Re: [Asterisk-Users] weird call transfer problem
Anton Krall wrote: Guys. I just had a weird problem. I have my Dial cmd configured with mwtWT as parameters however, a call came in thru a zap channel and I answered on a sip phone. I tried using # as configured on my features.conf file to transfer the call but the transfer prompt never came in, so I asked the person on the zap channel to do the same and voila, he did get the transfer prompt and entered and extension, but what happended is that I was the one that got transfered! Not him! So. Any ideas whats wrong? The sip phone is an ata, a handytone 286 and zaptel cards. Why cant I do the # transfer and they can but Im the one been transfered? The T option allows the *calling* user to transfer the call, which is what happened to you. The t option allows the call recipient to transfer the caller to another extension. So to stop that from happening, remove the T option from the dial command. As to why you yourself can't transfer it might have something to do with the ATA itself, check what dtmfmode is specified in sip.conf. From the sample sip.conf, it says: dtmfmode=info ; either RFC2833 or INFO for the BudgeTone flynn ___ 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
RE: [Asterisk-Users] weird call transfer problem
I have the dtmf commented out on sip.conf ;dtmfmode=rfc2833 And the ata have it configured as info The weird thing is tht if I am the one making the call, I CAN do transfers, I just cant make them if I am the one receiving the call. I understand that removing T will forbid the calling user to transfer but as far as I know, I should be able to transfer calls myself... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of El Flynn Sent: Miércoles, 13 de Abril de 2005 12:58 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] weird call transfer problem Anton Krall wrote: Guys. I just had a weird problem. I have my Dial cmd configured with mwtWT as parameters however, a call came in thru a zap channel and I answered on a sip phone. I tried using # as configured on my features.conf file to transfer the call but the transfer prompt never came in, so I asked the person on the zap channel to do the same and voila, he did get the transfer prompt and entered and extension, but what happended is that I was the one that got transfered! Not him! So. Any ideas whats wrong? The sip phone is an ata, a handytone 286 and zaptel cards. Why cant I do the # transfer and they can but Im the one been transfered? The T option allows the *calling* user to transfer the call, which is what happened to you. The t option allows the call recipient to transfer the caller to another extension. So to stop that from happening, remove the T option from the dial command. As to why you yourself can't transfer it might have something to do with the ATA itself, check what dtmfmode is specified in sip.conf. From the sample sip.conf, it says: dtmfmode=info ; either RFC2833 or INFO for the BudgeTone flynn ___ 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 ___ 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
Re: [Asterisk-Users] weird call transfer problem
On 4/13/05, Anton Krall [EMAIL PROTECTED] wrote: I have the dtmf commented out on sip.conf ;dtmfmode=rfc2833 And the ata have it configured as info The weird thing is tht if I am the one making the call, I CAN do transfers, I just cant make them if I am the one receiving the call. I understand that removing T will forbid the calling user to transfer but as far as I know, I should be able to transfer calls myself... No you shouldn't unless you have t and you are recieving the call, or you have T and you are making the call. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of El Flynn Sent: Miércoles, 13 de Abril de 2005 12:58 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] weird call transfer problem Anton Krall wrote: Guys. I just had a weird problem. I have my Dial cmd configured with mwtWT as parameters however, a call came in thru a zap channel and I answered on a sip phone. I tried using # as configured on my features.conf file to transfer the call but the transfer prompt never came in, so I asked the person on the zap channel to do the same and voila, he did get the transfer prompt and entered and extension, but what happended is that I was the one that got transfered! Not him! So. Any ideas whats wrong? The sip phone is an ata, a handytone 286 and zaptel cards. Why cant I do the # transfer and they can but Im the one been transfered? The T option allows the *calling* user to transfer the call, which is what happened to you. The t option allows the call recipient to transfer the caller to another extension. So to stop that from happening, remove the T option from the dial command. As to why you yourself can't transfer it might have something to do with the ATA itself, check what dtmfmode is specified in sip.conf. From the sample sip.conf, it says: dtmfmode=info ; either RFC2833 or INFO for the BudgeTone flynn ___ 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 ___ 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 ___ 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
[Asterisk-Users] weird call transfer problem
Guys. I just had a weird problem. I have my Dial cmd configured with mwtWT as parameters however, a call came in thru a zap channel and I answered on a sip phone. I tried using # as configured on my features.conf file to transfer the call but the transfer prompt never came in, so I asked the person on the zap channel to do the same and voila, he did get the transfer prompt and entered and extension, but what happended is that I was the one that got transfered! Not him! So. Any ideas whats wrong? The sip phone is an ata, a handytone 286 and zaptel cards. Why cant I do the # transfer and they can but Im the one been transfered? Hope you can help. ___ 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
