similar to: Hangup Chanel when a peer unregisters

I have a remote user with Eyebeam on a laptop. Internet connectivity seems good, there is no packet loss to that location from the PBX. Everytime the user starts eyebeam, the application tries to register. Asterisk accepts the registration but the reply never gets to the client application, so it thinks it has not been accepted and times out. Then Asterisk unregisters the extension. --

We have had 20 calls over the last month where the SIP channel has not identified that the person on the receiving end has hung up. Is there a way of fixing this ? TIA Julian

Good day all! I have an issue which has plagued me for quite sometime now...and as I close in on its cause, I have reached a point where additional info would be greatly helpful! When a SIP device dials another SIP device...Asterisk connects the calls and displays the channel information. If one of those SIP devices hangs up, Asterisk receives the hangup notice and disconnects the call/channel.

I'm trying to make H.323 trunk between AVAYA&Asterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob.

I have a SIP phone that is registered. i can make calls out from the phone. I can't make calls to the phone. What does the error message mean? How can I fix it? Thanks! 8 headers, 0 lines Destroying call '6b9fb03c4677b9266e1263fb0c7ea304@127.0.0.1' Jul 15 22:10:49 NOTICE[262159]: rtp.c:285 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible == CDR updated

Hi I am using inAccess channel driver. Compiled, installed. This is what I get when I am trying to start * --------------------------------------- [chan_oh323.so]WARNING[16384]: File loader.c, Line 226 (ast_load_resource): libh323_linux_x86_r.so.1.12: cannot open shared object file: No such file or directory WARNING[16384]: File loader.c, Line 394 (load_modules): Loading module chan_oh323.so

Response below... On Fri, Dec 27, 2019 at 12:02 PM David P <davidswalkabout at gmail.com> wrote: > > > > > I'm looking for a way of detecting in my dialplan when a peer becomes > > non-responsive after answering. [deleted] Is there a way to configure > > a handler for this state? > > > > We use v14.7.6 and we dial the peer this way: > > >

I setup my aah box with a sip trunk at irisxa.iristel.net Incaming it is ok but when I try to dial 8 and the nr where I want to call I get all line is busy. In my log I have these: Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command' Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command' Feb 22 14:33:19 VERBOSE[2721] logger.c: --

Hi,

Looking at the pastebin, the Vega device sends a CANCEL with reason: Reason: Q.850 ;cause=16. Cause 16 is normal clearing and suggests that the original caller has disconnected. I would take a look at the Vega's logs Regards, Steve On Thu, 5 Mar 2015 at 11:41 ricky gutierrez <xserverlinux at gmail.com> wrote: > > > On Wednesday, March 4, 2015, ricky gutierrez

Hi all, I have H.323 Gateway, and i want to make a peer to route calls to this GW. But i don't know is oh323.conf supporting to add peer type entry with all feature. Please let me know how i can add H.323 GW type peer? -------- Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: abdulzu@hotmail.com GoogleTalk: lateef.np@gmail.com YM!: abdul_zu Doha

Dears; I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides). My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial the

Hi; How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how? Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it). What is happening with me that from time to time, I find some DAHDI channels are stayed connected

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1

Hello group, I put together a simple PHP based conferencing manager for Asterisk 11.3 I used ODBC MYSQL for conference IDs and PINs. All this is working as desired but I would love to add an active conferences display to the front end. It seems to me that AMI is the way to go but I have no idea how to accomplish this or even where to begin. Any guidance is appreciated. Pat...

Hi all.. I have a problem with my asterisk installation. I'm using a Wilcard X100P clone in Spain. Incoming calls work fine, but when I make a outgoing call, a hear the ringing, and the peer phone ring, when the peer answer, asterisk hangup the call, or say busy. This is my conf: zaptel.conf: --------- loadzone = es defaultzone=es fxsks=1 zapata.conf ---------- [channels]

Hi, Anyone here using sipgate.de ? It worked for months, but for a couple of days now I'm unable to register with them. My account is ok, because I can login to the website. Asterisk keeps showing me: Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n) I looked at the sip debug stuff, and all I

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk

Does the System() dialplan application have a limit on how long it can run? Either a time limit, or server load limit? I'm trying to pipe the output of Sphinx2 into Text2Wave, but Asterisk just runs by it to the next extension priority, with no errors. If I run the same command via the system shell, all is good, though it does take a few seconds, probably about 5 seconds to run. Yes,