On Sun, Nov 29, 2009 at 7:51 PM, Phinux Zhang <[email protected]> wrote: > Hello All > > We are working on deployment of sipXecs 4.0.4 in our company, but we have > the following two problems related with sip trunk, could you please help me > to take a look and give me some suggestions? Thanks in advance for any > advices. > > 1. We used aster...@home as our production phone system, when I dial from > Asterisk to sipXecs, the AutoAttends(will use AA for short) works fine, but > I can't dial any extension from AA, AA keeps saying the extension is > invalid, but I am sure the extension I dialed is valid, and AA even can't > recognize the number specified on AA configuration page. > > 2. I can dial extension from AA when using PSTN line, but after I dialed one > extension, AA said "Please wait....", after that just silence, I can't hear > dial tone, but the phone I dialed ding ring, it's very strange. > > I hope I can fix this today, or we have to switch to other solution like > Trixbox. I am waiting for you, I really hope you professional guys can help > to figure it out. Thanks in advance, and if you need any information, just > let me know, I'll do what I can.
FreeSWITCH will only send to the port that it receives from ( i.e. it does its own NAT compensation ). This could be the issue you are running into - i.e. your RTP source is not symmetric. Ranga > > Thank you very much. > > Regards > > Phinux > > _______________________________________________ > 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/ > -- M. Ranganathan _______________________________________________ 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/
