Re: [asterisk-users] One way sound when Using Dial cmd without t option (SOLVED) Need explanation
The t, much like reinvite = no keeps asterisk listening to the audio stream to detect dtmf input if dtmf mode is in-band, what is happening is that the sip reinvite is failing, due to a firewall rule or a routing problem and you end up with only one connected RTP stream. Asterisk does not require the t option. Anthony Moe Navid wrote: 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] mailto:[EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED] mailto:[EMAIL PROTECTED], Mohammad A. Navid [EMAIL PROTECTED] mailto:[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] mailto:[EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] mailto:[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 ___ -- 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
Re: [asterisk-users] One way sound when Using Dial cmd without t option (SOLVED) Need explanation
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
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
Re: [asterisk-users] One way sound when Using Dial cmd without t option (SOLVED) Need explanation
In article [EMAIL PROTECTED], Moe Navid [EMAIL PROTECTED] wrote: Thanks Tony for you reply. Did my suggestion fix the problem? Ah yes, I just noticed you said it in the subject line. Do you have any idea why Asterisk require t in Dial command? Yes, t specifies that the called party may transfer the call by pressing # (or some other sequence defined in features.conf). Likewise T says the same about the calling party. In either case, Asterisk needs to remain in the media path so that it can listen for the DTMF. If neither option is specified, Asterisk may try to optimise itself out of the media path by getting the two SIP endpoints to talk to each other directly. I think this is what is happening when you get one-way audio, since the two endpoints may not know how to reach each other directly (particularly if NAT is involved). Putting reinvite=no in sip.conf for either endpoint (but best to do it for the Service Provider endpoint) tells Asterisk never to optimise itself out of the media path, even if t and T are not specified. Cheers Tony 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 -=-=-=-=-=- [Alternative: text/html] -=-=-=-=-=- -=-=-=-=-=- ___ -- 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 -=-=-=-=-=- -- 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
