SIP trunk incoming calls rejected

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SIP trunk incoming calls rejected

Postby frodefi » Wed Mar 28, 2007 11:38 am

I am using a SIP service provider in Norway called Phonect(.no).

I can call out from the PBX, I can call internally. But incoming calls are rejected by the Asterisk system.

I have set an "All unmatched incoming calls" for incoming calls.

What is wrong...?

Here is debug-info (SIP/2.0 404 Not Found is somewhere down the line here):


--- (9 headers 0 lines) ---

Really destroying SIP dialog 'daecc16-bf8e7026@' Method: ACK


<--- SIP read from --->

INVITE sip:21548900@ SIP/2.0

Via: SIP/2.0/UDP;branch=z9hG4bK9296e31a5a659eeb851d126819ff8ae2;rport

Max-Forwards: 70

From: 21543404 <sip:21543404@>;tag=a5ad1eff5ba0731b27559760dd9a9f5c

To: <sip:21548900@>

Call-ID: 6db43373-b6c66d1a@

CSeq: 200 INVITE

Contact: Anonymous <sip:>

Expires: 300

User-Agent: Sippy

cisco-GUID: 1266001414-4118920608-4210516184-2409223586

h323-conf-id: 1266001414-4118920608-4210516184-2409223586

Content-Length: 447

Content-Type: application/sdp


o=Sippy 185131084 0 IN IP4


t=0 0


m=audio 16438 RTP/AVP 8 0 2 4 18 96 97 98 101

c=IN IP4

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:4 G723/8000

a=rtpmap:18 G729a/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15





--- (14 headers 20 lines) ---

Sending to : 5061 (NAT)

Using INVITE request as basis request - 6db43373-b6c66d1a@

Found no matching peer or user for ''

Found RTP audio format 8

Found RTP audio format 0

Found RTP audio format 2

Found RTP audio format 4

Found RTP audio format 18

Found RTP audio format 96

Found RTP audio format 97

Found RTP audio format 98

Found RTP audio format 101

Peer audio RTP is at port

Found description format PCMA for ID 8

Found description format PCMU for ID 0

Found description format G726-32 for ID 2

Found description format G723 for ID 4

Found description format G729a for ID 18

Found description format G726-40 for ID 96

Found description format G726-24 for ID 97

Found description format G726-16 for ID 98

Found description format telephone-event for ID 101

Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

Peer audio RTP is at port

Looking for 21548900 in default (domain

<--- Reliably Transmitting (NAT) to --->

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP;branch=z9hG4bK9296e31a5a659eeb851d126819ff8ae2;received=;rport=5061

From: 21543404 <sip:21543404@>;tag=a5ad1eff5ba0731b27559760dd9a9f5c

To: <sip:21548900@>;tag=as23d1080f

Call-ID: 6db43373-b6c66d1a@

CSeq: 200 INVITE

User-Agent: Asterisk PBX


Supported: replaces

Content-Length: 0
Posts: 12
Joined: Wed Mar 28, 2007 11:24 am

Same issue

Postby LrdCasimir » Wed Apr 11, 2007 8:46 am

I'm having the same issue. I've tried manually changing the inbound calling rules and I'm getting the same message. Am I looking in the right place?
Posts: 1
Joined: Wed Apr 11, 2007 8:23 am


Postby frodefi » Mon Apr 16, 2007 11:59 pm

I found that the following line in the SIP-trunk needed to be changed from empty to:

insecure = port,invite

I hope this helps you too!
Posts: 12
Joined: Wed Mar 28, 2007 11:24 am

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