Not able to make calls

[junalmeida]

Hi,
Using the "almost" standard configuration from suggestions, I still cannot make calls. Here are some logs during intra-extensions calls. I appreciate if you can point out if its a mistake on my configuration.

-- Executing [s@macro-dial-one:51] NoOp("SIP/400-000030f4", "") in new stack
    -- Executing [s@macro-dial-one: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 [s@func-apply-sipheaders:1] NoOp("SIP/500-000030f5", "Applying SIP Headers to channel") in new stack
    -- Executing [s@func-apply-sipheaders:2] Set("SIP/500-000030f5", "SIPHEADERKEYS=") in new stack
    -- Executing [s@func-apply-sipheaders:3] While("SIP/500-000030f5", "0") in new stack
    -- Jumping to priority 6
    -- Executing [s@func-apply-sipheaders: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 [h@from-internal:1] Macro("SIP/400-000030f4", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/400-000030f4", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("SIP/400-000030f4", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] NoOp("SIP/400-000030f4", "SIP/500-000030f5 monior file= ") in new stack
    -- Executing [s@macro-hangupcall: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 [s@macro-hangupcall: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

[junalmeida]

Any suggestions on this?

[dinocore1]

I get the same problem. It seems like the call starts, but no RTP packets are ever transmitted. Is there something wrong with the docker networking configuration that is blocking RTP packets?

[tiredofit]

Here are the ports I have defined in my docker-compose.yml:

    ports:
      - 4445:4445
      - 5060:5060/udp
      - 5061:5061
      - 18000-18100:18000-18100/udp
      - 8089:8089

And here are the RTP environment variables I have defined

      - RTP_START=18000
      - RTP_FINISH=18100

We're currently in production with no issues relating to RTP at this time..

[junalmeida]

@tiredofit I had to start over from stratch and the things works now. Just a little tip: the default implementation of SIP now uses 5060 in udp, but your docker-compose.yml sample maps only tcp ports.

[kirbah]

Spent entire evening trying to configure everything. Default port in the example docker compose will be great to update to use UDP that is default configuration

ports:

• 5060:5060/udp