Re: [asterisk-users] Problems with DTMF on IVRs
Have you set dtmf mode rfc2833 or avt in your phone? On Thu, Aug 28, 2008 at 4:29 PM, Chris Mason (Lists) [EMAIL PROTECTED]wrote: I have a client with 30 extensions, all Polycom 501 phones, an Asterisk 1.2.30.2 installation, and trunking over SIP to TelIAX. Everything works fine except where they need to use DTMF to navigate IVRs such as Dell.com. The tones are not recognized at all. My sip.conf lists for each extension: dtmfmode=rfc2833 and in the [general] section: relaxdtmf=yes I have a very similar system at my office nad DTMF works. Any ideas why this does not work? Chris Mason Comet Systems -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem using blind transfer
Hi All, I'm having strange issue while doing blind transfer of calling channel. My call scenario is as below: 1. Call comes to Asterisk from other switch. 2. AGI script is executed for billing and routing which dials the sip user from withing script. 3. tT options are passed to dial command so both channels can transfer the call. 4. Calling channel can transfer the call by pressing # and entering new destination however when callee tries to transfer the call after pressing # and entering the destination number, both the channels are dropped and transfer doesn't work. Can anyone guide me what is wrong in this scenario? Remember that all operations are done from withing AGI script. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Static IP for SIP?
Having static IP is not necessary. You can register to the gateway using username/password and send calls. If your provider do authentication based on static IP then only you need it. On Mon, Aug 25, 2008 at 4:53 PM, Shariq Khan [EMAIL PROTECTED]wrote: Beginner Question --- Is it necessary to have an static or fixed IP for asterisk for dialing out on SIP. Is there any effect on the call, if i dont have any static IP only for outgoing calling? Shariq ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with dtmf in voicemailmain
Hi All, I'm having strange problem in dtmf detection. As far as I understand, if we set dtmfmode=auto in asterisk for particular user, asterisk will accept all dtmf modes sent from device. I have set dtmfmode=auto in asterisk and inband in device. When I call voicemail extension to check voicemail through voicemailmain, dtmf is not working at all. Can anyone please let me know what is wrong. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
