Hi Josh Thank you for the info. I read the sipXbridge page, yes, as you guess, "dead air" problem has been fixed.
And about AA problem, actually, we have PSTN gateway and IAX gateway configured on Asterisk, if I use PSTN gateway to dial sipXecs AA, I can dial exntensions, but if I use IAX gateway, I can't. Just for sure, you mean if I add dtmfmode=inband on IAX configuation page will solve this problem, is that right? The aster...@home currently is in production mode, I wouldn't touch it unless I am sure. Thanks again. Regards Phinux On Mon, Nov 30, 2009 at 3:22 PM, Josh Patten <[email protected]>wrote: > It would appear to me that something is not running through sipXbridge or > there is a misconfiguration with one of your gateways > > what type of PSTN gateway are you using, or does it run through Asterisk? > > Did you set Asterisk to point to port 5080 on your sipX box after > configuring sipXbridge? sipXbridge runs on port 5080. > http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configurationfor > more info on using sipXbridge. This should fix the "dead air" after the > "Please hold while I transfer your call" message > > I am 99% sure that the AA problems are related to DTMF. try setting your > DTMF settings to one of the options here: > http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode > > If you are using an old version of Asterisk (1.2 or so) then you cannot > directly connect the two systems, you have to use an "intermediary" like > sipXbridge due to the inadequacies of Asterisk's SIP stack. > > If you need more Asterisk configuration information I will try to oblige, > as I know a bit about Asterisk. > > Phinux Zhang 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. > > 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/ > > >
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