i think this also happens with cisco callmanager way back using h323. this is fixed (as far as callmanager is concerned) by a patch submitted to the mailing list a few months back by marian durkovic (search the archive). i don't think that patch reached the cvs though... or did it?
On Thu, 2004-05-20 at 02:46, Rechenberg, Andrew wrote: > Good day, > > I have a puzzling issue that people in the IRC channel recommended I > post to the list so here goes :) > > I am trying to call a SIP softphone from an H.323 hardphone. The > hardphone is connected to a Definity Prologix R12 PBX with a MedPro card > and a CLAN. The Avaya is setup to send any call to extension 1609 down > an H.323 trunk group that is destined for the Asterisk server. When I > call 1609 from my hardphone, my SIP softphone rings, I answer it, and > the call is established. However, there is only one-way audio during > the call, from the hardphone to the SIP client; not vice versa (from the > SIP client to the hardphone). > > I can see audio being injected into the SIP client via the client's > audio level meters so I don't believe the problem to be with the SIP > client. I also know that SIP to SIP works from my server because I > called another IRC user with my SIP client through the Asterisk server > across the Internet. > > I have disallowed all codecs except G.711 uLaw so I don't believe the > issue to be a result of mismatched codecs. A packet capture, and > debugging output from the Asterisk console show the call setup and then > there is just traffic between the hardphone IP, Asterisk, and the SIP > client. There is also no NAT involved in this call - the hardphone and > soft phone are on different 10.x.x.x networks only separated by a Cisco > switch/MSFC, but there is no NAT. > > All of my configs are standard from a 'make install' of Asterisk except > for h323.conf and sip.conf (shown below). Extensions.conf is stock save > for the extension I added for the SIP softphone. > > Does anyone have any idea what could be causing the one-way audio? > Below is an ASCII representation of the call setup, as well as my > h323.conf and sip.conf files minus comments, and the Asterisk server > setup and software. Any help on this issue is much appreciated. > > Thanks, > Andy. > > > > Call Diagram > -- > Hardphone --> Definity Prologix --> Asterisk --> SIP client > > -- Audio --> > > > Asterisk Server > -- > Fedora Core 1 with updates > kernel-2.4.22-1.2188.nptl_48.rhfc1.at > kernel-module-alsa-2.4.22-1.2188.nptl_48.rhfc1.at-1.0.4-23.rhfc1.at > alsa-driver-1.0.4-23.rhfc1.at > alsa-lib-1.0.4-12.rhfc1.at > alsa-utils-1.0.4-7.rhfc1.at > Openh323 1.12.2 compiled from source (no other RPMS) > Pwlib 1.5.2 compile from source (no other RPMS) > Asterisk CVS-HEAD-5/10/04-20:43:43 and CVS-HEAD-5/19/04-10:18:12 > Multimedia audio controller: Ensoniq ES1371 [AudioPCI-97] (rev 09) > > > Other gear > -- > Avaya Definity Prologue R12 with Metro and CLAN > Avaya 4612IP hardphone > SIP clients: Windows Messenger 4.7.2009, X-Lite 1103a > > > sip.conf > -- > [general] > port = 5060 > bindaddr = 0.0.0.0 > context = default > disallow=all > allow=ulaw > canreinvite=no > > [1609] > type=friend > host=dynamic > username=1609 > secret=password > mailbox=1609 > canreinvite=no > nat=no > > > h323.conf > -- > [general] > port = 1720 > bindaddr = 0.0.0.0 > canreinvite=no > disallow=all > allow=ulaw > dtmfmode=inband > context=default > > > > > _______________________________________________ > 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
