Really need help with a cisco router to asterisk box issue

Ok here is the setup and the problem … I’m a complete newb with asterisk so if you can help out that would be great

i have a cisco 881 and two fxs ports with two analog phones attached (ext 6100 and 6150)

They can call the asterisk box and each other no problems at all.

All the softphones I have can also communicate with asterisk box and the two analog phones and vice versa no problems.
I have configured a VOIP provider as the trunk and the softphones can all call outside numbers no problem at all (with 9 1 ) in front but the analog phones can’t at all. Below is what shows up in the asterisk logs.

[May 13 12:14:18] NOTICE[3407] chan_sip.c: Call from ‘’ to extension ‘9125xxxxxxxx’ rejected because extension not found.
[May 13 12:14:26] NOTICE[3407] chan_sip.c: Call from ‘’ to extension ‘9125xxxxxxxx’ rejected because extension not found.
[May 13 12:15:02] NOTICE[3407] chan_sip.c: Call from ‘’ to extension ‘9125xxxxxxxx’ rejected because extension not found.
[May 13 12:15:10] NOTICE[3407] chan_sip.c: Call from ‘’ to extension ‘9125xxxxxxxx’ rejected because extension not found.
[May 13 12:15:18] NOTICE[3407] chan_sip.c: Call from ‘’ to extension ‘9125xxxxxxxx’ rejected because extension not found.
[May 13 12:17:02] NOTICE[3407] chan_sip.c: Call from ‘’ to extension ‘9125xxxxxxxx’ rejected because extension not found.
[May 13 12:17:49] NOTICE[3407] chan_sip.c: Call from ‘’ to extension ‘9125xxxxxxxx’ rejected because extension not found.

Thanks in advance.

this is how it is looking on the console (ip’s and phone numbers obfuscated a bit):

<--- SIP read from 69.XX.XX.21:63647 --->
INVITE sip:91250XXXXXXX@69.XX.XX.23:5060 SIP/2.0
Via: SIP/2.0/UDP 69.XX.XX.21:5060;branch=z9hG4bKA1775
Remote-Party-ID: "Beta" <sip:69.XX.XX.21>;party=calling;screen=no;privacy=off
From: "unknown" <sip:69.XX.XX.23>;tag=1BF378-CC0
To: <sip:91250XXXXXXX@69.XX.XX.23>
Date: Tue, 13 May 2008 14:55:34 GMT
Call-ID: 79733A2F-203311DD-80618575-A42EC5B3@69.XX.XX.21
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
Cisco-Guid: 1994434739-540217821-2153547125-2754528691
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1210690534
Contact: <sip:69.XX.XX.21:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 188

v=0
o=CiscoSystemsSIP-GW-UserAgent 8246 7594 IN IP4 69.XX.XX.21
s=SIP Call
c=IN IP4 69.XX.XX.21
t=0 0
m=audio 18410 RTP/AVP 0
c=IN IP4 69.XX.XX.21
a=rtpmap:0 PCMU/8000
a=ptime:20

<------------->
--- (21 headers 9 lines) ---
Sending to 69.XX.XX.21 : 5060 (no NAT)
Using INVITE request as basis request - 79733A2F-203311DD-80618575-A42EC5B3@69.XX.XX.21
Found no matching peer or user for '69.XX.XX.21:63647'
Found RTP audio format 0
Peer audio RTP is at port 69.XX.XX.21:18410
Found audio description format PCMU for ID 0
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 69.XX.XX.21:18410
Looking for 91250XXXXXXX in local (domain 69.XX.XX.23)

<--- Reliably Transmitting (no NAT) to 69.XX.XX.21:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 69.XX.XX.21:5060;branch=z9hG4bKA1775;received=69.XX.XX.21
From: "unknown" <sip:69.XX.XX.23>;tag=1BF378-CC0
To: <sip:91250XXXXXXX@69.XX.XX.23>;tag=as7d40bb8e
Call-ID: 79733A2F-203311DD-80618575-A42EC5B3@69.XX.XX.21
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
[May 13 14:54:38] NOTICE[3407]: chan_sip.c:13885 handle_request_invite: Call from '' to extension '91250XXXXXXX' rejected because extension not found.
Scheduling destruction of SIP dialog '79733A2F-203311DD-80618575-A42EC5B3@69.XX.XX.21' in 32000 ms (Method: INVITE)
localhost*CLI>
<--- SIP read from 69.XX.XX.21:5060 --->
ACK sip:91250XXXXXXX@69.XX.XX.23:5060 SIP/2.0
Via: SIP/2.0/UDP 69.XX.XX.21:5060;branch=z9hG4bKA1775
From: "unknown" <sip:69.XX.XX.23>;tag=1BF378-CC0
To: <sip:91250XXXXXXX@69.XX.XX.23>;tag=as7d40bb8e
Date: Tue, 13 May 2008 14:55:34 GMT
Call-ID: 79733A2F-203311DD-80618575-A42EC5B3@69.XX.XX.21
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

in comparison to a softphone client that is able to make these calls; this line looks funny

From: "unknown" <sip:69.XX.XX.23>;tag=1BF378-CC0

and this line looks funny

Looking for 91250XXXXXXX in local (domain 69.XX.XX.23)

where on a call that works these lines appear more like:

From: local <sip:6201@69.XX.XX.23>;tag=3783457115

and

Looking for 91250XXXXXXX in numberplan-custom-1 (domain asterisk)

help~!

Hi, did you find a solution for this? I’m experiencing the exact same problem with a Cisco 7940G telephone, my network also includes several Linksys PAP2 boxes and soft phones, none of which have any problems. The Cisco however can only dial local extensions and receive calls.

I’m having the exact same problem. My x-lite can dial out fine but my cisco 7941G only gets reorder. The only difference I can see is in what I think is a reference to the context. I belive default is refering to my extensions.conf. The dial plan does have include = default.

From sip debug:

cisco:
Looking for 97xxxxxxxxxx in default (domain 192.168.x.x)

x-lite:
Looking for 97xxxxxxxxxx in numberplan-custom-1 (domain 192.168.x.x)

Why does the cisco connect with default and the x-list uses the correct dial plan?

What does you sip setting look like for the analog phones. Seems like you don’t have caller id’s set properly.

/Parkley