similar to: Hangup after 1 second of ringing ?

Hello, I'm having a bit of trouble getting asterisk + tdm400p + rhino channel bank set up just right. The problem is a single ring on a telephone connected to the channel bank after the line is hung up. I've seen a couple messages similar to this, but I haven't seen any resolutions to the problem. Using the tdm400p card, with the first port connected to a 24-port fxs rhino channel

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

I have 3 phones, SIP/3, SIP/6 and SIP/9 SIP/3 subscribes on hint on SIP/9 Phone 6 calls phone 9, blf on phone 3 flashes until 9 picks up, then it is steady red. That's correct. But when 9 hangs up the hint goes to "InUse&Ringing", the light on 3 is still flashing. It keeps flashing until somebody calls 9 and hangs up again. -- Executing [9 at local:1]

This is a new post, but its really a three-time retread. I hope someone has a clue on this, as it could be helpful in many circumstances: I am looking for a way to dial 'special' an extension (in house, like 102), which are all Polycom IP. I'd like to ring the extension as normal, but have the option of, while the line is ringing, to press a digit, hop out to a new context and/or

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi!. I am having problems with getting asterisk to detect when someone hangs up. I have a TDM400P with one FXO module connected to my telco, and also a FXS-module connected to my phone. The FXS-module detects hangups just fine, but I can't get the FXO to detect them. I am pretty sure i have disconnect supervision on my phoneline since when I

I hope somebody has seen this before... I'm trying to use a Dial command on a inbound call to ring multiple destinations. The calls come in to me from the provider on IAX2, and one of the destinations I try to ring is a IAX2 to call to my cell phone. When I add the IAX2 destination into the Dial command, the setup I am trying to achieve works (i.e. my Zap, SIP, and cell phone all ring) but

Hi, I have simulated ?Chan phone? driver according to my own driver code and I am able to make internal and external [trunk] Asterisk calls. Only issue I am facing is with hangup in ringing state of incoming call. (1) Make a call from external X-lite to FXS and FXS is in ringing state now (2) Disconnect the caller [X-lite] (3) X-lite sending cancel message to asterisk but hangup

I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the

I ran into a couple of odd problems with a new Asterisk box I have set up... I have a T100P connected to a channelized T-1 (24ch) to the PSTN. The T-1 is set up properly so far as I know and seems to work perfectly with one exception: When a call comes in, asterisk starts a 'simple switch' for it, goes through my 's' extension progression until it gets to the first Playback

Dear All I have this setup [asterisk]----[mediant2000]-------E1 Trunk----------[Avaya] When i call from avaya to asterisk i got long ringing tone then hangup but when i call from asterisk to avaya i got 4 ringback and then hangup with this error *CLI> Jun 26 01:26:08 NOTICE[5555]: chan_local.c:523 local_alloc: No such extension/context 1022 at mysip

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. When I pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if I dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

Hi: I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with kewlstart signalling. When an outside caller calls the server, the TDM-400 goes off-hook and provides a ringing tone to the caller. If the caller hangs up before the receiving party answers the phone, Asterisk fails to detect the hang-up. The TDM-400 stays off-hook, hogging the line, while Asterisk rings the

Hi All, I would like to know if you can confirm that, if using origination via AMI, as documented here: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate it is not possible to set the max duration of a call. I mean: what you would do with the L (limit) parameter of the command Dial, is not possible when originating. As well as using the absolute timeout, as documented

I've been using the following macro to ring SIP and IAX devices for a few seconds, and then add on a cell phone if there is no answer on the SIP or IAX device. Periodic problems began a few versions ago and now the problem happens every time with 1.2.9 and 1.2.9.1. The problem is that when a call from the PRI falls through to voicemail, the call is dropped before the voicemail greeting

Hello, on previous versions of asterisk, extension h and H make us know who ended a call (caller or callee). In the last * versions, seems that only h extension is used, as stated here http://www.voip-info.org/wiki/view/Asterisk+standard+extensions In the last versions, how do we know which end terminate a call (SIP, ISDN, Analog, ...) in h extension ? Will the

Hi, I have 1.2.9.1 installed. It always rings first available agents for 15 seconds, then rings and hangs up the next agents straight away, then ring the next agents for 15 seconds. It goes as a loop. Any one has the following same problem? Thanks. Agents.conf [general] persistentagents=yes [agents] autologoff=60 wrapuptime=15000 ackcall=no group=1 agent => 7130,7130,agent1 agent =>

hi, Just wanted to inform everyone, if you're using the latest bristuff's you might (depends on the country!) have hangup issues. The issue appears every time you dial an external number, and hangup after letting it ring for a few times. Then the remote party keeps ringing. In some situations (we only encountered this while dialing to other * servers) it keeps the line open on the

Anyone experience ring oddities with extensions.conf rollovers? Let me summarize... One of my extensions.conf file is built to ring during the day, ring/go to voicemail after a certain time: [main-aa] exten => s,1,GotoIfTime(17:00-8:30|mon-fri|*|*|*?main-night-aa,s,1) exten => s,2,GotoIfTime(*|sat-sun|*|*|*?main-night-aa,s,1) exten => s,3,Dial(SIP/201,25,tr) exten =>

Hi, I am new to this group.I searched for my problem in the forum but could not find any solution. So here it goes: In my work place we have an asterisk box. Everything works fine except the fact that when I first call the work phone number from my cell the auto-attendend works fine but If I hang-up and call the same number again, the call keeps ringing instead of auto-attendent answering. I also

Hi, For some time now I have noticed that our RBS T1 (asterisk 1.4.35, Dahdi 2.3.0+2.3.0, TE410P) often has channels stuck in the state "Ringing", like this poor chap who got stuck on two calls in a row, apparently: [excerpt from "core show channels"] SIP/7157997-0000534b 7760308 at business:1 Ring Dial(Dahdi/g0/7760308) DAHDI/3-1 5130262 at from-pstn:1