Re: [asterisk-users] DIDs required of Paris and Gottenburg Sweden
Hi Kashif, I use didx.net you can get did numbers in many countries On Jul 3, 2008, at 10:25 PM, Kashif Naeem wrote: Hello All, We need the DIDs of Paris and Gottenburg, Sweden. Can anyone provide ? Please reply with rates. Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- 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
Re: [asterisk-users] One way sound when Using Dial cmd without t option (SOLVED) Need explanation
Thanks Tony for you reply. Do you have any idea why Asterisk require t in Dial command? Cheers, Moe On Sun, May 18, 2008 at 1:14 AM, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Mohammad A. Navid [EMAIL PROTECTED] wrote: I'm implementing a simple calling card feature for testing purpose. I have a DID number, when I called my DID number and enter the phone number to call, Asterisk would dial the number for me but the sound was only one way. After hours of struggling with the problem, I found out that I need to add t to my dial options, this is the correct way of dialing out: - Dial(SIP/carrier/310555|20|t) Now I need to know what was going on? Why with option t both parties can hear each other, but without option t in dial cmd only one party could hear? Another interesting issue is, if I use Answer() command at the begining the sound becomes one way even if I use t in options. Try adding reinvite=no to the sip.conf or users.conf definition for your SIP service provider. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
