I to Originate channels using AMI.When the other end indicates the channel is ringing, I need to do some system notification work.Everything works great when the ITSP sends a 180 Ringing response.Through AMI events I see the channel state changed ..
Im trying to understand how to configure Asterisk 13s PJSIP stack. Ive read the pages in  and still have a couple of questions to ask.In my lab, Ive configured an Asterisk 13 box to act as an ITSP box and another Asterisk 13 to act as an IPBX.Im try..
Im trunking with an ITSP that, when treating an outbound to an unknown destination, either:- send a SIP error code (I cant be more explicit, at the moment),- or cast a pre-recorded audio message using Early Media.At the same time, Im also trunking w..
happy new year!I am still trying to make T.38 work. In the meantime, I have upgraded to Asterisk 13.1.0, and I am using the most recent PJSIP library (compiling that stuff myself). My local fax software is capable of T.38, as is my ITSP; Asterisk s..
Hey guys and gals-Right now, Im using FreePBX to handle providing voice services to a handful of customers. However, it just isnt cutting it for features, billing, customer access (portal stuff), etc. What do you recommend? Is there an ITSP portal/panel/platf..
Ive been running an Asterisk server (18.104.22.168.2) for over a year without any major issues. All of a sudden people are unable to login to their voicemail because Asterisk is seeing DTMF twice for each digit the caller pushes. Weve noticed the prob..
Guys, Seeing an issue with 22.214.171.124.2 and also 126.96.36.199 When we do call forwarding if the call coming in to be forwarded asterisk sends the invite out to our ITSP as firstname.lastname@example.org instead of username@domain. When call comes in with CLI ..
I know this questions is not really asterisk related, however I know a lot of people here are in the industry. I was curious if anyone here could provide insight on how to become a facilities based CLEC. I did a lot of google-ing and read info on voip-info…
If you are providing a hosted phone system to customers, how do you deliver the calls? If you are a end user, how does your provider deliver the calls to you? The reason I ask is I read and hear a lot of issues where people are getting dropped ca..
Were trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number. UK Landline->voipfone->asterisk 1.4->voipfone->UK landline About 1 i..