[Asterisk-Users] More HEAD wierdness (chan_sip, jitterbuffer/PLC
problems) -UPDATE
Kristian Kielhofner
kris at krisk.org
Sat Mar 19 18:44:40 MST 2005
Kristian Kielhofner wrote:
> Hello,
>
> After checking out CVS HEAD from yesterday (for those new
> PLC/Jitterbuffer patches), I was affected by bug 3795 with my Polycom
> IP600's. After seing it resolved as of this morning (thanks Mark), I
> decided to try again...
>
> I can answer incoming calls. No problem there. Putting calls on
> hold, however, results in my Polycom IP600 indicating the call on hold,
> but the caller does not hear any music. If I go through enough cycles
> of holding/resuming the call, the IP600 eventually locks up in hold,
> with the resume softkey doing nothing. The only way to get out is to
> have the caller hangup or reset the phone. If the caller hangs up, the
> Polycom drops the call, and a few seconds later the caller's phone rings
> as though the Polycom was calling it back. If you answer the call you
> hear nothing and the call is dropped after about one second.
>
> Similar behavior on my 7960. I can place the call on hold, but the
> caller hears no music. Same with before, if I cycle hold/resume enough,
> the phone locks up. Only the Cisco just drops the call.
>
> The SIP debug of the conversation between * and the IP 600 (firmware
> 1.4.1) can be found below. I can provide the Cisco as well (7.2).
>
> http://www.krisk.org/asterisk/sip_debug.log
Fixed in CVS. Thanks again.
> Now for more on MOH. I am using native MOH with my files in ulaw
> format. I setup an extension for running MusicOnHold, and have been
> listening to my hold music for 8 minutes with my Polycom IP600, so I
> know that * sees the files and can play them. However, if someone calls
> in over IAX and executes that same extension, MOH plays for a couple of
> seconds and stops. No word on the * console as to what happened.
>
> I use g729 almost exclusively. I am confused as to the status of
> g729 with the new PLC/jitterbuffer stuff, but that doesn't seem to be it
> because I can transcode from ulaw -> g729 on the phone with my MOH
> extension with no problems. "sip show channels" shows that the channel
> is using g729, and the MOH plays for forever and ever. My iax.conf has
> trunk=no trunktimestamps=no and jitterbuffer=yes. If I set
> jitterbuffer=no, MOH works perfectly. So it appears to be the
> jitterbuffer in IAX that is causing problems with MOH there.
>
> I hope that this message wasn't too verbose, but I hope that any of
> these PLC/jitterbuffer problems can get ironed out, because it looks
> awesome!!!
Sorry to reply to my own post, but here is a new interesting tidbit of
information: If a call comes in and dials my MOH extension, it plays MOH
for a few seconds before stopping. If a call comes in, dials a SIP
extension and answers the call, the audio is great (two humans talking).
If I place that call on hold, the MOH sounds fine too (for the 2
minutes that I listened). Truely strange.
I don't think I mentioned that the other end of that IAX2 connection is
NuFone. But I have tried SIxTel (iax.cc) and it has the same result.
--
Kristian Kielhofner
More information about the asterisk-users
mailing list