[Asterisk-Users] One-way audio with H.323 --> SIP call

[Kelvin Chua]

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
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users