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 [EMAIL PROTECTED]: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:/[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK32e866aa;rport
From: "Admin System 34" <sip:[EMAIL PROTECTED]>;tag=as2d52e2ca
To: <sip:/[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=as2d52e2ca
To: <sip:/[EMAIL PROTECTED]>;tag=33558825
Date: Tue, 07 Mar 2006 14:49:34 GMT
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=as2d52e2ca
To: <sip:/[EMAIL PROTECTED]>;tag=33558825
Date: Tue, 07 Mar 2006 14:49:34 GMT
Call-ID: [EMAIL PROTECTED]
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:/[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK32e866aa;rport
From: "Admin System 34" <sip:[EMAIL PROTECTED]>;tag=as2d52e2ca
To: <sip:/[EMAIL PROTECTED]>;tag=33558825
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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 '[EMAIL PROTECTED]'
== 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 [EMAIL PROTECTED]: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:/[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK00e00a23;rport
From: "Admin System 34" <sip:[EMAIL PROTECTED]>;tag=as4fdb4cfa
To: <sip:/[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=as4fdb4cfa
To: <sip:/[EMAIL PROTECTED]>;tag=33558827
Date: Tue, 07 Mar 2006 14:49:46 GMT
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=as4fdb4cfa
To: <sip:/[EMAIL PROTECTED]>;tag=33558827
Date: Tue, 07 Mar 2006 14:49:46 GMT
Call-ID: [EMAIL PROTECTED]
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:/[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK00e00a23;rport
From: "Admin System 34" <sip:[EMAIL PROTECTED]>;tag=as4fdb4cfa
To: <sip:/[EMAIL PROTECTED]>;tag=33558827
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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 '[EMAIL PROTECTED]'
== 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>
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