Anyone please explain me How Account code is use for billing.,
I registered sip phone(X-Lite) on Asterisk 188.8.131.52 on CentOS 5.5 64 bit but
not successful, Can anyone help me to do it?
Thanks and best regards.
Is there an option in sip.conf for 1.6.2 that would add this to the INVITE? Supported: ms-early-media In my invites I see: Supported: replaces, timer But I have not seen any option that would add the ms-early-media option. Here’s a link to the RFC3..
I have two questions for the group. #1 – Im looking to use some GSM SIM cards with my Asterisk PBX. Can anyone recommend a gateway?I need about 10-15 SIM slots. #2 – Im also looking to connect Asterisk to an SS7 signaled DS1 (24 channels) for inbo..
What the score with IPv6 in Asterisk now? Ive had a google about and found the http://www.asteriskv6.org/ site but if the filename on the download link is anything to go by its a few years old… (And a weird anti-download firewall!) Cheer..
Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide?Thanks and hope the community will not reject my curiosity! :)Best Regards,ValluSevan..
Asterisk is making a call to a peer. In 200 ok, peer is sending its application server ip in contact field, so asterisk sends ACK to that IP. RTP starts flowing between endpoints and peer plays an IVR and asks for destination number. After entering destinat..
Ive recently installed Asterisk 184.108.40.206. Id like to connect GSM Trunk to it. I purchased a few Mobigater ProOpen gateways. It states that I should use chan_celliax module to it. On the gsmopen site I see a comment in the documentation that I can inst..
Dear asterisk users, A few weeks ago Ive been attacked by a DOS on REGISTER that Ive solved with a fail2ban script. Now, since a few hours, I have my asterisk 220.127.116.11 running at 100% CPU again. Ive checked the log and it shows nothing related to fai..