[Freeswitch-users] SIT tones and SIP Trunk provider.

[Vinuth Madinur]

Thanks Michael. I'll go through the resources you mentioned.
Thanks,
Vinuth.


On Tue, Oct 13, 2009 at 2:15 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur <vinuth.madinur at gmail.com
> > wrote:
>
>> Hi,
>> Does Freeswitch detect all of these hangup cases mentioned here [
>> http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a
>> SIP Trunk provider?
>>
>> If not, should I put in tone_detect application in the dialplan for
>> detecting the SITs?
>>
>> Won't freeswitch have to depend on the SIP status sent from SIP trunk to
>> know the hangup status? So, I'm wondering if tone_detect will work at all?
>>
>>
> Vinuth,
>
> As usual, "it depends." Your provider is the key to this whole operation.
> If the SIP provider sends the information inband then you will definitely
> need to use tone_detect to look for the SIT tones. However, if the
> information comes back with the normal SIP messages then you're good to go.
> I've seen more than a few SIP providers do both, which means that you have
> to prepare for both cases.
>
> My advice to you is to get pcaps of failed calls and analyze them with
> Wireshark. If you need help analyzing them then put your pcaps on a web
> server and post a link so that others can download them. The wiki has some
> information on grabbing pcaps:
>
> http://wiki.freeswitch.org/wiki/Packet_Capture
>
> If you haven't already done so, go to cluecon.com and download the torrent
> file that has the ClueCon speaker presentations. The last presentation on
> Day 3 is Jason Garland and he walks you through using Wireshark for
> analyzing a SIP call, including both the signaling (SIP) and the media (RTP)
> parts of the call. BTW, if you have a copy of "VoIP Deployment For Dummies"
> it has a small section on using Wireshark for call analysis.
>
> -MC
>
>
>
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