similar to: Detecting which party initiates a hangup

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

Hi guys, I spent a couple of hours in Google, but the problem appears to be uncommon, so I'd like to ask about it here. The problem is exactly the opposite to "Asterisk does not detect FXO hangup". In my case it's the Telco who does not appear to be detecting Asterisk's hangups. Telco is Telus in Vancouver, Canada. The setup is very simple - Telco -> FXO/TDM400p

Hi Everyone, I am trying to find the issue of dropped calls in the middle of the conversation. The system is Elastix. Anyway to know which party hangup the channel in case of Asterisk 1.4 and Sangoma analogue cards? (this is not PRI) Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've got an old analog PBX and I'm trying to connect an FXO port on my asterisk server to one of the extensions on the old PBX. This should work as en extension on the old PBX should be providing dialtone, battery current and ring voltage. However, when the old PBX "hangs up" asterisk doesn't appear to be detecting the hangup (the DAHDI channel stays in use). Can anyone

I think the problem it's in your dialplan, extensions.conf: ; voicemail management [voicemail] include => misc exten => 6245,1,VoiceMailMain2() exten => 6245,2,Hangup() Check the last line, I have the same problem and was because I wrote wrong the Hangup instruction... Regards! Date: Thu, 10 Jun 2004 09:54:32 -0600 From: Chris Hirsch <chris@base2technology.com> To:

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

Hi, Odd problem have just noticed in that when I call into the PBX DAHDI detects the call and hands it off to the extension, if I then hang up it still continues to process through the dialplan. This is what I have in chan_dahdi.conf: [channels] language=en echocancel=yes usecallerid=yes cidsignalling=v23 sendcalleridafter = 2 hanguponpolarityswitch=yes rxgain=2.0 txgain=3.0 progzone=uk

Hello people, this is my first mail to this list. I'm new to asterisk and trying to set up an IVR. So far my dialplan works nice connecting with softphones, but I'm having problems to detect hangups on the analog line. Here are the details: -Clone pc with Ubuntu server 10.04 64 bits -Asterisk 1.6 -Openvox A400P, lspci -v says: 01:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX

Hello, I've got a test setup with 2 asterisk boxes: Asterisk1 with: asterisk 11.5.1 dahdi 2.7.0.1 Digium TDM400 with 2 FXO ports Asterisk2 with: asterisk 11.5.1 dahdi 2.7.0 Digium TDM400 with 2 FXS ports Asterisk1 has the following AEL Dialplan: context remote { s => { Answer(); Dial(DAHDI/g1/7005); }; }; When a call from Asterisk2 comes in, it is correctly

I've searched the Wiki and this forum with little success. I have a TDM400P in my server which functions fine. Except it will continue ringing about 3 times after hangup. I.e. it's failing to detect the hangup tone. I was previously running a Sipura 3000 and had the same issue. After researching and some timely assistance I was able to determine the hangup tones applicable to

Hi, Im working with the following versions: -asterisk-1.2.12.1 -zaptel-1.2.9.1 And with the following card: 00:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 201 I/O ports at c800 [size=256] Memory at fe000000 (32-bit, non-prefetchable)

Hi everybody... I'm having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody

Can anyone suggest where I might begin looking for an answer please? I have just installed a TDM400P (2x FXS and 1x FXO modules installed) The first problem is that it does not seem to be able to detect if the remote party has hung up when a call comes through on the FXO. For example, if someone calls in, and then hangs up at any time after it starts ringing, Asterisk carries on as though the

Canuck15, No, I hadn't played with the gains. But I've now done so and no difference unfortunately. Thanks for the suggestion though. I have discovered that after Asterisk has answered the call and the remote caller has hung up, if I lift the receiver on a phone connected to the line (in parallel with Asterisk), Asterisk then DOES instantly hang up. Would it be reasonable to assume the

I suppose this is rather an informative e-mail than a question. However if people had similar experiences or could comment what the differences are in other countries or with business analog lines, it would be interesting. It took me a week until a BT engineer was sent to my home home, since BT tech support was unable to provide information about the problem. Problem: Calling party controls how

Dear all, When I make a call using my PRI line, all goes well, but suddently the call hangs up. I searched the asterisk logs, and I found that. Write to 55 failed: Unknown error 500 Short write: 0/15 (Unknown error 500) What does this mean? Why this occurs? How could I solve that? Someone could tell me if it was a primary error (the primary shows red alert in all its channels) or it could be a

I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and checked the conf files but I can't see anything that I've missed. Cheers Matt

I have set up an application which records incoming calls (customer problems), and then places outgoing calls (handing the problems to support people). I have written some AGI programs (in C) to implement parts of the functionality. So far I am very pleased with the overall application. But I have one very annoying, very nasty problem: The C AGI is handling a dialog with a user on an outgoing

Hi, I'm trying to set up a call-back system using auto-dialout files. I want the call to be terminated when a specific timeout (defined in the .call file) is detected. Both parties should then be hangup. The problem is that the timeout is never detected... How to solve this? Thank you, Pierre .call file ---------- Channel: IAX2/:@xxx.xxx.xxx.xxx/0111111111 Callerid: 111111111

Hi List, I would like to know if there is a way to detect gateways which time out (because of network problems or hardware failure for instance) when you send traffic to them. So when you do: Dial(SIP/number@gateway) If a call couldn't get through because the gateway has timed out, i want to do something about it. The idea would be to suspend gateway which time out for 60 minutes,