I have been told that my logic in extentions.conf is wrong in trying to configure a SIP software phone called Express Talk (remote) . Id like to make outgoing calls and calls to local extensions. Could someone please look at my configuration files..
nice people 🙂 I have been struggling with trying to get Zaptel from http://www.slimey.org/zaptel-solaris.tar.gz on a Solaris 5.11 VM I obtained from the OpenSolaris Website. I have tried installing all the necessary packages, yet I keep getting err..
I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant High-Availability.I was wondering if its possible for Asterisk to also use multiple database servers for Realtime?Currently with Realtime I am only able to point to a sin..
Sorry never mind! I got it to work after sof-linking to /lib/, and loading res_jabber.so first, chan_gtalk.so second. So in summary: ln -s /usr/local/lib/lib/ asterisk-cli>modules load res_jabber.so asterisk-cli>modules load chan_gtalk.so Cheers! *Jo..
I am having a problem with a Rhino channelbank.I have an Asterisk server running 18.104.22.168 and DAHDI 2.4.0 on a CentOS 5.5 system.We have a TE420 card with the first port used in E1 mode (R2, 20 channels) and the fourth is in T1 mode for the channelbank..
All, Id just like to verify what the correct operation of the timeout parameter is for the dial application. Im not sure if Ive encountered a bug or a configuration issue. When a sip phone is not responding to invites on an outbound call, the dial applicat..
all, I have a server which is sending INVITEs with a From and Contact header that contains a domain part of the address (an IP address) that I cant explain. My sip.conf does not set a domain. For example in the following line the 123.456.789.012 is ..
On 30 Nov 2010, at 09:47, Michael Nausch wrote: > I tried to configure Asterisk 1.8 on one of my test-hosts. > > Ive installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/): > > [Nov 30 10:35:53] WARNING[728..
All; Can I run the IAX on TCP port instead of UDP port? If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be bet..
Does anyone know how to check the TRANSFERREDTarget Number is a
local extension ora PSTN number.