I am having a similar problem with one-way audio from an Avaya hardphone calling a SIP soft phone. Audio from the hardphone is heard on the receiving end (SIP), but audio is not heard on the hardphone. I know audio is being injected into the sound card and being processed by the SIP client (I've tried both X-Lite and Windows Messenger 4.7.2009) because the audio meters show signal coming into the client however nothing is heard on the other end.
I am using the following: CVS-HEAD 5/21/04 Pwlib-1.5.2 Openh323-1.12.2 Regards, Andy. > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of T. Chan > Sent: Tuesday, June 01, 2004 1:25 PM > To: Dmitry Mishchenko; [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] RE: H323 > > Dear All, > > Thanks, but I was already using a pre May 25 CVS version. > Does anyone else > have any idea please? Thanks > > TC > > -----Original Message----- > From: Dmitry Mishchenko [mailto:[EMAIL PROTECTED] > Sent: Tuesday, June 01, 2004 6:22 AM > To: [EMAIL PROTECTED]; T. Chan > Subject: Re: [Asterisk-Users] RE: H323 > > > On Tuesday 01 June 2004 00:56, T. Chan wrote: > > Dear All, > > > > I have used Asterisk for a few months and I have been using > a January CVS > > version, it has been working but has been regularly crashing. I use > > Asterisk mostly as a softswitch application receiving H323 > calls from > > customers and send to H323 carriers. Since I have been > using an older CVS > > version, the OpenH323 and Pwlib libraries in use have been > 1.11.7 and > > 1.4.11 > > respectively. > > > > I was thinking of using a current asterisk version and see > if it is more > > stable comparing to the version in use. I upgraded to new > version, and I > > understand that with the new version and the H323 code, I > need to use the > > 1.12.2 and 1.5.2 versions of the OpenH323 and Pwlib libraries > respectively. > > I have, therefore, erased the whole Pwlib and Openh323 > folders, recreated > > with the new versions and did the ./configure.....make > clean.....make opt > > procedures to compile the drivers. > > > > I have then compiled all the zaptel, libpri, asterisk as > usual, but when I > > ran the asterisk, I noticed that most calls (calls mostly > were sent from > > Cisco AS5300 and Cisco AS5350) were getting one way audio, > the calling > > party was not able to hear anything even the call was > connected, I am not > > sure if the called party would hear anything, but obviously > something is > > not working properly. > > > > I have not exactly the same but rather similar issue with the latest > cvs-head. > There are recent changes in call of on_start_logical_channel() > After moving it to > MyH323_ExternalRTPChannel::OnReceivedAckPDU it stopped > being called in my configuration. As a result I don't get any > audio after > call established. And with older approach when > on_start_logical_channel was > called at MyH323Connection::OnStartLogicalChannel it was working fine. > This change was done on May 25 so you may try to use older > code from CVS > before this date. > Jeremy saying the latest version approach is fine, but its > not working for > me :(. > > Dmitry > > > Can any of your experts out there help please, thanks? > > > > TC > > --- > > Outgoing mail is certified Virus Free. > > Checked by AVG anti-virus system (http://www.grisoft.com). > > Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 > > --- > Outgoing mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
