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Jerry Geis

2006-Mar-06 14:00 UTC

[Asterisk-Users] call manager integration

I am getting this error from call manager (4.0) and asterisk 1.2.4

I have canreinvite=yes on the call manager setup.

I can call into the asterisk box from call manager. THat seems to work.
When I am calling out of the box using a call file I see 
this entry from call manager...

What might be the problem with my setup?

THanks,

JErry



----------------

<Date>03/06/2006 13:58:36.374</Date> 
  <Cluster>CO-CCMPUB-01-Cluster</Cluster> 
  <CMHost>10.101.66.10</CMHost> 
  <TraceType>Trace</TraceType> 
  <CTag>2,100,114,1.347</CTag> 
  <SrcDev>10.66.101.10</SrcDev> 
  <SrcIp>INVITE</SrcIp> 
  <CTMapKey /> 
  <CTMapVal /> 
  <info>Cisco CallManagerDigit analysis: wait_DaReq - cepn=[]
BlockFlag=[1]</info>
  </trace>
- <trace>
  <Date>03/06/2006 13:58:36.374</Date> 
  <Cluster>CO-CCMPUB-01-Cluster</Cluster> 
  <CMHost>10.101.66.10</CMHost> 
  <TraceType>Trace</TraceType> 
  <CTag>2,100,114,1.347</CTag> 
  <SrcDev>10.66.101.10</SrcDev> 
  <SrcIp>INVITE</SrcIp> 
  <CTMapKey /> 
  <CTMapVal />

Greg Oliver

2006-Mar-06 14:08 UTC

head link

[Asterisk-Users] call manager integration

On Mon, 2006-03-06 at 15:00, Jerry Geis wrote:> I am getting this error from call manager (4.0) and asterisk 1.2.4
> 
> I have canreinvite=yes on the call manager setup.
> 
> I can call into the asterisk box from call manager. THat seems to work.
> When I am calling out of the box using a call file I see 
> this entry from call manager...
> 
> What might be the problem with my setup?
> 
What is the output on the console with sip debug turned on?

-Greg

Jerry Geis

2006-Mar-06 14:42 UTC

head link

[Asterisk-Users] call manager integration

>On Mon, 2006-03-06 at 15:00, Jerry Geis wrote:
>/ I am getting this error from call manager (4.0) and asterisk 1.2.4/>/ 
/>/ I have canreinvite=yes on the call manager setup.
/>/ 
/>/ I can call into the asterisk box from call manager. THat seems to work.
/>/ When I am calling out of the box using a call file I see 
/>/ this entry from call manager...
/>/ 
/>/ What might be the problem with my setup?
/>/ 
/>What is the output on the console with sip debug turned on?
>-Greg

greg

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.

co-drpage-01*CLI>     -- Attempting call on SIP/CallManager//3726 for 
smvoice_callprogress@smvoice-dialout:1 
<mailto:smvoice_callprogress@smvoice-dialout:1> (Retry 1)
       > Channel SIP/CallManager-03a0 was never answered.
co-drpage-01*CLI> Mar  6 13:57:49 NOTICE[4283]: pbx_spool.c:270 
attempt_thread: Call failed to go through, reason 8
co-drpage-01*CLI> Mar  6 13:58:24 WARNING[4298]: 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
co-drpage-01*CLI>     -- Attempting call on SIP/CallManager//3726 for 
smvoice_callprogress@smvoice-dialout:1 
<mailto:smvoice_callprogress@smvoice-dialout:1> (Retry 1)
co-drpage-01*CLI>   == Spawn extension (smvoice-dialout, failed, 1) 
exited non-zero on 'OutgoingSpoolFailed'
Mar  6 13:58:25 NOTICE[4298]: pbx_spool.c:270 attempt_thread: Call 
failed to go through, reason 8
co-drpage-01*CLI>        > Channel SIP/CallManager-9209 was never
answered.
co-drpage-01*CLI> co-drpage-01*CLI>
co-drpage-01*CLI> Mar  6 13:58:00 WARNING[4290]: cdr.c:548 
ast_cdr_disposition: Cause not handled
co-drpage-01*CLI> co-drpage-01*CLI>     -- Executing 
AGI("OutgoingSpoolFailed", "smvoice|-digium_failed") in new
stack
co-drpage-01*CLI> co-drpage-01*CLI>     -- Launched AGI Script 
/var/lib/asterisk/agi-bin/smvoice
co-drpage-01*CLI> co-drpage-01*CLI>   == Spawn extension 
(smvoice-dialout, failed, 1) exited non-zero on 'OutgoingSpoolFailed'
co-drpage-01*CLI> co-drpage-01*CLI> Mar  6 13:58:01 NOTICE[4290]: 
pbx_spool.c:270 attempt_thread: Call failed to go through, reason 8
co-drpage-01*CLI> co-drpage-01*CLI> co-drpage-01*CLI>

