Hello all,
I've read as many of the asterisk balancing threads as I can find. Either
my situation is unusual or I simply haven't understood anything I've read.
In short, I'm building an web/phone mashup which uses Asterisk's AGI to get
its work done. My only users are on the PSTN connecte
Hello,
On 3/5/11 2:16 AM, Larry Baumbach wrote:
I am a newbie to the world of VIOP. I am attempting to set up an ATA with SIP.
I created a SIP at account iptel.com and received an email
confirmation stating "We are reserving the following SIP address for
you: sip:larry.baumb...@iptel.org".
Hi Bernhard,
this should be available now on master git branch -- unfortunately it
took me quite a while, due to lot of time offline during the beginning
of this year.
I did the basic tests to see if it preserves now the call-id and
from-tag after 401, can you test with Asterisk and tell if
Thank you for the tip, Ovidiu!
The problem was with my dictionary indeed. There were two attributes
with duplicate values of "1". I've fixed the dictionary, and now everything
works fine.
Thanks again!
Regards,
Fedor.
2011/3/5 Ovidiu Sas
> You need to check the dictionaries on your kamailio se
Hi Alex,
it took me quite a while due to traveling, but now the issue should be
fixed on git. Indeed there was an issue with the indexes when accessing
the xavp as PV.
Thanks,
Daniel
On 12/24/10 12:27 PM, Alex Hermann wrote:
On Friday 24 December 2010, Daniel-Constantin Mierla wrote:
On 12
