[Freeswitch-users] SIT tones and SIP Trunk provider.
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. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIT tones and SIP Trunk provider.
On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur vinuth.madi...@gmail.comwrote: 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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIT tones and SIP Trunk provider.
Thanks Michael. I'll go through the resources you mentioned. Thanks, Vinuth. On Tue, Oct 13, 2009 at 2:15 AM, Michael Collins m...@freeswitch.org wrote: On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur vinuth.madi...@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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
