Integration with Asterisk/Trixbox

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Integration with Asterisk/Trixbox

John Quick
James

The default behaviour for Asterisk is to send re-invites to the connected parties that will re-direct
the RTP stream to go directly between the end-points instead of going through Asterisk.

In theory the option "canreinvite=no" should prevent this happening, but I have never found it works.
Instead, the trick that always works for me is to add an option to the Dial command that tells
Asterisk to look for DTMF during the call. Suitable options include "t", T", "h", "H", "w", "W" or
"L". Check for details here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

...and this means it is really a question related to Asterisk and not OpenSIPS.

John Quick
Systems Consultant
Smartvox Limited

> 2009/5/20 James Lamanna <jlamanna at gmail.com>:
>> Hi,
>> I want to use OpenSIPs as the registrar (and NAT handler) for an
>> Asterisk/Trixbox installation.
>> I've got things partially working, but I've totally made a mess of my
>> config (I can post it if you would like).
>>
>> Some things that I need:
>>
>> I'm having problems with SIP<->SIP calls because I need asterisk to
>> stay in the media stream, so really the call has to be routed like:
>>
>> phone1 <--> opensips <--> asterisk <--> opensips <--> phone2.
>>
>> Does anyone have any configs that come close to this that I could stare at?

> Set "canreinvite=no" for opensips peer in sip.conf.


>> The ones I've found on the web are useful in some ways, but not in others.

> This question is more related to Asterisk.




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Re: Integration with Asterisk/Trixbox

Iñaki Baz Castillo
2009/5/22 John Quick <[hidden email]>:
> James
>
> The default behaviour for Asterisk is to send re-invites to the connected parties that will re-direct
> the RTP stream to go directly between the end-points instead of going through Asterisk.
>
> In theory the option "canreinvite=no" should prevent this happening, but I have never found it works.

Perhaps "canreinivite=never" will force it definitively (not sure).


> Instead, the trick that always works for me is to add an option to the Dial command that tells
> Asterisk to look for DTMF during the call. Suitable options include "t", T", "h", "H", "w", "W" or
> "L".

"t" and "T" options will force RTP through Asterisk if the peers are
configured with dtmfmode=rfc2833. If you set "dtmfmode=info" then the
audio will not force to pass through Asterisk (it  will depend on
other factors as canreinvite and so).



> http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
>
> ...and this means it is really a question related to Asterisk and not OpenSIPS.

For sure :)
Unfortunatelly it seems that people integrating OpenSIPS with Asterisk
always comes to OpenSIPS maillist to ask question, in fact, about
Asterisk :(



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Iñaki Baz Castillo
<[hidden email]>

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Re: Integration with Asterisk/Trixbox

Alex Balashov
Iñaki Baz Castillo wrote:

> For sure :)
> Unfortunatelly it seems that people integrating OpenSIPS with Asterisk
> always comes to OpenSIPS maillist to ask question, in fact, about
> Asterisk :(

There's always the SER-Asterisk-Interwork list:

http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork

--
Alex Balashov
Evariste Systems
Web     : http://www.evaristesys.com/
Tel     : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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