030f4", "") in new stack
-- Executing [[email protected]:52] Dial("SIP/400-000030f4", "SIP/500,,HhTtrIb(func-apply-sipheaders^s^1)") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- SIP/500-000030f5 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [[email protected]:1] NoOp("SIP/500-000030f5", "Applying SIP Headers to channel") in new stack
-- Executing [[email protected]:2] Set("SIP/500-000030f5", "SIPHEADERKEYS=") in new stack
-- Executing [[email protected]:3] While("SIP/500-000030f5", "0") in new stack
-- Jumping to priority 6
-- Executing [[email protected]:7] Return("SIP/500-000030f5", "") in new stack
  == Spawn extension (from-internal, 500, 1) exited non-zero on 'SIP/500-000030f5'
-- SIP/500-000030f5 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called SIP/500
-- Connected line update to SIP/400-000030f4 prevented.
-- Connected line update to SIP/400-000030f4 prevented.
-- SIP/500-000030f5 is ringing
   > 0x7fe08c009f90 -- Strict RTP learning after remote address set to: 10.0.4.150:16400
-- Connected line update to SIP/400-000030f4 prevented.
-- SIP/500-000030f5 answered SIP/400-000030f4
-- Channel SIP/500-000030f5 joined 'simple_bridge' basic-bridge <91c72704-da14-4652-a996-cdadcc492d49>
-- Channel SIP/400-000030f4 joined 'simple_bridge' basic-bridge <91c72704-da14-4652-a996-cdadcc492d49>
[2018-04-08 10:28:03] NOTICE[5152]: chan_sip.c:29611 check_rtp_timeout: Disconnecting call 'SIP/400-000030f4' for lack of RTP activity in 31 seconds
[2018-04-08 10:28:03] NOTICE[5152]: chan_sip.c:29611 check_rtp_timeout: Disconnecting call 'SIP/500-000030f5' for lack of RTP activity in 31 seconds
-- Channel SIP/400-000030f4 left 'simple_bridge' basic-bridge <91c72704-da14-4652-a996-cdadcc492d49>
-- Channel SIP/500-000030f5 left 'simple_bridge' basic-bridge <91c72704-da14-4652-a996-cdadcc492d49>
  == Spawn extension (macro-dial-one, s, 52) exited non-zero on 'SIP/400-000030f4' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/400-000030f4' in macro 'exten-vm'
  == Spawn extension (from-internal, 500, 2) exited non-zero on 'SIP/400-000030f4'
-- Executing [[email protected]:1] Macro("SIP/400-000030f4", "hangupcall") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/400-000030f4", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [[email protected]:3] ExecIf("SIP/400-000030f4", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [[email protected]:4] NoOp("SIP/400-000030f4", "SIP/500-000030f5 monior file= ") in new stack
-- Executing [[email protected]:5] AGI("SIP/400-000030f4", "attendedtransfer-rec-restart.php,SIP/500-000030f5,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
-- <SIP/400-000030f4>AGI Script attendedtransfer-rec-restart.php completed, returning 0
-- Executing [[email protected]:6] Hangup("SIP/400-000030f4", "") in new stack
  == Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/400-000030f4' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/400-000030f4'
[2018-04-08 10:28:04] WARNING[5152]: chan_sip.c:4077 retrans_pkt: Retransmission timeout reached on transmission lC2NwJbGosBXLyvMRQHgUQ.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response

Current Asterisk Version: 14.7.6

Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
200                       (Unspecified)                            D  Yes        Yes         A  0        UNKNOWN
300                       (Unspecified)                            D  Yes        Yes         A  0        UNKNOWN
400/400                   yyy.yyy.2.80                             D  Yes        Yes         A  34193    OK (23 ms)
500/500                   yyy.yyy.2.80                             D  Yes        Yes         A  1024     OK (25 ms)
600/600                   xxx.xxx.164.9                            D  Yes        Yes         A  5060     OK (23 ms)
700                       (Unspecified)                            D  Yes        Yes         A  0        UNKNOWN
telemar/telemar           172.17.0.1                               D  Yes        Yes            38001    OK (22 ms)
7 sip peers [Monitored: 4 online, 3 offline Unmonitored: 0 online, 0 offline]

BTW, don’t know why the last peer (that is a trunk) shows an internal IP. It comes from the same network from 600/600 extension

you have to double check the network again.

Yes, but how? I checked the azure VM port forward and all seems OK for me. Also checked the redirection of docker. What else do you suggest?

Is FreePBX and/or the phones behind a NAT device?

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