Hope some can help me. I have a PSTN coming into TDM400 into Asterisk. We also have direct telephones connected to the PSTN bypassing the Asterisk. When a call comes in on the PSTN the direct connected phones ring first and if no one picks up , Asterisk picks and get routed to internal sip phones. I am not able to find what I should tune to make the calls always go through asterisk without the direct telephones ringing. Things used to work right, suddenly, I have this problem after a recent storm. Thanks, -braman -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100407/235b6eed/attachment.htm
Hi Try usecallerid=no usually the asterisk server will wait two rings before answering when usecallerid is set to yes. Flavio E. Goncalves www.asteriskguide.com 2010/4/7 Balu Raman <braman09 at gmail.com>> Hope some can help me. > I have a PSTN coming into TDM400 into Asterisk. We also have direct > telephones connected to the PSTN bypassing the Asterisk. When a call comes > in on the PSTN the direct connected phones ring first and if no one picks up > , Asterisk picks and get routed to internal sip phones. I am not able to > find what I should tune to make the calls always go through asterisk without > the direct telephones ringing. Things used to work right, suddenly, I have > this problem after a recent storm. > Thanks, > -braman > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100407/68da4022/attachment.htm
On Wed, 7 Apr 2010, Balu Raman wrote:> I have a PSTN coming into TDM400 into Asterisk. We also have direct > telephones connected to the PSTN bypassing the Asterisk. When a call > comes in on the PSTN the direct connected phones ring first and if no > one picks up , Asterisk picks and get routed to internal sip phones. I > am not able to find what I should tune to make the calls always go > through asterisk without the direct telephones ringing. Things used to > work right, suddenly, I have this problem after a recent storm.0) A more specific subject will get more specific answers. Your description sounds like: You have an incoming PSTN connected to a junction strip. You have telephones and your Asterisk server connected to the junction strip. "Ringing" will be present at the telephones and the Asterisk server at the same time because they are connected together at the same point. There is no way for Asterisk to ring before the telephones unless your telephones don't ring immediately. If you want Asterisk to handle all incoming calls, the telephones will need to be connected to the TDM400 as well as the incoming PSTN. What changes were introduced because of the storm? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000
On Wed, Apr 7, 2010 at 11:07 AM, Balu Raman <braman09 at gmail.com> wrote:> Hope some can help me. > I have a PSTN coming into TDM400 into Asterisk. We also have direct > telephones connected to the PSTN bypassing the Asterisk. When a call comes > in on the PSTN the direct connected phones ring first and if no one picks up > , Asterisk picks and get routed to internal sip phones. I am not able to > find what I should tune to make the calls always go through asterisk without > the direct telephones ringing. Things used to work right, suddenly, I have > this problem after a recent storm. > Thanks, > -braman > >Can you turn the ring volume all the way down on the POTS phones? That is the only way I can think of that would prevent them from "Ringing" Thanks, Steve Totaro -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100407/df7de140/attachment.htm