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


 
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vinuth.madinur at gmai...
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PostPosted: Mon Oct 12, 2009 3:08 pm    Post subject: [Freeswitch-users] SIT tones and SIP Trunk provider. Reply with quote
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?


Please provide your advice.


Thanks,
Vinuth.
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msc at freeswitch.org
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PostPosted: Mon Oct 12, 2009 3:53 pm    Post subject: [Freeswitch-users] SIT tones and SIP Trunk provider. Reply with quote
On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur <vinuth.madinur@gmail.com (vinuth.madinur@gmail.com)> wrote:
Quote:
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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vinuth.madinur at gmai...
Guest
PostPosted: Mon Oct 12, 2009 4:29 pm    Post subject: [Freeswitch-users] SIT tones and SIP Trunk provider. Reply with quote
Thanks Michael. I'll go through the resources you mentioned.

Thanks,
Vinuth.


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


On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur <vinuth.madinur@gmail.com (vinuth.madinur@gmail.com)> wrote:
Quote:
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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