[Asterisk-Users] Several H323 bugfixes - working SIP <-> H.323
translator
Kelvin Chua
kchua at up.edu.ph
Fri Mar 26 05:37:30 MST 2004
this patch however worked for me, all calls through the patched
chan_h323 are ok, hold, transfer, etc works perfectly. except that there
is no music on hold, while in fact asterisk shows that it is playing,
yet there is no audio heard on the callmanager side.
so i had test on both oh323 0.5.10 and h323(patched) cvs march 20 the
problem with oh323 is that when a call is placed on hold by a
callmanager phone, after resuming, the audio from * to ccm is lagged by
3-4 seconds. while the audio from ccm to * is ok. i already posted this
problem in version 0.5.5. has anybody found a workaround for this?
On Fri, 2004-03-19 at 18:25, Paul Cheng wrote:
> Hi,
>
> The patches also did not help us and in fact created some new problems.
> The old chan_h323 could pass on early audio and provider messages, but
> after the patch, this capability is gone and the channel only rings and
> rings while the provider is sending the message.
>
> We've had no problems with the existing chan_h323 other than that it
> doesn't return the right indication state to Asterisk, so Asterisk
> can't branch for busy versus congestion.
>
> But this is obviously only for our setup.
>
> On Mar 19, 2004, at 9:12 AM, Marian Durkovic wrote:
>
> > On Thu, Mar 18, 2004 at 12:22:57PM -0500, Billy Huddleston wrote:
> >> I just tried this, and it's not working for me.. I can't call a 2600
> >> or a
> >> CCM... What version of OpenH323 and PWLIB did you all use?
> >
> > Are you able to call those without the patches? If not, the patches
> > won't
> > help you, since you probably have some other problem..
> >
> > M.
> >
> >>
> >>
> >> ----- Original Message -----
> >> From: "Marian Durkovic" <md at bts.sk>
> >> To: <asterisk-users at lists.digium.com>
> >> Sent: Thursday, March 18, 2004 10:35 AM
> >> Subject: [Asterisk-Users] Several H323 bugfixes - working SIP <->
> >> H.323
> >> translator
> >>
> >>
> >>> Hi all,
> >>>
> >>> in an effort to create a SIP <-> H.323 translator we've found and
> >>> fixed
> >>> several problems in H.323 channel. These inlcude:
> >>>
> >>> for SIP->H.323 calls
> >>>
> >>> - no ringback tone
> >>> - ringback not related to H.323 events
> >>> - one-way audio with Cisco CallManager
> >>> - incorrect Caller ID
> >>>
> >>> for H.323->SIP calls
> >>>
> >>> - not able to establish call with Cisco IOS 12.3(4)T
> >>> - ringback not related to SIP events
> >>> - no support for 183 Call Progress
> >>> - incorrect Caller ID
> >>>
> >>>
> >>> Please find the patches against aterisk 0.7.2 release below.
> >>>
> >>>
> >>> M.
> >>>
> >>>
> >>> ---------------------------------------------------------------------
> >>> -----
> >>> ----
> >>> ----
> >>> ---- Marian Durkovic network manager
> >>> ----
> >>> ----
> >>> ----
> >>> ---- Slovak Technical University Tel: +421 2 524 51 301
> >>> ----
> >>> ---- Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
> >>> ----
> >>> ---- 812 43 Bratislava, Slovak Republic E-mail/sip: md at bts.sk
> >>> ----
> >>> ----
> >>> ----
> >>> ---------------------------------------------------------------------
> >>> -----
> >>>
> >>
> >
> >
> > -----------------------------------------------------------------------
> > ---
> > ----
> > ----
> > ---- Marian Durkovic network manager
> > ----
> > ----
> > ----
> > ---- Slovak Technical University Tel: +421 2 524 51 301
> > ----
> > ---- Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
> > ----
> > ---- 812 43 Bratislava, Slovak Republic E-mail/sip: md at bts.sk
> > ----
> > ----
> > ----
> > -----------------------------------------------------------------------
> > ---
> > _______________________________________________
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>
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