Tel: URI to extension

Hey Guys,
is This a Asterisk bug or am I doing something wrong?
Asterisk ver 20.2.0

when a tel: URI is sent to asterisk, it tries to match the whole URI in the Dial Plan.

Call (TCP:10.90.250.52:45266) to extension ‘3345551234;phone-context=ims.mnc435.mcc311.3gppnetwork.org’ rejected because extension not found in context

<— Received SIP request (2453 bytes) from TCP:10.90.250.52:45266 —>
INVITE sip:3345551234;phone-context=ims.mnc435.mcc311.3gppnetwork.org@softswitch.ims.mnc435.mcc311.3gppnetwork.org:5060;Transport=tcp SIP/2.0
Record-Route: sip:10.90.250.52;transport=tcp;r2=on;lr=on;ftag=OqV7j2HpOO;did=0f2.e1c2
Record-Route: sip:10.90.250.52;transport=tcp;r2=on;lr=on;ftag=OqV7j2HpOO;did=0f2.e1c2
Record-Route: sip:mo@pcscf.ims.mnc435.mcc311.3gppnetwork.org:5060;r2=on;lr=on;ftag=OqV7j2HpOO;rm=8;did=0f2.5211
Record-Route: sip:mo@pcscf.ims.mnc435.mcc311.3gppnetwork.org;transport=tcp;r2=on;lr=on;ftag=OqV7j2HpOO;rm=8;did=0f2.5211
Accept-Contact: *;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”
Supported: 100rel,path,precondition,replaces,timer
From: tel:3342012832;tag=OqV7j2HpOO
To: tel:3345551234;phone-context=ims.mnc435.mcc311.3gppnetwork.org
Call-ID: 9Wof3M0a3be8RURGPcMUD25L
Session-ID: 8f0a3e8451995cfd6711b55536903da9
Contact: sip:10.46.1.14:5060;alias=10.46.1.14~5060~2;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”
CSeq: 1 INVITE
Via: SIP/2.0/TCP 10.90.250.52;branch=z9hG4bK7231.c2aa450b9ec677a9ece758800023e0d3.0;i=b
Via: SIP/2.0/TCP pcscf.ims.mnc435.mcc311.3gppnetwork.org:5060;received=10.90.250.50;branch=z9hG4bK7231.b460ee2bd121f0aa93ba8049aae717d8.0;i=263
Via: SIP/2.0/TCP 10.46.1.14:5060;received=10.46.1.14;branch=z9hG4bKhv6FC6byrD0HFxO;rport=5060
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
Max-Forwards: 68
User-Agent: iOS/16.4 iPhone
P-Access-Network-Info: 3GPP-E-UTRAN-FDD;utran-cell-id-3gpp=31143500010000101
Content-Type: application/sdp
Content-Length: 576
P-Charging-Vector: icid-value=4956530A5AFA32C51500001504000000; icid-generated-at=10.90.250.50
Feature-Caps: *;+g.3gpp.trf=“sip:trf.ims.mnc435.mcc311.3gppnetwork.org;lr
P-Visited-Network-ID: ims.mnc435.mcc311.3gppnetwork.org
P-Asserted-Identity: <sip:3342012832@>

v=0
o=tel:3342012832 1680422933 1680422933 IN IP4 10.90.250.59
s=-
c=IN IP4 10.90.250.59
t=0 0
m=audio 34148 RTP/AVP 104 110 102 108 105 100
a=maxptime:240
a=des:qos optional local sendrecv
a=curr:qos local none
a=des:qos optional remote sendrecv
a=curr:qos remote none
a=rtpmap:104 AMR-WB/16000
a=rtpmap:110 AMR-WB/16000
a=fmtp:110 octet-align=1
a=rtpmap:102 AMR/8000
a=rtpmap:108 AMR/8000
a=fmtp:108 octet-align=1
a=rtpmap:105 telephone-event/16000
a=fmtp:105 0-15
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=rtcp:34149
a=ptime:20

[Apr 2 08:08:53] NOTICE[150116]: res_pjsip_session.c:4000 new_invite: scscf.ims.mnc435.mcc311.3gppnetwork.org: Call (TCP:10.90.250.52:45266) to extension ‘3345551234;phone-context=ims.mnc435.mcc311.3gppnetwork.org’ rejected because extension not found in context ‘mobile’.
<— Transmitting SIP response (1049 bytes) to TCP:10.90.250.52:45266 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.90.250.52;rport=45266;received=10.90.250.52;branch=z9hG4bK7231.c2aa450b9ec677a9ece758800023e0d3.0;i=b
Via: SIP/2.0/TCP pcscf.ims.mnc435.mcc311.3gppnetwork.org:5060;received=10.90.250.50;branch=z9hG4bK7231.b460ee2bd121f0aa93ba8049aae717d8.0;i=263
Via: SIP/2.0/TCP 10.46.1.14:5060;rport=5060;received=10.46.1.14;branch=z9hG4bKhv6FC6byrD0HFxO
Record-Route: sip:10.90.250.52:45266;transport=TCP;lr;r2=on;ftag=OqV7j2HpOO;did=0f2.e1c2
Record-Route: sip:10.90.250.52;transport=tcp;lr;r2=on;ftag=OqV7j2HpOO;did=0f2.e1c2
Record-Route: sip:mo@pcscf.ims.mnc435.mcc311.3gppnetwork.org:5060;lr;r2=on;ftag=OqV7j2HpOO;rm=8;did=0f2.5211
Record-Route: sip:mo@pcscf.ims.mnc435.mcc311.3gppnetwork.org;transport=tcp;lr;r2=on;ftag=OqV7j2HpOO;rm=8;did=0f2.5211
Call-ID: 9Wof3M0a3be8RURGPcMUD25L
From: tel:3342012832;tag=OqV7j2HpOO
To: tel:3345551234;phone-context=ims.mnc435.mcc311.3gppnetwork.org;tag=388adfaa-c08f-4e36-b2e4-6449892b4527
CSeq: 1 INVITE
Server: Asterisk PBX 20.2.0
Content-Length: 0

It’s receiving a SIP URI in the request URI, not a tel uri. What you’ve provided would constitute the user portion and so it would be used as-is.

I see that now, Thank you for the reply.

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