[Asterisk-Users] Call Manager Express Peer

[Nathan Alberti]

I have the following configuration and am obviously missing something 
small that is causing * not to work as expected.


I have the following defined in sip.conf

[ccme-in]
type=peer
host=10.0.9.1
context=devel_in
disallow=all
allow=alaw
nat=no
canreinvite=yes
qualify=yes

and [devel_in] is defined in extentions.conf

However when I try to call via the dial peer I have configured on the 
cisco (below) I get :

Feb 22 18:37:40 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot 
find extension context 'default'

Which is correct, meaning the context declaration is not being respected.

------
dial-peer voice 101 voip
 destination-pattern 10.
 session protocol sipv2
 session target ipv4:10.0.0.133
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
-------


My bad or something else ??

TIA,

Nathan.



Here is a sip debug for that peer:


Sending to 10.0.9.1 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.0.9.1:19206
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 
0x1 (g723)
Looking for 101 in default
Feb 22 18:44:25 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot 
find extension context 'default'
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.9.1:5060;branch=z9hG4bK3047A
From: "Test Phone 1" <sip:95555001 at 10.0.9.1>;tag=17AFD44-10AD
To: <sip:101 at 10.0.0.133>;tag=as3edc130d
Call-ID: 2C2C3A74-83F511D9-8450EFE0-1F555CD9 at 10.0.9.1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101 at 10.0.0.133>
Content-Length: 0


 to 10.0.9.1:5060
Destroying call '2C2C3A74-83F511D9-8450EFE0-1F555CD9 at 10.0.9.1'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.0.9.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.133:5060;branch=z9hG4bK2b06e290
From: "asterisk" <sip:asterisk at 10.0.0.133>;tag=as0a8b5343
To: <sip:10.0.9.1>
Contact: <sip:asterisk at 10.0.0.133>
Call-ID: 6d840c056f0f06c241e744263a64623b at 10.0.0.133
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 22 Feb 2005 10:44:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 10.0.9.1:5060
Destroying call '6d840c056f0f06c241e744263a64623b at 10.0.0.133'