Sip trunk regitration failed

tli · May 6, 2025, 4:15pm

Hello Guys ,

facing trouble with pjsip trunk registration on my freepbx server , i am able to ping the sip provider ip addr from the asterisk console , but registration got failed , the debug did not give me more or not clear to me, any idea ?

david55 · May 6, 2025, 5:33pm

Please provide logs with pjsip set logger on in effect. Otherwise it could one of many misconfigurations or network problems.

tli · May 6, 2025, 5:46pm

Logs below , i keep receiving it

<— Transmitting SIP request (656 bytes) to UDP:sipproviderip@:5060 —>
REGISTER sip:sipproviderip@:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.245.100:5060;rport;branch=z9hG4bKPj2a95191c-76d3-44a8-a9fb-7f3ddb000cfe
From: sip:[email protected];tag=2aad6c46-eabe-4e4a-932c-309dff1faa41
To: sip:[email protected]
Call-ID: 4492bd89-de3a-4315-9e01-2b484c677886
CSeq: 56045 REGISTER
Contact: sip:[email protected]:5060;line=tulwhem
Expires: 3600
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Route: sip:hai.voip.digicelgroup.com:5060
Max-Forwards: 70
User-Agent: FPBX-16.0.40.13(18.16.0)
Content-Length: 0

[2025-05-06 13:42:29] WARNING[14281]: res_pjsip_outbound_registration.c:1047 schedule_retry: No response received from ‘sip:hai.voip.digicelgroup.com:5060’ on registration attempt to ‘sip:[email protected]:5060’, retrying in ‘60’

david55 · May 6, 2025, 7:38pm

The request isn’t reaching the other end or their reply isn’t getting through.

You are sending private addresses in the Via and Contact headers. The rport option should make a reply possible, so this not necessarily the complete problem, but the Contact header will mean that the provider doesn’t know where it send incoming calls, unless they use a workround of using the source IP address instead.

Configure the correct public addresses, and local networks.

<--- Transmitting SIP request (656 bytes) to UDP:sipproviderip@:5060 --->
REGISTER sip:sipproviderip@:5060 SIP/2.0

There is a redaction error. @ shouldn’t appear in the first line, and should be followed by an address or domain in the second. If not a redaction error, the server URI is badly configured.

Stewart1 · May 7, 2025, 12:51am

Please try these settings. If it still won’t register, post new pjsip logger output.

Asterisk SIP Settings:
Confirm that External Address and Local Networks are correctly set.
On the pjsip tab, the entries for External IP Address, External Signaling Port and Local network for each transport should be left blank. If you have set them otherwise, please explain.
If you changed any of the above, after Submit and Apply Config you must restart Asterisk.

Trunk settings:
SIP Server: hai.voip.digicelgroup.com
Username: 50928190149
Secret: (password supplied by your provider)
Outbound Proxy:
sip:1.2.3.4\;lr\;hide
(Replace 1.2.3.4 with your sip provider IPv4 address. Note backslash semicolon in two places.)
Client URI: (leave blank)
Server URI: (leave blank)

tli · May 7, 2025, 2:10pm

Thanks for your comment , what about the upd port 5060 , do i need open it in the firewall side ?

david55 · May 7, 2025, 4:39pm

The port you use for SIP signalling needs to be open and port forwarded, if NATted, on the routers, etc. There is some limited ability for dynamic rules to be used, but it is best to use static rules.

Your configured RTP range also needs to be the same. Again dynamic rules may soemtimes work, but it is best to use static ones.

tli · May 12, 2025, 2:26pm

Hello guys,

I am posting a pacap file , sip flow diagram between my pbx and the sip provider ,
I remember a normal sip flow should be content the INVITE, TRYING , and the ACK
below are all i got , could you please give an idea of what could be the blocking point ,

but i am still able to ping the sip provider ip addr from the pbx , any idea.

david55 · May 12, 2025, 2:38pm

The normal sequence you describe is for a call, not a registration, and the 100 is optional, but a final response (200 for a success) is mandatory.

I don’t see a pcap file, but you aren’t getting any responses. That could be because the request isn’t reaching the other end, or because the reply isn’t reaching you. You need to narrow down which applies.

In the former case, you ought to get an ICMP destination unreachable, but many people have security policies that prevent those working.

HawaiianHopeOrg · May 14, 2025, 2:18am

What is the firewall you are using ? In pfSense, when you set the outbound 5060 port you have to make it static as pfSense will by default randomize the outbound port.

tli · May 20, 2025, 2:02pm

The blocking point was on firewall side , the ip addr of the sip trunk seat in different to the LAN , so i allow sip trunk @ in the firewall side it s all ok - I am still searching how i was able to ping it before.

Thank you !