Hi Ranganathan I don't exactly understand what you mean, could you please give me more details? Thank you.
Regards Phinux On Mon, Nov 30, 2009 at 1:38 PM, M. Ranganathan <[email protected]> wrote: > 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 >
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