Outbound call to SIP Trunk with PJSIP

jneuendorf · December 12, 2023, 9:10pm

Hi,

I have an Asterisk 20 instance running on a VPS without NAT and I am using PJSIP as channel driver.
It is registered to a SIP Trunk and inbound calls seem to work.
Currently, I’m only interested in outbound calls to real phones (i.e. mobile phones) though. Thus, there are no local extensions – just Asterisk and the SIP Trunk.

For testing purposes I simply try to originate a call from the CLI (so no dial plan) to my mobile phone.
However, neither is my phone ringing nor does the SIP Trunk Provider show any outbound calls.
And I cannot see any errors or traffic:

localhost*CLI> pjsip set logger on
PJSIP Logging enabled
localhost*CLI> channel originate PJSIP/49157xxxxxxxx@trunk application Playback hello-world
    -- Called 49157xxxxxxxx@trunk

I also have a full logging (full => notice,warning,error,debug,verbose(5),dtmf) which outputs the following:

[Dec 12 20:23:59] DEBUG[142561] chan_pjsip.c:  49157xxxxxxxx@trunk Topology:  <0:audio-0:audio:sendrecv (slin)>
[Dec 12 20:23:59] DEBUG[142521] chan_pjsip.c:  49157xxxxxxxx@trunk
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c: trunk 49157xxxxxxxx Topology:  <0:audio-0:audio:sendrecv (slin)>
[Dec 12 20:23:59] DEBUG[142521] chan_pjsip.c:  trunk
[Dec 12 20:23:59] DEBUG[142521] chan_pjsip.c:  Direct media no glare mitigation
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session/pjsip_session_caps.c: 'trunk' Caps for outgoing audio call with pref 'remote_merge' - remote: (slin) local: (alaw|ulaw) joint: (alaw|ulaw)
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  
[Dec 12 20:23:59] DEBUG[142521] chan_pjsip.c:  
[Dec 12 20:23:59] DEBUG[142561] chan_pjsip.c:  trunk
[Dec 12 20:23:59] DEBUG[142561] channel_internal_api.c:  <initializing>: Formats: (none)
[Dec 12 20:23:59] DEBUG[142561] channel_internal_api.c:  Channel is being initialized or destroyed
[Dec 12 20:23:59] DEBUG[142561] stasis.c: Creating topic. name: channel:1702412639.2, detail: 
[Dec 12 20:23:59] DEBUG[142561] stasis.c: Topic 'channel:1702412639.2': 0x7f5f68003410 created
[Dec 12 20:23:59] DEBUG[142561] channel.c: Channel 0x7f5f68001cd0 'PJSIP/trunk-00000002' allocated
[Dec 12 20:23:59] DEBUG[142561] chan_pjsip.c:  Topology:  <0:audio-0:audio:sendrecv (alaw|ulaw)> Formats: (alaw|ulaw)
[Dec 12 20:23:59] DEBUG[142561] chan_pjsip.c:  Compatible? yes
[Dec 12 20:23:59] DEBUG[142561] channel_internal_api.c:  PJSIP/trunk-00000002: MultistreamFormats: (alaw|ulaw)
[Dec 12 20:23:59] DEBUG[142561] channel_internal_api.c:  Set native formats but not topology
[Dec 12 20:23:59] DEBUG[142561] channel_internal_api.c:  PJSIP/trunk-00000002:  <0:audio-0:audio:sendrecv (alaw|ulaw)>
[Dec 12 20:23:59] DEBUG[142561] channel_internal_api.c:  Used provided topology
[Dec 12 20:23:59] DEBUG[142561] chan_pjsip.c:  
[Dec 12 20:23:59] DEBUG[142561] chan_pjsip.c:  Channel: PJSIP/trunk-00000002
[Dec 12 20:23:59] DEBUG[142495] threadpool.c: Increasing threadpool stasis/pool's size by 1
[Dec 12 20:23:59] DEBUG[142630] chan_pjsip.c:  PJSIP/trunk-00000002 Topology:  <0:audio-0:audio:sendrecv (alaw|ulaw)>
[Dec 12 20:23:59] DEBUG[142630] chan_pjsip.c:  'call' task pushed
[Dec 12 20:23:59] VERBOSE[142630] dial.c: Called 49157xxxxxxxx@trunk
[Dec 12 20:23:59] DEBUG[142521] chan_pjsip.c:  PJSIP/trunk-00000002 Topology:  <0:audio-0:audio:sendrecv (alaw|ulaw)>
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  PJSIP/trunk-00000002
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  PJSIP/trunk-00000002
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  PJSIP/trunk-00000002: Processing streams
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  PJSIP/trunk-00000002 Adding position 0
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  Creating new media session
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  Setting media session as default for audio
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  Done
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  PJSIP/trunk-00000002 Stream: 0:audio-0:audio:sendrecv (alaw|ulaw)
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_sdp_rtp.c:  PJSIP/trunk-00000002 Type: audio 0:audio-0:audio:sendrecv (alaw|ulaw)
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_sdp_rtp.c: Transport transport-udp bound to 0.0.0.0: Using it for RTP media.
[Dec 12 20:23:59] DEBUG[142521] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f5fcc2592b0'
[Dec 12 20:23:59] DEBUG[142521] res_rtp_asterisk.c: (0x7f5fcc2592b0) RTP allocated port 10098
[Dec 12 20:23:59] DEBUG[142521] rtp_engine.c: RTP instance '0x7f5fcc2592b0' is setup and ready to go
[Dec 12 20:23:59] DEBUG[142521] acl.c: Multiple addresses. Using the first only
[Dec 12 20:23:59] DEBUG[142521] res_rtp_asterisk.c: () RTCP setup on RTP instance
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_sdp_rtp.c:  RC: 1
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  Handled
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  PJSIP/trunk-00000002: Adding bundle groups (if available)
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  PJSIP/trunk-00000002: Copying connection details
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  PJSIP/trunk-00000002
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  PJSIP/trunk-00000002: Method is INVITE
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_geolocation.c:  PJSIP/trunk-00000002
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_geolocation.c:  PJSIP/trunk-00000002: Endpoint has no geoloc_outgoing_call_profile. Skipping.
[Dec 12 20:23:59] DEBUG[142521] res_pjsip_session.c:  PJSIP/trunk-00000002
[Dec 12 20:23:59] DEBUG[142521] chan_pjsip.c:  RC: 0

