[Asterisk-Users] Asterisk Call Recording and SIP canreinvite

Waldo Rubinstein waldo at trianet.net
Thu Dec 8 09:24:07 MST 2005


This and Time Bandit's comment makes sense. I didn't realize that  
these options in the Dial string will "force" Asterisk to stay in the  
media path even if canreinvite=yes.

I'll give it a try.

Thanks,
Waldo

On Dec 8, 2005, at 11:18 AM, Steve Totaro wrote:

> There may be a better way but off the top of my head this idea jumped
> out.  It assumes that you know prior to making the call that you  
> need to
> record it and that you have phones capable of multiple lines.
>
> Setup a second line with a different entry in sip.conf with
> canreinvite=no and use that line to make your calls.
>
> Other than that I see reference on the wiki to an H option in dial but
> have never used it.  I think you will still need to know prior to
> dialing whether you will want to record the call or not so you can  
> dial
> the exten that uses the H option.
>
> If you get this to work, please post your results back to this thread.
>
> "Re: Re: H option
> by flobi on Monday 25 of July, 2005 [10:43:46]
> why not just set canreinvite=yes and on the calls where you don't want
> reinvite use the H option (if it actually does disable reinvite) or  
> the
> T or t which also disable reinvite.
>
> 7960G Seems to need canreinvite=no as well.
> by Anonymous on Friday 29 of October, 2004 [22:22:43]
> Running P0S3-07-2-00.
>
> Re: H option
> by Anonymous on Monday 26 of July, 2004 [10:10:07]
> (:confused:) Hmm... Now I started to wonder, if it's somehow  
> possible to
> override the canreinvite=no setting on per call basis. Anyone?
>
> H option
> by Anonymous on Saturday 10 of July, 2004 [04:15:13]
> Asterisk will not reinvite if the H option is used in the Dial  
> command."
>
> http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
>
> Thanks,
> Steve
>
>>
>> I understand. But because the majority of calls are not to be
>> recorded, I don't have a need to keep Asterisk in the media path all
>> the time. That's why I'm wondering if you could dynamically keep it
>> in the media path or not.
>>
>> - Waldo
>>
>> On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote:
>>
>>> Well, then set canreinvite=no
>>>
>>>>
>>>> If that's the case, is it possible to override the canreinvite
>>>> attribute of a SIP peer in extensions.conf before a call is made or
>>>> answered by that peer?
>>>>
>>>> - Waldo
>>>>
>>>> On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote:
>>>>
>>>>>>
>>>>>> Is there a way to optionally keep asterisk in the media path in
>>> order
>>>>>> to record calls using the Monitor command? For example, if I have
> a
>>>>>> SIP peer that is defined with canreinvite=yes, this means that if
>>>>>> possible, Asterisk will not be in the media path. However, what
>>>>>> happens if the user presses something like *1 (defined in
>>>>>> features.conf) to record the call? Will the call be forced to go
>>>>>> through Asterisk automatically?
>>>>>>
>>>>>> Thanks,
>>>>>> Waldo
>>>>>
>>>>>
>>>>> I could be wrong but I am pretty sure that once the asterisk is
> out
>>> of
>>>>> the media path then features like *1 will not work since asterisk
>>>>> is not
>>>>> able to listen for it.
>>>>>
>>>>> Thanks,
>>>>> Steve
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