Timothy R. McKee wrote:
I just ran an SVN update to attempt resolution of this issue and now there
is a completely different issue...Very strange.
1. inbound call comes into phone A and is answered.
2. transfer button pressed
3. number of phone B is entered
4. phone B rings and is answere
I just ran an SVN update to attempt resolution of this issue and now there
is a completely different issue...Very strange.
1. inbound call comes into phone A and is answered.
2. transfer button pressed
3. number of phone B is entered
4. phone B rings and is answered. audio between A and B
On Mon, 2006-09-04 at 08:44 +1000, Avi Miller wrote:
> Avi Miller wrote:
> > Is there a Wiki page or similar describing how to checkout SVN for
> > Asterisk? Also, will I need to checkout and compile SVN versions of
> > Zaptel/Libpri/Addons (as I use all three)?
>
> Replying to myself to say tha
Avi Miller wrote:
Replying to myself to say that I've found Digium's instructions and I'm
testing SVN on my test server now. :)
And again to say that it seems work just fine with the SVN code. Thanks
Kevin!
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National Manager - Special Projects
< Sydney / Melbourne / Canberra / Hobart / Lo
Avi Miller wrote:
Is there a Wiki page or similar describing how to checkout SVN for
Asterisk? Also, will I need to checkout and compile SVN versions of
Zaptel/Libpri/Addons (as I use all three)?
Replying to myself to say that I've found Digium's instructions and I'm
testing SVN on my test se
Kevin P. Fleming wrote:
if could download SVN branch-1.2 and try it out on your system to see if it
solves your issue.
Is there a Wiki page or similar describing how to checkout SVN for
Asterisk? Also, will I need to checkout and compile SVN versions of
Zaptel/Libpri/Addons (as I use all thr
The svn branch-1.2 is very stable, probably more stable then the rpms
and other distro's out there, as fixes are applied when problems are
identified and corrected. Sometime later, the svn branch-1.2 is used to
create packages.
Kevin Smith wrote:
Well personally I am just glad I wasn't the on
Well personally I am just glad I wasn't the only one seeing the
problem. As much as I don't like the place 100% of the blame on
something unless I fully know what is going on, in this case Asterisk,
but I couldn't see any solution but a bug.
Personally I wouldn't mind testing out the branch,
- Richard Scobie <[EMAIL PROTECTED]> wrote:
> Dave Fullerton wrote:
> >
> > I just verified it here as well. Running Asterisk 1.2.11 and two
> polycom
>
> I'll throw in a "me too" here, with the addition that it also occurs
> with "canreinvite=no".
There were multiple problems in this area
Dave Fullerton wrote:
I just verified it here as well. Running Asterisk 1.2.11 and two polycom
I'll throw in a "me too" here, with the addition that it also occurs
with "canreinvite=no".
Regards,
Richard
___
--Bandwidth and Colocation provided
I just verified it here as well. Running Asterisk 1.2.11 and two polycom
phones running 1.6.7 firmware with canreinvite=yes. Putting the call on
hold and then off does bring the audio back. From what I can tell by
looking at the lights on my switch, something is still sending the RTP
traffic
Hi Avi,
I had a similar problem. Have extension 405 put the call on hold (after
the transfer) and then off hold. I am willing to bet it will bring back
the audio stream. I posted something similar a few weeks ago and if
anyone thought it was a bug, to let me know what information I needed to
Hey guys,
I've been trying to change my Asterisk setups to use canreinvite=yes.
I'm having a small problem with my Polycom IP501 phones and transferring
calls.
If a call comes in via my ISDN BRI lines (using chan-capi), I can
successfully transfer the call using the Polycom Blind Transfer op
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