[Asterisk-Users] call manager integration

Tue Mar 7 08:06:52 MST 2006

On Mon, 2006-03-06 at 15:42, Jerry Geis wrote:

>>/ here is some of the output. I am no longer the to spcifically do sip 
/>>/ debug but this is what I have.
/>>/ along with my sip.conf snip.
/>>/ 
/>>/ The call to extension 3726 never rings. so it never gets answered.
/>>/ 
/
>Are you sure your sip trunk and route pattern are in the same
>partition/CSS by chance?>

>Without more info (AGI script and SIP debug), I really can't be much
>more help.  Your sip.conf entry is good though.

>Your callmanager context from extensions.conf will help as well.

>-Greg

Greg,

here is the sip debug output... Again I can call into the asterisk box but cant call out
with call files. You mentioned my sip.conf entry looked OK and I have canreinvite=yes in that file
for the CallManager.

Thanks, 

Jerry

----------------
sip debug
SIP Debugging re-enabled
    -- Attempting call on SIP/CallManager//3726 for smvoice_callprogress at smvoice-dialout:1 (Retry 1)
We're at 10.101.69.200 port 12592
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 10.101.66.10:5060:
INVITE sip:/3726 at 10.101.66.10 SIP/2.0
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK32e866aa;rport
From: "Admin System 34" <sip:0 at 10.101.69.200>;tag=as2d52e2ca
To: <sip:/3726 at 10.101.66.10>
Contact: <sip:0 at 10.101.69.200>
Call-ID: 071ae801226b08ba613364ef6efcc9ff at 10.101.69.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 07 Mar 2006 14:49:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 4082 4082 IN IP4 10.101.69.200
s=session
c=IN IP4 10.101.69.200
t=0 0
m=audio 12592 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
co-drpage-01*CLI>
<-- SIP read from 10.101.66.10:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK32e866aa;rport
From: "Admin System 34" <sip:0 at 10.101.69.200>;tag=as2d52e2ca
To: <sip:/3726 at 10.101.66.10>;tag=33558825
Date: Tue, 07 Mar 2006 14:49:34 GMT
Call-ID: 071ae801226b08ba613364ef6efcc9ff at 10.101.69.200
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

--- (9 headers 0 lines)---
co-drpage-01*CLI>
<-- SIP read from 10.101.66.10:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK32e866aa;rport
From: "Admin System 34" <sip:0 at 10.101.69.200>;tag=as2d52e2ca
To: <sip:/3726 at 10.101.66.10>;tag=33558825
Date: Tue, 07 Mar 2006 14:49:34 GMT
Call-ID: 071ae801226b08ba613364ef6efcc9ff at 10.101.69.200
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

--- (9 headers 0 lines)---
Transmitting (no NAT) to 10.101.66.10:5060:
ACK sip:/3726 at 10.101.66.10 SIP/2.0
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK32e866aa;rport
From: "Admin System 34" <sip:0 at 10.101.69.200>;tag=as2d52e2ca
To: <sip:/3726 at 10.101.66.10>;tag=33558825
Contact: <sip:0 at 10.101.69.200>
Call-ID: 071ae801226b08ba613364ef6efcc9ff at 10.101.69.200
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
       > Channel SIP/CallManager-a48d was never answered.
Mar  7 08:49:32 WARNING[5219]: cdr.c:548 ast_cdr_disposition: Cause not handled
    -- Executing AGI("OutgoingSpoolFailed", "smvoice|-digium_failed") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
Destroying call '071ae801226b08ba613364ef6efcc9ff at 10.101.69.200'
  == Spawn extension (smvoice-dialout, failed, 1) exited non-zero on 'OutgoingSpoolFailed'
Mar  7 08:49:34 NOTICE[5219]: pbx_spool.c:270 attempt_thread: Call failed to go through, reason 8
    -- Attempting call on SIP/CallManager//3726 for smvoice_callprogress at smvoice-dialout:1 (Retry 1)
We're at 10.101.69.200 port 19812
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 10.101.66.10:5060:
INVITE sip:/3726 at 10.101.66.10 SIP/2.0
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK00e00a23;rport
From: "Admin System 34" <sip:0 at 10.101.69.200>;tag=as4fdb4cfa
To: <sip:/3726 at 10.101.66.10>
Contact: <sip:0 at 10.101.69.200>
Call-ID: 7b18ce5c56f2063524aca3a667e07d7c at 10.101.69.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 07 Mar 2006 14:49:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 4082 4082 IN IP4 10.101.69.200
s=session
c=IN IP4 10.101.69.200
t=0 0
m=audio 19812 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
co-drpage-01*CLI>
<-- SIP read from 10.101.66.10:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK00e00a23;rport
From: "Admin System 34" <sip:0 at 10.101.69.200>;tag=as4fdb4cfa
To: <sip:/3726 at 10.101.66.10>;tag=33558827
Date: Tue, 07 Mar 2006 14:49:46 GMT
Call-ID: 7b18ce5c56f2063524aca3a667e07d7c at 10.101.69.200
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

--- (9 headers 0 lines)---
co-drpage-01*CLI>
<-- SIP read from 10.101.66.10:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK00e00a23;rport
From: "Admin System 34" <sip:0 at 10.101.69.200>;tag=as4fdb4cfa
To: <sip:/3726 at 10.101.66.10>;tag=33558827
Date: Tue, 07 Mar 2006 14:49:46 GMT
Call-ID: 7b18ce5c56f2063524aca3a667e07d7c at 10.101.69.200
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

--- (9 headers 0 lines)---
Transmitting (no NAT) to 10.101.66.10:5060:
ACK sip:/3726 at 10.101.66.10 SIP/2.0
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK00e00a23;rport
From: "Admin System 34" <sip:0 at 10.101.69.200>;tag=as4fdb4cfa
To: <sip:/3726 at 10.101.66.10>;tag=33558827
Contact: <sip:0 at 10.101.69.200>
Call-ID: 7b18ce5c56f2063524aca3a667e07d7c at 10.101.69.200
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
       > Channel SIP/CallManager-48d6 was never answered.
Mar  7 08:49:45 WARNING[5226]: cdr.c:548 ast_cdr_disposition: Cause not handled
    -- Executing AGI("OutgoingSpoolFailed", "smvoice|-digium_failed") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
Destroying call '7b18ce5c56f2063524aca3a667e07d7c at 10.101.69.200'
  == Spawn extension (smvoice-dialout, failed, 1) exited non-zero on 'OutgoingSpoolFailed'
Mar  7 08:49:46 NOTICE[5226]: pbx_spool.c:270 attempt_thread: Call failed to go through, reason 8
co-drpage-01*CLI>