I am having and issue I hope someone can help with.. I have calls that often come in that need to be blocked. We wish to do this without answering the call. The issue is our carriers have fail over servers and will try sending the call from each when we block the call. If we send a hangup with a Sip 401 they will stop the route advance on their end. The issues is we have been sending a hangup/cause code 21 (Call rejected) But they are receiving a 403 Forbidden.. Is there any hangup code that we can…
We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI card in it connecting to the PTSN. The frontend handles 99% of dialplan logic and just hands off anything outgoing to the backend via IAX2, which dials out on one of the open channels.
Lately we've been getting a disconnected calls. Keeping the consoles running it doesn't seem to be the PRI initiating the hangups, as I'll when I see hangups intiiated on the backend / PRI side:
-- Span 2: Channel 0/21 got hangup request, cause 16
Instead, I'm seeing
This morning someone tried to make sip call through my Asterisk. My server just drop these calls and record them in CDR with IP address:
2012-11-28 06:30:51 SIP/216... 1000 "1000" <1000> hangup 999011972592249388 ANSWERED 00:01 Hacker: 220.127.116.11 2. 2012-11-28 06:30:49 SIP/216... 1000 "1000" <1000> Hangup 88011972592249388 ANSWERED 00:01 Hacker: 18.104.22.168 3. 2012-11-28 06:30:46 SIP/216... 1000 "1000" <1000> Answer 99011972592249388 ANSWERED 00:02 4. 2012-11-28 06:30:43 SIP/216... 1000 "1000" <1000> Answer 1011972592249388 ANSWERED 00:02 5. 2012-11-28 06:30:39 SIP/216... 1000 "1000" <1000> Hangup 2011972592249388 ANSWERED 00:00 Hacker: 22.214.171.124 6. 2012-11-28 06:30:33 SIP/216... 1000 "1000" <1000> Hangup 7011972592249388 ANSWERED 00:01 Hacker: 126.96.36.199 7. 2012-11-28…
Simultaneous Caller/callee Hangup; Hangup Extensions Execute Only Once; Unable To Determine If Destination Channel UpReport
This is a question regarding whether there's any way within hangup extensions to determine whether the caller or callee leg (or both) of a bridged call has hung up. The test case I have is running under Asterisk 188.8.131.52, but the behaviour is observed in 184.108.40.206 (and also 220.127.116.11).
Within the dialplan, the Dial() application with the "F" flag, so that once the caller hangs up, the dialplan jumps to a new priority which enables the called party to enter some digits which describe the outcome of the call. Also, the "g" flag is used to attempt to continue execution of…
I have had a case where after a hangup on the Alsa CHANNEL asterisk still thinks the line or call is active.
rtptimeout` rtpholdtimeout` rtpkeepalive`
in my sip.conf file to help with this but it had no effect.
How can I ensure a session HANGS up and is not stale????
Is there a way for the next incoming call to VERIFY that console/ALSA channel is still valid. I dont want to hangup a real connection - I want to give a busy tone for sure.
But if the session is not valid I need it gone.
How can I do that. I am using 1.4.43
Is it possible to miss a UDP SIP packet to hangup a call?
Using 1.4.43 I had a call from on Asterisk box (server) to a low end client (chan_alsa) not hangup.
Could this be due to missed UDP SIP packet to hangup?
Is there anyway for a client asterisk (chan_alsa again) to monitor the connection and if the channel is not there to hangup?
I am trying to control the whole call using a FastAGI script. To that
effect I launch a FastAGI script (written with asterisk-java). Basically, I want to DIAL from within the FastAGI script. When the call
ends I want to control the hangup (if executed at the remote end), and
depending on the cause, dial again, play a message, or hang up. This is a
pretty standard telephony scenario. I did it before by executing the AGI,
setting variables, calling the DIAL command from the dialplan, and then
executing a second AGI…
I'm testing a few IAX trunk scenarios in a controlled lab. From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also plays tt-weasels as it's supposed to, but it's not the tt-weasels under its extension. It also dials the s extension. I only placed the s extension in the dial plan to verify that the traffic was going across the IAX trunk and hitting the correct context. Any help…