similar to: How to send hangup command to call in progress.

Displaying 20 results from an estimated 40000 matches similar to: "How to send hangup command to call in progress."

2009 Mar 25
1
help - How to send hangup command to call in progress.
Hi, I want to send hangup command to the call which was logged in earlier via cli. Lets say to '5aec0e7207b24c8e1bdb511a460f7368 at callcentric.com Basically I want to hang up the call when ever I want but from the script. Thanks, Singh -------------- next part -------------- An HTML attachment was scrubbed... URL:
2008 Nov 18
1
setting up callback
Greetings Asterisk users! I'm trying to setup Asterisk system to act as a callback system together with callcentric (http://callcentric.com) but it appears that I hit common DTMF issue and I want to workaround this problem. Basically my current setup is the following: 1) I have dedicated Asterisk server that it is linked to my callcentric account 2) I have US phone number (DID) from
2014 Apr 14
1
how to configure callcentric peer
On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:
2005 Feb 17
4
[Bug 984] Hangup to script while forced command ...
http://bugzilla.mindrot.org/show_bug.cgi?id=984 Summary: Hangup to script while forced command ... Product: Portable OpenSSH Version: 3.9p1 Platform: All OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: sshd AssignedTo: openssh-bugs at mindrot.org ReportedBy: webmyster
2009 May 12
2
Hangup()-command does not hang up the line
When I call my Asterisk-server from my cell phone on one of the PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card, and in the dialplan the end of a context is reached and Asterisk needs to execute the Hangup()-command, I notice the following : - Asterisk tells me that the conversation was hung up (the log files tell me the command was executed) - On my cell phone I hear
2019 Feb 28
3
Asterisk - can't hear other side. Or other side does not hear us
Antony, It is correct. Noone connects to Asterisk box/server from outside.Callcentric SIP trunk configured and Asterisk maintains connection to it itself. No special ports opened, nothing. Connection happens from us to Callcentric and all calls routed in from CallcentricI don't know exactly how it's doing it by it works. Again, keep in mind it is working for many years for most / 90+% of
2017 Dec 14
4
SIP trunks going to the wrong context
Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes
2008 Jul 28
2
Callcentric Issues
Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf
2009 Jan 17
1
Sip Trunk registration
Hi Can anybody help me on this ? I am using Asterisknow 1.5.0-Beta(Freepbx) I am having a problem getting the sip trunks to register. It makes no different which provider one is using. Trunk name: callcentric Peer Details: context=from-pstn fromdomain=callcentric.com fromuser=1777xxxxxxx host=callcentric.com insecure=very secret=pasword type=peer username=1777xxxxxxx Register String:
2009 Feb 24
1
API hangup command
If a call is established to a destination device.... Phone to other device (phone or something). Then I issue the "monitor" command to record the person speaking. Now I want to STOP the end device call --- BUT ---- I want to continue to record the person speaking and sometime later deliver the entire message - is this possible???? Looking at the hangup command its going to shutdown
2007 Aug 13
1
Can't HANGUP call or channel on 1.4.9
I've isolated this problem the furthest that I can, and I'm now convinced this is a bug in asterisk. I have a context in extensions.conf like so: [my_context] exten => _X.,1,AGI(my_agi|${EXTEN}|${CHANNEL}) exten => _X.,2,GOTO(my_other_context|${EXTEN}|1) exten => h,1,DeadAGI(my_agi_cleanup) For the purposes of this scenario, my_agi simply will try to HANGUP the channel to
2008 Jun 21
1
Fwd: Detection of Answer, hangup, busy etc while using Dial command
---------- Forwarded message ---------- From: Arun Kumar Chaudhary <uniquearun04 at gmail.com> Date: Sat, Jun 21, 2008 at 4:51 PM Subject: Detection of Answer, hangup,busy etc while using Dial command To: asterisk-users at lists.digium.com. Hi Guys, I am in kanpur, India. I am using Dial() command in my phpagi script. I am unable to detect whether it is connected to the dialed number, if
2009 May 06
0
astcc - outgoing call does not hangup properly
Hi, I am using ASTCC and trying to setup a calling card platform. The problem that I have is that astcc does not hangup calls correctly: 1. If I try to dial a number, call goes through fine. When I hang up the call from my side I get this: -- Called 192.168.1.56/1XX6872XXXX (masked a few digits) -- SIP/192.168.1.56-086c5000 is making progress passing it to
2003 Jun 03
2
Detect hangup on unanswered POTS call
I've been using * at home for a while now and I'm quite happy with how it works. Having voicemail emailed to me and notify my cell phone via SMS is a great way to impress my friends. :-) The inbound context for my X101P looks something like this: exten => s,1,Dial(SIP/analog1&SIP/analog2,20) exten => s,2,Answer exten => s,3,Voicemail(u1234) exten => s,4,Hangup The
2018 Oct 16
2
Is there any way to pass caller id to
Thanks all, I did contact Callcentric about it and their tech support helped meget those headers established. They even helped to troubleshoot Asterisk dialplan. A the end all works as it should. Thank you,Ivan Message: 2 Date: Mon, 15 Oct 2018 23:39:31 +0200 From: Daniel Tryba <daniel at tryba.nl> To: Asterisk Users Mailing List - Non-Commercial Discussion     <asterisk-users at
2005 Mar 13
2
sending a DTMF tone before hangup
OK here is a possible tricky one. I have a rocom door entry system which connects to an FXS port on my TDM400P card. When the door button is pressed it initiates an 's' extension which dials a number of SIP extensions. When a SIP phone is picked up the user can speak to the person at the door and press the 7 digit which sends at DTMF tone to the rocom unit opening the door. All this
2018 Nov 16
2
Queue not dialing out to cell phone for some reason
My settings for the queue.log are in the [general] section of logger.conf I'm running 13, I didn't see what version you said you were running. If I wanted to add a LOCAL channel to my queue I'd do it as member => LOCAL/7124 at kiniston-intern,0,John,hint:7124 at kiniston-intern On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch <idemkovitch at yahoo.com> wrote: > John,
2014 May 23
1
Way off topic: gvoice and callcentric
To deal with google dropping xmpp for voice, I've gotten a callcentric number. The cc number connects to asterisk, and all works fine. Then I set up the cc number as the gvoice forwarding number. If I'm on the gvoice site, I can make a call and it will ring my cc number and then the outside number. That also works fine. BUT, when an outside call comes into gvoice it forwards the call
2007 Aug 09
0
VOIP Provider- Callcentric
Asterisk Users, I am looking for Sip Providers for my Asterisk 1.2.13, running Debian Etch system with McLeodUSA's T1 service. Has anybody ever used Callcentric for their Sip Provider? Any service issues with Callcentric? Best Regards, John _________________________________________________________________ Messenger Caf? ? open for fun 24/7. Hot games, cool activities served daily.
2010 Aug 10
4
How to determine which party hangup the call? cause of Hang-up needed.‏
Hi Everyone Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell Canada. User claims that call hangup without any interferance to the phone set. Is there ANYWAY to find out which party hang-up the call or if the call was cut-off due to other reasons? I checked the *"asteriskcdrb"* table and it's pretty much useless in this case as it only logs the duration and