On Mon, 2009-11-30 at 10:46 +0000, Nitin wrote: > Hello > > We have successful trixbox running since last 1 year and I have tried many > times to replace it with sipx(since 3.10) before I finally give up, I ask > this > forum the problems I am facing. > > Our setup is one Vegastream50 Europa and one GXW4104 used as gateway and > approx > 20 Polycom 550+ 6xCisco7975. Trix works fine. > > For experiment, I switch off trix, and setup sipx at the same IP address. I > setup all users and pins same as the trix (so that I donot have to > reconfigure > anything). > > I have tried the following with the results - > a) Flipping the polycoms to ftp - this way, they are autoprovisioned by sipx > and works most of the time. > > b) Tries using ciscoplus to generate the Cisco xml file - It generates buggy > file(but reading - I came to know I had to insert the processnodename). > Cisco > registers after that. > > c) Vegastream - Never was able to make it work. In dial plan, sending to > TA:sipx_address never worked. But sending to TEL:100 (autoattendant) works. > > d) GXW4104 - switching became extremely unreliable(though it is super > reliable > with trix). While autoattendant is playing, there will be strange noises and > the number will be busy. I guess codec neg. issue with sipx. It never has > happened with Trix. To be sure, I upgraded the firmware to latest. (1.25.xx) > > e) Since Vega was able to direct calls to AA, I tested to find that AA will > only trnasfer calls to Polycoms. All others - ext not avail. But Polycom can > call Cisco, so I believe there is bug with AA. And no - its not DTMF issue as > I > tried even on timing out, it doesnot go to defualt extension(which happens to > be cisco). > > e) 3CX softphone as test had hit and miss call transfers. Bria - worked most > of > the time. > > So, my findings are - > a) Sipx works only with Polycom > b) Sipx has serious issues with Vegastream and Grandstream gateways. > c) Sipx doesnot like Cisco and eventhough it stays regsitered, its lying!! > > Please comment - Is sipx is designed to work ONLY with polycom and VOIP > providers?
No - sipXecs is designed to work with any good SIP implementation. > Its not supposed to be replacement of asterisk? Being an asterisk replacement is not a design goal. Your problems are probably all solvable, but it won't work to just put the sipXecs system where the asterisk system was and not change anything else. If you'd like help with working out the issues, I suggest starting by choosing just one problem, carefully describing your complete configuration, and collecting traces of the network traffic for that one problem and posting them here. See: http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer#Getting_SIP_Messages_to_display _______________________________________________ sipx-users mailing list [email protected] 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/
