What application front end are you using to SipXtapi? sipXezPhone? PlaceCall?
Here are a few possibilities what it could be: Could be a NAT problem filtering out RTP. You might try analyzing the sipXtapi logs to see if media streams are being set up and sent to your Asterisk server. You can then also analyze your Asterisk logs to verify if media streams are received by your sipXtapi UA, and verify if media connections are being set up from Asterisk to the X-Lite UA. 1. Could be a problem with the microphone on your side. Have you selected the right device for Mic (how you do so depends on the application front end)? Thats a few things you can try. On 5/9/07, Xiaoming wang <[EMAIL PROTECTED]> wrote:
Hi keith, I rebuild version 9100, 8900, 8700. I test all versions. result: remote party can only hear. Is it possible some disble comminication with other SIP server instead of SIPX? Since version I check out on 2006-08-07 is work. Thanks Martin ----- Original Message ----- *From:* Keith Kyzivat <[EMAIL PROTECTED]> *To:* Jaroslav Libák <[EMAIL PROTECTED]> *Cc:* [email protected] *Sent:* Tuesday, May 08, 2007 11:57 AM *Subject:* Re: [sipxtapi-dev] branches\ SipXtapi Ahh you found the root of the problem? On 5/8/07, Jaroslav Libák <[EMAIL PROTECTED]> wrote: > > >9100 is currently pretty stable. > >The head of the sipXtapi branch currently is currently experiencing a > high rate of development, so I can't guarantee it will work 100% >entirely > correct. > >Reports of what is broken does help us though! > > I use r9100 with some leak fixes in executable wxCommunicator which can > be downloaded from sourceforge, so it can be tested without compilation. > > I have already reported a problem introduced in r9386 by Dan Petrie. > Dialing takes 1 second longer than before, so now it takes roughly 2 seconds > instead of 1. > > Jaro > _______________________________________________ > sipxtapi-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/ > -- Keith Kyzivat SIPez LLC. SIP VoIP, IM and Presence Consulting http://www.SIPez.com tel: +1 (617) 273-4000 ------------------------------ _______________________________________________ sipxtapi-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/
-- Keith Kyzivat SIPez LLC. SIP VoIP, IM and Presence Consulting http://www.SIPez.com tel: +1 (617) 273-4000
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