Note that

ICE support is disabled (Can't make internal/outbound calls with PJSIP - #6 by jcolp)
I also tried setting from_user and from_domain on the endpoint in the pjsip.conf (How do I make an outbound call to a mobile number? - #11 by zscore)
The RC: 1 seems to be ok if I read the code correctly (create_outgoing_sdp_stream’s return value is checked against < 0 in https://github.com/asterisk/asterisk/blob/master/res/res_pjsip_session.c#L5059).

What I don’t understand: Why can’t I see any SIP traffic? Any error message or SIP package would be a hint towards a solution but no output makes debugging difficult.

I appreciate any idea and thank you for spending your time for me!

Regards, Jim

Resources I’ve look through before posting yet another question:

penguinpbx · January 4, 2024, 5:58pm

Maybe try removing the “(5)” so it is just “verbose” ?

What is the output of CLI command “logger show channels” ?

jneuendorf · January 12, 2024, 9:36am

Hey, thanks for your reply.

It seems there was something wrong with my pjsip.conf and extensions.conf which caused a long delay, so I guess the logging didn’t get far enough so see traffic.

I think the verbose(5) is correct (Logging Configuration - Asterisk Documentation) and verbose enough (https://docs.asterisk.org/Operation/Logging/Verbosity-in-Core-and-Remote-Consoles/).

Thanks again for your time, Jim

system · February 11, 2024, 9:36am

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