Dear List, I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable attended transfer feature. but i just can't do it work. I've already set "atxfer = *" (and many other combinations) and all extensions on extensions.conf have the t and T option. But when I'm going to test, it doesn't work. Is there any other file that i have to configure in order to make it work? I've already looked at google so many times and nothing Does anybody have an idea?? Regards -- Arlen Nascimento
Attended transfer is really four functions 1. Put the caller on Hold while you dial another number 2. Speak to the dialed number (announce the call) 3. Patch the call on hold to the other party using transfer button. 4. Disconnect (otherwise this would be a 3 party conference) How these functions work depend on what type of device the operator is using. SIP phones have this functionality ie a hold button, a transfer button and multi-line appearances. If you are using an ATA with an ordinary phone and standard dial-pad then you may be able to put a call on hold by using the "*" and transfer by "#". But obviously one is limited to the vacant digits on the dial pad (DTMF). Note: If your analog (POTS) phone has a "hold" button this will not work as the hold button simply applies a resistive load to "hold" the loop in an off-hook status. Hope this helps... Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada> Dear List, > > I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable > attended transfer feature. but i just can't do it work. I've already > set "atxfer = *" (and many other combinations) and all extensions on > extensions.conf have the t and T option. But when I'm going to test, > it doesn't work. Is there any other file that i have to configure in > order to make it work? I've already looked at google so many times and > nothing > > Does anybody have an idea?? > > Regards > -- > Arlen Nascimento > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Henry.L.Coleman wrote:> Attended transfer is really four functions > 1. Put the caller on Hold while you dial another number > 2. Speak to the dialed number (announce the call) > 3. Patch the call on hold to the other party using transfer button. > 4. Disconnect (otherwise this would be a 3 party conference) > > How these functions work depend on what type of device the operator is > using. SIP phones have this functionality ie a hold button, a transfer > button and multi-line appearances. If you are using an ATA with an > ordinary > phone and standard dial-pad then you may be able to put a call on hold by > using the "*" and transfer by "#". But obviously one is limited to the > vacant digits on the dial pad (DTMF).With an ATA you would use FLASH (aka RECALL)
Henry, according with voip-info.org, attended transfer is "While on conversation with another party, you dial the atxfer key sequence. Asterisk says "Transfer" then gives you a dial tone, while putting the other party on hold. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hangs up and you will be back to your original conversation." The callee is put on hold "automatically" Eric, attended transfer is only possible with an ATA?? On 12/5/06, Eric ManxPower Wieling <eric@fnords.org> wrote:> Henry.L.Coleman wrote: > > Attended transfer is really four functions > > 1. Put the caller on Hold while you dial another number > > 2. Speak to the dialed number (announce the call) > > 3. Patch the call on hold to the other party using transfer button. > > 4. Disconnect (otherwise this would be a 3 party conference) > > > > How these functions work depend on what type of device the operator is > > using. SIP phones have this functionality ie a hold button, a transfer > > button and multi-line appearances. If you are using an ATA with an > > ordinary > > phone and standard dial-pad then you may be able to put a call on hold by > > using the "*" and transfer by "#". But obviously one is limited to the > > vacant digits on the dial pad (DTMF). > > With an ATA you would use FLASH (aka RECALL) > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Arlen Nascimento
I had problems with featuredigittimeout . It was too short and betwen digit and digit was happened a timeout. modify to featuredigittimeout = 1000 2006/12/5, Arlen Nascimento <arlen.nascimento@gmail.com>:> > Dear List, > > I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable > attended transfer feature. but i just can't do it work. I've already > set "atxfer = *" (and many other combinations) and all extensions on > extensions.conf have the t and T option. But when I'm going to test, > it doesn't work. Is there any other file that i have to configure in > order to make it work? I've already looked at google so many times and > nothing > > Does anybody have an idea?? > > Regards > -- > Arlen Nascimento > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > 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/20061206/24684dc8/attachment.htm