Hi,
I had a hard time finding this list since I couldn't find it on
www.ekiga.org btw.
Anyway I have two left open issues with Ekiga 3.0.x. I'm using it for
daily conference calls via SIP and did so with Ekiga 2.
So the following issues are actually regressions from Ekiga 2:
I can't find a way
Damien Sandras schrieb:
> Le mardi 20 janvier 2009 à 17:37 +0100, Wolfgang Rosenauer a écrit :
>> Hi,
>>
>> I had a hard time finding this list since I couldn't find it on
>> www.ekiga.org btw.
>>
>> Anyway I have two left open issues with Ekiga 3.0.x
Damien Sandras schrieb:
> Le mardi 20 janvier 2009 à 19:04 +0100, Wolfgang Rosenauer a écrit :
>> Damien Sandras schrieb:
>>> Le mardi 20 janvier 2009 à 17:37 +0100, Wolfgang Rosenauer a écrit :
>>>> Hi,
>>>>
>>>> I had a hard time finding th
Damien Sandras schrieb:
> Le mardi 20 janvier 2009 à 19:28 +0100, Wolfgang Rosenauer a écrit :
>> Not before each call in my special case. I have one provider which is
>> only able to call real phones and one (ekiga.net) I'm usually use to do
>> real VOIP calls to SIP end
yannick schrieb:
> Le mardi 20 janvier 2009 à 22:57 +0100, Wolfgang Rosenauer a écrit :
>> hum? Probably I misunderstand but how would auto-completion help me in
>> that case?
>> Basically I have two connection types:
>>
>> ekiga.net -> some_sip_addr...@som
Damien Sandras schrieb:
> Le mercredi 21 janvier 2009 à 00:54 +0100, Wolfgang Rosenauer a écrit :
>> yannick schrieb:
>>> Le mardi 20 janvier 2009 à 22:57 +0100, Wolfgang Rosenauer a écrit :
>>>> hum? Probably I misunderstand but how would auto-completion help me in
Hi,
yannick schrieb:
> Le mercredi 21 janvier 2009 à 00:54 +0100, Wolfgang Rosenauer a écrit :
>> yannick schrieb:
>> ok, thanks for the hint. But probably I'm still missing a point. How do
>> I make sure that some_sip_addr...@somedomain.com gets called through
>>
Damien Sandras schrieb:
>> I've thought that be cause the Asterisk I connect to shows the following
>> information about my connection:
>>
>> wrosenauer Wolfgang Rosenauer Channel: SIP/ekiga.net-88031130
>>
>> So my Ekiga.net address is transmitted
Hi,
the following could be just pilot errors. I'm not yet a power user of
ekiga but need a reliable SIP client.
So I started to test 3.2.6 with Gnome 2.28pre but ran into some issues
which makes it not really usable currently. I'm trying to call a
physical phone through a VOIP provider (1und1) (a
On 10/07/2009 02:05 PM, Michael Rickmann wrote:
> Before going into details, sip.1und1.de is a "limited" server in that it
> only accepts public IP adresses. If you are using Ekiga from behind a
> NAT router or a computer with several interfaces you will have to use
> Ekiga's not well documented wo
Hi,
On 10/07/2009 02:29 PM, Eugen Dedu wrote:
>> was still there but then it stopped. Trying to hangup the call resulted
>> in a frozen ekiga which had to be killed forcefully.
>
> Please send us a gdb backtrace, see the wiki debugging section.
Just was in a call when sound output suddenly stopp
On 10/09/2009 11:06 AM, Eugen Dedu wrote:
> Both logs are not useful. Please always start ekiga like this:
> gdb --args ekiga-snapshot -d 4 2>&1 | tee gdb-ekiga.txt
> and when the freeze appears, type Ctrl-c, afterwards thread apply all
> bt, as shown at
> http://wiki.ekiga.org/index.php/Debuggin
On 10/13/2009 02:23 PM, Damien Sandras wrote:
> Le mardi 13 octobre 2009 à 11:13 +0200, Wolfgang Rosenauer a écrit :
>> Attached is a full backtrace as created by the above command after the
>> audio stopped one way and tried to hang up.
>
> Is it reproducable if you disabl
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