Re: [asterisk-users] Audiocodes Mediant 1000, Polycom, and no ringback on transfer
I finally have the solution, so thought I would post back to the list for completeness. It ended up being a series of changes. First, on the gateway, set Disconnect on Broken Connection to false. Then, for the Polycom phones, set voIpProt.SIP.allowTransferOnProceeding to 1 in the sip.cfg. Next, set progressinband=yes in sip.conf. Finally, in my dialplan, I had to remove calls to Answer() before calling dial. With all of this, the gateway is working brilliantly! Thanks, James ___ --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
Re: [asterisk-users] Audiocodes Mediant 1000, Polycom, and no ringback on transfer
On Mon, 2007-01-15 at 15:26 -0600, David Gomillion wrote: I don't think you can do that. Here's why: on the Polycom's, the Transfer button doesn't reappear until the transferree picks up the phone. Unless something changed in the firmware recently. But, if you're completing it before the 3rd party answers, it's not an attended transfer. I found it all depends on the dialplan, and the sip.cfg for the phone. If you call Answer() before Dial(), it will allow it. There is also a setting in sip.cfg for the phones, voIpProt.SIP.allowTransferOnProceeding that I think allows that as well. I should have mentioned in my original post, but MOH works just fine. When I complete the transfer, the MOH stops, and that's when the dead air starts. Anyone else have any suggestions? Thanks, James ___ --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
[asterisk-users] Audiocodes Mediant 1000, Polycom, and no ringback on transfer
I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I am using Polycom 501's and 601',s and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, the PSTN party hears dead air until the call is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any one have any experience with this type of setup? Thanks, James ___ --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
Re: [asterisk-users] Audiocodes Mediant 1000, Polycom, and no ringback on transfer
On 1/15/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I don't have any experience with an Audiocodes Meidant 1000, but I'll try to help you I am using Polycom 501's and 601',s We have a lot of these and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, I don't think you can do that. Here's why: on the Polycom's, the Transfer button doesn't reappear until the transferree picks up the phone. Unless something changed in the firmware recently. But, if you're completing it before the 3rd party answers, it's not an attended transfer. the PSTN party hears dead air until the call is answered or goes to voicemail. I would start by making sure the Music on Hold actually works, and that the SIP phones are properly configured to use a MOH context that actually exists. If those things check out, I would try using a blind transfer and see what happens, try transferring when the 3rd party answers (VM or whatever), and watch the console carefully with as much verbosity as possible. I'm not really sure where to start my troubleshooting. Any one have any experience with this type of setup? Hope this helps, David Thanks, James ___ --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 ___ --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
