Sorry, haven't tried with an HA setup. -----Original Message----- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kyle Haefner Sent: Tuesday, August 23, 2011 9:36 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Auto Attendant transfer to Conference through Audiocodes
Hi Todd, Thanks for the suggestion, I actually use the internal DNS and SRV tables on the Audiocodes. Todd have you ever tried this with an HA setup. I was just looking back through and it seems that it is the slave server that resends the INVITE back to the gateway. Kyle On Tue, Aug 23, 2011 at 10:15 AM, Todd Hodgen <thod...@frontier.com> wrote: > Kyle, Just to confirm your setup is workable with sipxecs, I've set > that scenario up with an ITSP and sipxbridge, and it did work correctly. > > By chance is your Audiocodes setup with the IP address of sipXecs > instead of the FQDN? I recently had a similar issue with an Eqygi > gateway that was fixed by changing IP address to FQDN. > > -----Original Message----- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kyle > Haefner > Sent: Tuesday, August 23, 2011 9:04 AM > To: sipx-users > Subject: [sipx-users] Auto Attendant transfer to Conference through > Audiocodes > > Hi Users, > > I have openUC 4.4 connected through an Audiocodes Mediant 2000 > (firmware 5.8). I'm trying to set up an AutoAttendant to transfer > calls into personal conferences. What I would like is an AA that asks > the the caller to enter the conference bridge number (prefix+5 digit > user extension) and then be transferred into the conference. > > What works: > If I explicitly set up an AA option set to transfer to the conference > (ie button #1 -> conference extension) then the caller is correctly > transferred into the conference. > > What does not work: > If, at the prompt, I just dial the conference number, then I hear the > "Please wait while I transfer your call" then the call drops. > > The REFER messages look nearly Identical in both calls, but the > following INVITE from the gateway is completely different. > > Working: > 17d:19h:29m:18s INVITE sip:71...@example.com SIP/2.0 > > Not-Working: > 17d:19h:22m:51s INVITE sip:_c...@example.com SIP/2.0 > > The non-workign one seems to loop back to the gateway (sipxecs send > the INVITE back to the gateway as if it does not have > :_c...@example.com > > Attached are some sanitized logs from the Audiocodes. Does anyone > have this type of scenario working? > > Thanks! > > Kyle > > > > > -- > Kyle Haefner, M.S. > Communication Systems Programmer > Colorado State University > Fort Collins, CO > Email: kyle.haef...@colostate.edu > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 10.0.1392 / Virus Database: 1520/3852 - Release Date: > 08/23/11 > > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- Kyle Haefner, M.S. Communication Systems Programmer Colorado State University Fort Collins, CO Phone: 970-491-1012 Email: kyle.haef...@colostate.edu _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ----- No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1392 / Virus Database: 1520/3852 - Release Date: 08/23/11 _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
