asterisk users - Oct 2010 - AMI Originate

When calling Originate Action, it rings my phone.  If I do not answer, I
receive a Channel Event: Hangup, followed by receiving an
OriginateResponse event with a Failure Response, Reason 3.

 

My phone continues to ring.

If I answer the phone at this point, it attempts to answer, but does not
succeed.

 

Looking at the asterisk debug, it appears to destroy the SIP dialog for
the call.  It also destroys the RTP instance.

 

When I answer, I receive messages...

[Oct  1 15:35:34] DEBUG[3129]: chan_sip.c:6206 find_call: That's odd...
Got a response on a call we dont know about. Callid
2c47c6e4740289d90a0d1337261fd704 at 192.168.9.241

[Oct  1 15:35:34] DEBUG[3129]: chan_sip.c:21256 handle_request_do:
Invalid SIP message - rejected , no callid, len 715

[Oct  1 15:35:35] DEBUG[3129]: chan_sip.c:6206 find_call: That's odd...
Got a response on a call we dont know about. Callid
2c47c6e4740289d90a0d1337261fd704 at 192.168.9.241

[Oct  1 15:35:35] DEBUG[3129]: chan_sip.c:21256 handle_request_do:
Invalid SIP message - rejected , no callid, len 715

[Oct  1 15:35:36] DEBUG[3129]: chan_sip.c:6206 find_call: That's odd...
Got a response on a call we dont know about. Callid
2c47c6e4740289d90a0d1337261fd704 at 192.168.9.241

[Oct  1 15:35:36] DEBUG[3129]: chan_sip.c:21256 handle_request_do:
Invalid SIP message - rejected , no callid, len 715

[Oct  1 15:35:38] DEBUG[3129]: chan_sip.c:6206 find_call: That's odd...
Got a response on a call we dont know about. Callid
2c47c6e4740289d90a0d1337261fd704 at 192.168.9.241

[Oct  1 15:35:38] DEBUG[3129]: chan_sip.c:21256 handle_request_do:
Invalid SIP message - rejected , no callid, len 715

 

If I answer before the timeout, it connects to the dialplan, answers,
plays, and hangs up as I expected.

 

 

Am I sending something wrong?

 

Action: Originate

ActionID: 100

Channel: SIP/1000

Exten: 1

Context: createcall

Priority: 1

Timeout: 3

CallerID: SIP/1000

Variable: OriginateCallId=100

Async: true

 

Is there a configuration setting I am missing?

 

I've tried calling a Linksys SIP phone and I've also tried it with
PhonerLite SIP Client, both are doing the same thing.

 

Have a great day!

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_____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Cropp
Sent: Friday, October 01, 2010 3:50 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] AMI Originate

 

<snip>

Timeout: 3

 

3 seconds to answer the phone?  You never get coffee?

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3 miliseconds...

2010/10/1 Danny Nicholas <danny at debsinc.com>
>
>
>
>  ------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Dan Cropp
> *Sent:* Friday, October 01, 2010 3:50 PM
> *To:* asterisk-users at lists.digium.com
> *Subject:* [asterisk-users] AMI Originate
>
>
>
> <snip>
>
> Timeout: 3
>
>
>
> 3 seconds to answer the phone?  You never get coffee?
>
> --
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-- 
Rodrigo Lang
http://rodrigorecipes.blogspot.com/<http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html>
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