Re: [sipx-users] Call Forwarding: Sipxecs with Asterisk MediaGateway

[Josh Patten]

Correct. The problem with Asterisk 1.6, FreeSWITCH, and YaTE is that they freak out when dealing with REFER on an attended transfer (YaTE simply doesn't support REFER). For the time being I was using Asterisk 1.6 to handle REFER which was working in all cases except for when doing an attended transfer and when call forwarding was enabled "at the same time" on an extension. Freeswitch was able to handle the call forwarding just fine, but flopped on an attended transfer. YaTE simply responded with a "501 not implemented" error to any REFER packets sent to it. My Adtran device simply ignores REFER. While I agree that full REFER should be implemented in all SIP devices, the reality is it's just not there on a lot of devices and software. Tony Graziano wrote: But this is straight to your adtran and no freeswitch or asterisk media gateway? On Mon, Nov 16, 2009 at 11:09 AM, Josh Patten <joshpat...@gmail.com> wrote: I  am happy to report that with sipXbridge patch21 there have been no dropped calls or one way audio reported so far this morning (about 2 hours worth of calling). I'll post and create a bug report if this changes. M. Ranganathan wrote: > On Sat, Nov 14, 2009 at 1:00 AM, Josh Patten <joshpat...@gmail.com> wrote: > >> It also appears that YaTE is the same way. that one was a little easier to >> set up, but it's the same old song and dance: REFER trips it up every time. >> >> I really wish sipXbridge was stable for me. Even with patch20 I drop to one >> way audio on my local LAN after about 5 minutes and locations that are just >> a couple of milliseconds ping away from the bridge software drop calls >> completely after a couple of minutes, and it's not my Adtran router, even >> when I'm bridging to other SIP devices this happens. I've sent Ranga a >> snapshot before but because we thought it was my Adtran router it never went >> anywhere. perhaps I should submit a bug with , or do you still have my >> original snapshots Ranga? >> > > > > Disclaimer: One should note that until our QA signs off on any given > version, that is to be thought of as a development versions that are > placed at your disposal for early testing. These snapshots are > unofficial updates to help those who do not wish to build source code. > >  We had recently made several changes as a result of interop testing > with Nortel CS1K and hence the inevitable regression that followed. It > might be that you hit something like that. It is going through QA and > I updated the patch again.  ( Our QA is pretty thorough but the nature > of problem is that it is very laborious and takes a while to do. > Thanks for bearing with us on that. ) > > Please feel free to send me a sipx-snapshot ( see instructions on > trouble shooting on the SIpXBridge wiki page ) after checking the logs > to make sure it is not something obvious on your end. > > > Regards, > > Ranga. > > > > > > > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/

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