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hello, I just installed a2billing, I did all the config, at least I guess .. but I still can not integrate sip accounts that I had created (with sip.conf ) in a2billing to apply their billing .. could someone tell me how to make the junction between asterisk and a2billing?? I also noticed that the file additional_a2billing_sip.conf : was always empty ... -------------- next part --------------

..., that is yet to be discovered. What I want to ask is - how can I split SIP and RTP traffic? Say, SIP goes via VPN, but after the call is initiated, servers reinvite each other with real IPs. Is that possible at all? Searching on Internet didn't give me a clue. -- With Best Regards Mikhail Lischuk -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130215/aee9ec22/attachment.htm>

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension

Hi, Does anyone had the problem of asterisk SendFax + spandsp sending only the first page of a multi-page TIFF file? Seams to be related to spandsp ECM config. Any thoughts about it? Thanks, Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121005/ac471600/attachment.htm>

I have a SIP line that is working fine when I make calls from IP phones. I can send and receive calls. The problem is that if I try to dial from the CLI using the originate command or use an AMI connection to originate a call I get the following error: originate SIP/protel-out/0445540881644 application playback tt-monkeys WARNING[12950]: chan_sip.c:20437 handle_response_invite: Received

Hi, Am getting dead or silence calls at sometimes for my agents, when I checked my CDR the caller-id shows my vendor's name and some shows as real customer name. When I call back again the real customer's number its reaching, the answering machine owned by customer. I have a confusion, or how to find out are these numbers are from any auto dialer or from real customers. Thanks.

Hi a new small question ;=) We have two Asterisk, connected in IAX2. On the first, in dialplan, we have: exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)}) we sent into the IAXVAR "ACCOUNTID" the accountcode. On the second, in dialplan, we have: exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) That's work, the second server get the variable. I

...if it is launched on caller or callee side? My Asterisk version is Asterisk SVN-branch-1.4-r290100 and I really hope that I will not have to update it because it will be a long and hard quest due to old and broken PRI card drivers 8( Hope for some wisdom and help. -- With Best Regards Mikhail Lischuk ITX Ukraine -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111215/0d8461d2/attachment.htm>

So a few people just reported that they couldn't make any calls. I logged into asterisk and at first everything on the console looked normal, then I got swamped with messages about too many open files. This is from my asterisk/messages log file: [Oct 2 16:46:00] WARNING[19429] rtp.c: Unable to allocate RTCP socket: Too many open files [Oct 2 16:46:00] WARNING[19429] udptl.c: Unable to

Asterisk 11.1 Is it possible to catch the fact that an IP phone has died in the middle of a call and do something with it in the dialplan? Background: we run a small call center. Our agents sit in two groups, with their IP phones running from 2 different switches. Every once in a while the power on one side of the room will go out, or one of the switches will die, or one of the agents will