Can't call in to Extension

Hello! Having some issues with a new extension at a location connected via a site to site VPN. New extension (400) can call outbound to external numbers and can call any other extension at different sites. When we try to call inbound to that extension, it goes straight to the unavailable voicemail without ringing. I have checked the extension is registered and it’s showing online. DND is off but there has to be something else I’m missing. Here is the error in the SIP trace log on the phone. I replaced IP info in the error message below.

Received from udp:%phonesystemIP%:5060 at 11/4/2023 12:48:53:614 (1151 bytes):

INVITE sip:[email protected]:5060 SIP/2.0
From: "%ext% sip:192@%phonesystemIP%;tag=30cf408c-0710-4d3b-9cf2-ebbeaa28d2dd
To: sip:400@%externalIP%
Call-ID: 0183ded4-12b2-414f-b11a-8401be51b132
CSeq: 4068 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: “Matt Voight” sip:192@%phonesystemIP%
Call-Info: ;answer-after=0
Alert-Info: http://127.0.0.1;info=Ring Answer
Max-Forwards: 70
User-Agent: FPBX-16.0.28(16.28.0)
Content-Type: application/sdp
Content-Length: 343
Via: SIP/2.0/UDP %phonesystemIP%;branch=z9hG4bKPj2840d717-3492-49b0-816d-974fcc616356
Contact: sip:asterisk@%phonesystemIP%

v=0
o=- 1425157323 1425157323 IN IP4 %phonesystemIP%
s=Asterisk
c=IN IP4 %phonesystemIP%
t=0 0
m=audio 15626 RTP/AVP 0 8 111 9 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
Sent to udp:%phonesystemIP%:5060 at 11/4/2023 12:48:53:618 (532 bytes):

SIP/2.0 404 Not found
Via: SIP/2.0/UDP %phonesystemIP%;branch=z9hG4bKPj2840d717-3492-49b0-816d-974fcc616356
From: “%ext%” sip:192@%phonesystemIP%;tag=30cf408c-0710-4d3b-9cf2-ebbeaa28d2dd
To: sip:400@%externalIP%
Call-ID: 0183ded4-12b2-414f-b11a-8401be51b132
CSeq: 4068 INVITE
User-Agent: snom870/8.7.3.25
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

Is it a pjsip extension? If so what do you see for the extension when you do pjsip show contacts?

are you able to ping that phone at the other network from the phone system at all?

INVITE sip:[email protected]:5060 SIP/2.0

Contact: <sip:asterisk@%phonesystemIP%>

You appear not to specified the VPN as a local network.

INVITE sip:[email protected]:5060 SIP/2.0

SIP/2.0 404 Not found

The phone doesn’t think it is 400, which is strange given this is, presumably, what it sent in the Contact header in the REGISTER request.

INVITE sip:[email protected]:5060 SIP/2.0

To: sip:400@%externalIP%

Or is there a SIP ALG, in a router, messing things up, as I don’t think you should be able to get the To and request URI differing. What does the pjsip set logger on logging, at the Asterisk end, show?

SIP ALG may be enabled, I’ll check that out along with the logs.

For specifying the local network, you mean in general SIP settings or somewhere else?

Yes, it’s a PJSIP extension. This is what it shows -
Contact: 400/sip:400@%externalIP:5060 4a565f244b Avail 37.322

Although, it’s the only extension that doesn’t show something like this after the SIP port - ;line=435yth0f;x

In your original post you are stating that you are using a VPN to connect between the sites but the extension is registering with an external IP (you might want to blank that out for security reasons). I would look at your site-to-site VPN to ensure that traffic is actually routing over that instead of over the WAN interface.

Thanks. Should’ve mentioned that the PBX is hosted externally through Freepbxhosting. We don’t have our other sites added under internal networks, however, and they are working.

Unless there is another place where the internal networks are added I’m missing. All of our extensions show the external address when I do those cmds.

Pjsip show aor 400 shows this -
Aor: 400 1
Contact: 400/sip:400@%externalIP%:5060 4a565f244b Avail 37.129

PJsip show aor 192 (working extension at another site) shows this -
Aor: 192 1
Contact: 192/sip:192@%externalIP%:1036;line=9w6jrzvo bfe20a53fa Avail 33.045

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