-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I've been unable to get app_transfer to work, could someone explain how it is supposed to work? Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1 dials ast2 using iax2 and gets instructed to transfer the call to a different extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing happens and after a few seconds, the line is hung up. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAHm+I2TEAILET3McRAkAPAJ4zamcrdGyJuqeOisEVeEpb/9BsFwCfS177 vAn/qlLoRfZWKmBuwz/+pKw=wH7d -----END PGP SIGNATURE-----
Tais M. Hansen wrote:> -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi, > > As I've been unable to get app_transfer to work, could someone > explain how it > is supposed to work? > > Currently I have two Asterisk boxes. A call comes in via zaptel to > ast1. ast1 > dials ast2 using iax2 and gets instructed to transfer the call to a > different extension. iax2 debug shows that a transfer cmd is sent to > ast1, but nothing > happens and after a few seconds, the line is hung up. > > - -- > Regards, > Tais M. Hansen > ComX Networks > Tel: +45-70257474 > Fax: +45-70257374 > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.2.3 (GNU/Linux) > > iD8DBQFAHm+I2TEAILET3McRAkAPAJ4zamcrdGyJuqeOisEVeEpb/9BsFwCfS177 > vAn/qlLoRfZWKmBuwz/+pKw> =wH7d > -----END PGP SIGNATURE----- > > _______________________________________________ > 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-usersHave put "t" in your Dial statement? i.e. Exten => someextension,1,Dial(IAX2/SOMETHING,20,t) Ta SJ
Is it possible to do transfers between extensions while using analog adapters with regular analog phones which do not have transfer buttons? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Own ISP! http://www.YourOwnISP.com
Hi, I'm afraid I don't know how to use the command Transfer. I have a couple of SIP users in the system and although exten => 35,1,Dial(SIP/33) works fine, exten => 35,1,Transfer(33) just don't work. All the description in the wiki is 'Transfer(exten)' without a single example. 35,1,Dial(SIP/33) would be a way to transfer the incoming call from 35 to 33, but what I want to do is to get 33 dialplan, not to dial 33. I mean, if 33 is 33,1,Voicemail that's what I would like to execute when calling 35. Could anybody help me? Thank you, Victor. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050621/7471b2fa/attachment.htm
Can * transfer call if I use canreinvite=yes in sip.conf? Can * start "automon" (recording) if I use canreinvite? If answers are no, then which one did you chouse for your configuration? Do you use "canreinvite=yes" so you can't do those stuff or you don't use this so you have high processor load? -- Tomislav Parcina ime.prezime@email.t-com.hr
Tomislav Parcina a ?crit :>Can * transfer call if I use canreinvite=yes in sip.conf? > >My understanding is that canreinvite only redirects the media path. Signaling and media are separate with SIP (which is what makes it so nice by the way).>Can * start "automon" (recording) if I use canreinvite? > >Dunno. If it works at all, asterisk will have to redirect the media stream to itself. Have you tried?