hi folks.
i've been experimenting with SILK codec and meet with some
success on incorporating it in pjsip (an open source sip client).
now i'm trying to do the same thing on Asterisk. any documentations,
pointers, etc i should look into? any help is appreciated.
--
Edwin Lam
I still can't figure out why this isn't working. I updated to the latest
version of Asterisk 1.8.1 with no luck. I am using Realtime for sipusers
and vmusers if that makes any difference. I tested this on a new install
and saw the following
under the folder where I installed Asterisk I had
Hi. I have an asterisk system under Debian Leni using asterisk 1.8 with
no Digium hardware -- and when I go into a meetme conference the system
either locks up or is 100% cpu utilized or something -- I can't type
anything and I have to physically reboot the system. The dahdi module is
loaded and
On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
Are there reasons to prefer the use of PRI over SIP or SIP over PRI?
Assuming you are talking to connect a PBX to the PSTN...
PRI advantages:
1. Relatively little equipment between the PTSN and the PBX. Less to
break or
Particularly interested if anyone has it working on Fedora 13+, with the
Fedora RPMS. I've tried both F13 (asterisk-1.6.2.12-0.1.rc1.fc13.i686)
and Rawhide (1.8 something), and both of them appear to be broken in the
same way.
Festival reports a connection, and a file is placed in the cache
On Jan 6, 2011, at 8:56 AM, Kevin P. Fleming wrote:
On 01/05/2011 08:12 PM, Thomas Rymes wrote:
OK, after my last message about fax detection, I feel a bit better informed
and able to press forward. I started looking into this because I was getting
lots of false positive fax detection
On Jan 6, 2011, at 10:05 AM, Andy Graybeal wrote:
On 01/05/2011 01:51 PM, Tom Rymes wrote:
On 01/05/2011 7:50 AM, Andy Graybeal wrote:
We've got two noisy kitchens that need to talk back and forth.
Andy,
Why, exactly, are you trying to combine an inter-kitchen intercom and
your phone
Thanks but I doubt it does pop-up of outlook contacts. It probably only does
outbound calling.
My main need is to have an outlook contact pop-up when a call comes in.
I also favor open source if possible.
Thanks
On Wed, Jan 5, 2011 at 4:02 AM, Giorgio Incantalupo
gincantal...@fgasoftware.com
On Thu, Dec 30, 2010 at 10:10:18AM +0100, Stefan Schmidt wrote:
my idea was if i can find a way that the first call of a peer has g711a
codec (like normally) and if a second call comes in, or has to be placed
for this peer i only offer g726 (40kbit) so i dont have a bandwith issue.
Hello,
I have been asked to implement the following design:
Load-balanced Kamailio servers handling registrations and routing.
Load-balanced asterisk feature servers handling voicemail and other things
Kamailio cannot do. Plus several load-balanced gateways, but they are not
relevant to my
Hi,
The channel name for DAHDI channels has changed in 1.8 with no information that
I can find in the ChangeLog.
The old format was DAHDI/XX-Y where XX was the real channel number.
It has changed to DAHDI/iZ/XX-YYY where XX is the callerid.
And Z is the number of the span in
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