[Asterisk-Users] Call Manager Express Peer

[Greg Oliver]

The only thing I have different in my CME dial-peers is "application 
session" for each of them.  Other than that, what you have works for me..

-Greg

Nathan Alberti wrote:
> 
> 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'
> 
> 
> 
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