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

[Michael Collins]

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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