Rechenberg, Andrew
2004-May-19 11:46 UTC
[Asterisk-Users] One-way audio with H.323 --> SIP call
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
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@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