Hello, I am getting problem for delay call hang-up with the below scenario: PSTN User (calling Party)------------->PSTN Line ----------> FXO with Asterisk Box----------->SIP IP Phone (called party) I am using X100P card with my Asterisk-1.0.7 box. I am also using Zaptel-1.0.7 version. When PSTN user makes call to my PSTN line and after getting IVR, PSTN user dial my SIP IP Phone extension, as soon as PSTN user gets one ring back tone, PSTN user cut off the current call. But SIP IP Phone rings till its timeout. I would appreciate if anyone give me solution for the above case. Regards Nahid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050707/177a739b/attachment.htm
i have similar problem, but the sip phone just rings 1 or 2 more times, not until the timeout expires. what is your config in zapata.conf specifically callprogress an busydetect parameters can help best regards On 7/7/05, Nahid Hossain <nahid@stitel.com> wrote:> > > > Hello, > > > > I am getting problem for delay call hang-up with the below scenario: > > > > PSTN User (calling Party)-----------?PSTN Line --------? FXO with Asterisk > Box---------?SIP IP Phone (called party) > > > > > > I am using X100P card with my Asterisk-1.0.7 box. I am also using > Zaptel-1.0.7 version. > > > > When PSTN user makes call to my PSTN line and after getting IVR, PSTN user > dial my SIP IP Phone extension, as soon as PSTN user gets one ring back > tone, PSTN user cut off the current call. But SIP IP Phone rings till its > timeout. > > > > I would appreciate if anyone give me solution for the above case. > > > > Regards > > Nahid > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"
Where are you located? What's not working is the remote party hangup detection, and callprogress only works on selected countries. Please, load your wcfxs (or wctdm) module with debug=1, and check /var/log/messages to see if the card is detecting polarity reversals when you answer the PSTN line and when the other party hangs up. If that's true, you may want to try this patch: http://www.maxosystem.net/asterisk/asterisk-stable-polarity.html Julian. On 7/7/05, Nahid Hossain <nahid@stitel.com> wrote:> Hello, > I am getting problem for delay call hang-up with the below scenario: