PJSIP creating channel issues

Asterisk Asterisk SIP
1

Hello community.

This is my last resort. I am having hard time to make outbound and inbound calls. I went through all topics related to this and still can’t make calls.

I am not using registration (auth username and password) to connect with my provider.

My system:

Ubuntu 24.04
Asterisk 18.26.2

Here is my PJSIP:

[provider]
type=endpoint
transport=transport-udp-all
identify_by=ip
context=from-provider
disallow=all
allow=alaw
aors=provider

[provider]
type=identify
endpoint=provider
match=13.xx.xx.yy
match=13.xx.xx.zz

[provider]
type=aor
qualify_frequency=120
contact=sip:13.xx.xx.yy:5060
contact=sip:13.xx.xx.zz:5060

Both endpoint and contact is AVAILABLE:
Endpoint: provider Not in use 0 of inf
Aor: provider 0
Contact: provider/sip:13.xx.xx.yy:5060 2e196a4b5e Avail 23.846
Contact: provider/sip:13.xx.xx.zz:5060 4df0c23058 Avail 20.553
Transport: transport-udp-all udp 3 96 0.0.0.0:5060

Contact: provider/sip:13.xx.xx.yy:5060 4df0c23058 Avail 19.579
Contact: provider/sip:13.xx.xx.zz:5060 2e196a4b5e Avail 20.064

My diaplan:
[from-internal]
exten => _+X.,1,Noop(Call from:${CALLERID(num)} TO:${EXTEN})
exten => _+X.,n,Dial(PJSIP/${EXTEN}@provider)
→ When I use DIAL like that I am getting:
→ res_pjsip_outbound_authenticator_digest.c:554 digest_create_request_with_auth: Endpoint: ‘buzzeasy’: Authentication credentials not accepted by server.

exten => _+X.,n,Dial(PJSIP/provider/${EXTEN})
→ When I use DIAL like that I am getting:
→ ERROR[108705]: res_pjsip.c:993 ast_sip_create_dialog_uac: Endpoint ‘provider’: Could not create dialog to invalid URI ‘+4953127095100’. Is endpoint registered and reachable?
→ ERROR[108705]: chan_pjsip.c:2702 request: Failed to create outgoing session to endpoint ‘provider’
→ NOTICE[108866][C-0000000c]: app_dial.c:2768 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)

Periodicaly I am getting output on console:
ERROR[108705]: res_pjsip_outbound_authenticator_digest.c:450 digest_create_request_with_auth: Host: ‘13.xx.xx.yy:5060’: There were no auth ids available
ERROR[108705]: res_pjsip_outbound_authenticator_digest.c:450 digest_create_request_with_auth: Host: ‘13.xx.xx.zz:5060’: There were no auth ids available

I understand asterisk and SIP, been in voice for over a decade, but I am new in PJSIP.

Please, help me.

2

This is the correct format for Dial(), and the remote side challenged for authentication. The configuration you’ve provided doesn’t show any authentication. So I would say this is most likely configuration, but can’t give specifics.

  1. Is authentication required, if so configure it.
  2. If this still fails then they may require “from_user” to be set on the endpoint to your username.
3

Hello.

Thank you for your quick reply.

Provider claims that they are not using authentication. Auth is not required. So I do not have auth configured. Yes, my configuration is without authentication on purpose.

That tells me that providers end is configured wrong.

Thank you.

4

Hello.

So, provider claims that they have removed authentication and are not trying to authenticate. But I still can’t make call.

I am getting message:
ERROR[109046]: res_pjsip_outbound_authenticator_digest.c:450 digest_create_request_with_auth: Endpoint: ‘provider’: There were no auth ids available

5

They still challenged. You can see the actual SIP traffic using “pjsip set logger on” which will show you what is actually going on.

6

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