[Asterisk-Users] SIP Qualify

[Douglas Garstang]

Can someone tell me why qualify=yes is required in sip.conf, before you can detect the dialled status of a channel? If I don't have qualify=yes against a peer, than checking ${DIALSTATUS} has no effect. Asterisk will just keep trying until the dial timeout expires.

Why can't asterisk actually LOOK at the SIP response returned by the SIP proxy and do something with it? Why can't it detect a failure to connect (ie proxy down) and do something based on that? Do the Asterisk developers realise how damn hard this makes it to build any kind of reliability into Asterisk????

Doug

-----Original Message-----
From: Warren Burstein [mailto:warren at softov.co.il]
Sent: Sunday, December 11, 2005 11:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] outgoing calls that last an unreasonably
longtime


Simone Cittadini wrote:

> Warren Burstein ha scritto:
>
>>
>>
>> What is frustrating is that the cdr file shows the dst as T rather 
>> than as the phone number dialed.  I realize that AbsoluteTimout 
>> causes it to jump to the T extension, but it would help to know who 
>> the user dialed (asking a week later isn't going to get any useful 
>> information out of the user).  It's not in the log file, either - 
>> would increasing the log level help here?
>>
>>
> I don't know how this AbsolutTimeout works, anyway I put all the info 
> I need in variables before the actual Dial, then in the h extension I 
> call SetUserField() (or whatever is called), helps me keeping track of 
> reasons for non-terminated calls ...

I am not using the userfield for anything so that sounds like a good 
idea.  It's SetCDRUserField by the way.
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