On 8/28/07, Stefan van der Eijk <[EMAIL PROTECTED]> wrote:
>
>
> On 7/9/07, Noah Miller <[EMAIL PROTECTED] > wrote:
> > Hi Stefan -
> >
> > > What I want to accomplish:
> > > - calls within the LAN are re-invited (RTP goes from endpoint to
> e
On 7/9/07, Noah Miller <[EMAIL PROTECTED] > wrote:
>
> Hi Stefan -
>
> > What I want to accomplish:
> > - calls within the LAN are re-invited (RTP goes from endpoint to
> endpoint)
> > - asterisk detects when a call is going beyond the local LAN (over the
> NAT),
> > and then stays in the middle.
>
On 6/5/07, Tom Rymes <[EMAIL PROTECTED]> wrote:
On Jun 5, 2007, at 9:46 AM, Cosmin Prund wrote:
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto: asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Henry Cobb
>> Sent: Tuesday, June 05, 2007 4:30 PM
>> To: Asterisk Users Mailing Li
I've been trying to get google talk to work, but no luck yet:
1. when the jabber / google talk modules are loaded, asterisk ends up
consuming all the CPU. This happens after a while (up to a day), not right
after asterisk is (re-)started.
2. While i've been able to register a google talk
On 2/11/07, Stefan van der Eijk <[EMAIL PROTECTED]> wrote:
Applied the patch, and when I call the gmail account registered on my
asterisk server. Asterisk didn't crash (like it used to do before).
I've had to restart asterisk a number of times over the last few days due to
Applied the patch, and when I call the gmail account registered on my
asterisk server. Asterisk didn't crash (like it used to do before).
However, I still can't receive gtalk calls on my asterisk server. I tried to
call from my gtalk client ([EMAIL PROTECTED] --> hosted google account) to my
aste
On 2/10/07, Luki <[EMAIL PROTECTED]> wrote:
Stefan,
> When I have 2 SIP endpoints that both aren't configured with
> "canreinvite=no" then I get no sound.
The Sipura 3102 definitely works fine with canreinvite=yes and I never
really had a problem with any of the Sipura devices in this respect,
imes.
I'd really like to get this working...
with kind regards,
Stefan van der Eijk
On 2/10/07, Il Neofita <[EMAIL PROTECTED]> wrote:
Hi,
I tried thousands of time and finally I am a step closer to the solution.
I recompile iksemel with the option --prefix=/usr
I erase my zaptel-
und.
Questions:
1. Are all of my SIP endpoints incompatible with the canreinvite=yes
option?
2. Is there a list of SIP endpoints that are known to work with
"canreinvite=yes"?
3. Are other people also experiencing this?
with kind regards,
Stefan van der Eijk
_
On 1/25/07, Matteo Brancaleoni <[EMAIL PROTECTED]> wrote:
Hi,
On Thu, 2007-01-25 at 08:03 +0100, Stefan van der Eijk wrote:
> Hi,
>
> I'm experiencing an issue with my x86_64 machine containing a
> Hauppauge PVR-500 (ivtv) and a Digium TDM400p (wctdm, part of zaptel)
>
06 taz kernel: ivtv0 info: read 4096 from encoder MPEG, got
4096
Jan 24 20:04:06 taz kernel: TDM PCI Master abort
(ivtv 0.10 trunk, debug level = 511).
Could this have something todo with these two pieces of hardware fighting
about DMA?
with kind regards,
Stefan van der Eijk
__
On 12/17/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
On Sun, Dec 17, 2006 at 12:35:41PM -0600, Kevin P. Fleming wrote:
> Samy Antoun wrote:
> > I noticed that the sound directory is missing from
asterisk-1.4.0-beta4.tar.gz.
>
> This is incorrect; the sounds directory is present and contains two
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