co-drpage-01*CLI>     -- Attempting call on SIP/CallManager//3726 for 
smvoice_callprogress@smvoice-dialout:1 
<mailto:smvoice_callprogress@smvoice-dialout:1> (Retry 1)
co-drpage-01*CLI>        > Channel SIP/CallManager-11f2 was never
answered.
co-drpage-01*CLI> Mar  6 13:58:12 WARNING[4294]: cdr.c:548 
ast_cdr_disposition: Cause not handled
co-drpage-01*CLI>     -- Executing AGI("OutgoingSpoolFailed", 
"smvoice|-digium_failed") in new stack
co-drpage-01*CLI>     -- Launched AGI Script 
/var/lib/asterisk/agi-bin/smvoice
co-drpage-01*CLI>   == Spawn extension (smvoice-dialout, failed, 1) 
exited non-zero on 'OutgoingSpoolFailed'
co-drpage-01*CLI> Mar  6 13:58:13 NOTICE[4294]: pbx_spool.c:270 
attempt_thread: Call failed to go through, reason 8
co-drpage-01*CLI>     -- Attempting call on SIP/CallManager//3726 for 
smvoice_callprogress@smvoice-dialout:1 
<mailto:smvoice_callprogress@smvoice-dialout:1> (Retry 1)
co-drpage-01*CLI>        > Channel SIP/CallManager-03a0 was never
answered.
co-drpage-01*CLI> Mar  6 13:58:24 WARNING[4298]: cdr.c:548 
ast_cdr_disposition: Cause not handled
co-drpage-01*CLI>     -- Executing AGI("OutgoingSpoolFailed", 
"smvoice|-digium_failed") in new stack
co-drpage-01*CLI>     -- Launched AGI Script 
/var/lib/asterisk/agi-bin/smvoice
co-drpage-01*CLI>   == Spawn extension (smvoice-dialout, failed, 1) 
exited non-zero on 'OutgoingSpoolFailed'
co-drpage-01*CLI> Mar  6 13:58:25 NOTICE[4298]: pbx_spool.c:270 
attempt_thread: Call failed to go through, reason 8
co-drpage-01*CLI> co-drpage-01*CLI>

    -- Attempting call on SIP/CallManager//3726 for 
smvoice_callprogress@smvoice-dialout:1 
<mailto:smvoice_callprogress@smvoice-dialout:1> (Retry 1)
       > Channel SIP/CallManager-7dcc was never answered.
Mar  6 13:58:36 WARNING[4302]: 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
  == Spawn extension (smvoice-dialout, failed, 1) exited non-zero on 
'OutgoingSpoolFailed'
Mar  6 13:58:37 NOTICE[4302]: pbx_spool.c:270 attempt_thread: Call 
failed to go through, reason 8
co-drpage-01*CLI>

sip.conf
                        [CallManager]
                        type=friend
                        host=10.101.66.10
                        context=from_call_manager
                        disallow=all
                        allow=alaw
                        allow=ulaw
                        allow=gsm
                        dtmfmode=rfc2833
                        canreinvite=yes

Jerry Geis

2006-Mar-07 08:06 UTC

head link

[Asterisk-Users] call manager integration

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@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@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@10.101.69.200>;tag=as2d52e2ca
To: <sip:/3726@10.101.66.10>
Contact: <sip:0@10.101.69.200>
Call-ID: 071ae801226b08ba613364ef6efcc9ff@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@10.101.69.200>;tag=as2d52e2ca
To: <sip:/3726@10.101.66.10>;tag=33558825
Date: Tue, 07 Mar 2006 14:49:34 GMT
Call-ID: 071ae801226b08ba613364ef6efcc9ff@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@10.101.69.200>;tag=as2d52e2ca
To: <sip:/3726@10.101.66.10>;tag=33558825
Date: Tue, 07 Mar 2006 14:49:34 GMT
Call-ID: 071ae801226b08ba613364ef6efcc9ff@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@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@10.101.69.200>;tag=as2d52e2ca
To: <sip:/3726@10.101.66.10>;tag=33558825
Contact: <sip:0@10.101.69.200>
Call-ID: 071ae801226b08ba613364ef6efcc9ff@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@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@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@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@10.101.69.200>;tag=as4fdb4cfa
To: <sip:/3726@10.101.66.10>
Contact: <sip:0@10.101.69.200>
Call-ID: 7b18ce5c56f2063524aca3a667e07d7c@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@10.101.69.200>;tag=as4fdb4cfa
To: <sip:/3726@10.101.66.10>;tag=33558827
Date: Tue, 07 Mar 2006 14:49:46 GMT
Call-ID: 7b18ce5c56f2063524aca3a667e07d7c@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@10.101.69.200>;tag=as4fdb4cfa
To: <sip:/3726@10.101.66.10>;tag=33558827
Date: Tue, 07 Mar 2006 14:49:46 GMT
Call-ID: 7b18ce5c56f2063524aca3a667e07d7c@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@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@10.101.69.200>;tag=as4fdb4cfa
To: <sip:/3726@10.101.66.10>;tag=33558827
Contact: <sip:0@10.101.69.200>
Call-ID: 7b18ce5c56f2063524aca3a667e07d7c@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@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>

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