search for: offhook

> -----Original Message----- > From: Jacqueline Lee [mailto:jlee@isdomaininc.com] > Sent: Friday, September 23, 2005 11:46 AM > To: asterisk-users@lists.digium.com > Subject: channel offhook state > > > We are using a digium card (TDM400) with asterisk for our access to the > PSTN. Initially when the server starts, all the zap channels on the card > are in the "onhook" state. As soon as a channel is used (for inbound or > outbound PSTN calls) the correspon...

My TDM400P w/ 4 FXO cards seems to have trouble with onhook/offhook switching. It dials perfectly, but does not seem to be changing the onhook/offhook state appropriately. It changes sometimes, but it's not really reliable. For example: When I booted the machine, it started as onhook. It remained "onhook" through the entire first call (which...

Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! =======================...

...and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be unplugged from TDM card, the line would become normal again. The offhook in Zap Show Channels is only for FXS cards >Sorry, if it is a well known problem, but I did not find any specific information yet.. Please answer...

I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with what seems to be correct settings (according to digium and asterisk wiki). As soon as I plug in my POTS line into FXO mod the line goes into offhook state (whether I have power to the card or not). Should this happen? When I try to call * box all I get is busy signal. I've installed stable version, cvs version, change the phone cord from 2 wire to 4 wire. My config files are standard from compile and all i've added is info from digiu...

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

Hello, Caller -> PRI -> SIP -> application. If application goes offhook right away, dtmf/audio works fine in both directions. If application, before going offhook (sending OK) plays a message and wants dtmf/voice from the caller, then caller can hear this message but his dtmf/voice don't reach application. Any way to configure it? Thank you. Alex Zarubin ----...

...; Threeway: <None> Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x0000 Actual Confmute: No Actual Hookstate: Offhook It seems to remain in this state until we reboot the box. This is our zapata.conf for the context in question: [channels] language=en context=from-analog-incoming signalling=fxs_ks usecallerid=yes callprogress=no busydetect=yes transfer=yes echocancel=yes echocancelwhenbridged=yes channel =>...

...my dial plan. I am calling from SIP extension 1100 and calling 770-555-1234, which is a local 10-digit phone number. -- Executing Dial("SIP/1100-9adc", "ZAP/g0/7705551234") in new stack -- Called g0/7705551234 -- Zap/4-1 answered SIP/1100-9adc The status of the channel is "Offhook" regardless of whether or not the phone line is actually Offhook or completely idle. I'm assuming that when the line seems idle, it should show as Onhook. ------------------------------------ asterisk2*CLI> zap show channel 4 ------------------------------------ Channel: 4CLI> File...

I have an interesting problem. I am attempting to install a payphone utilizing a Cisco ATA-188. The payphone actually works, but there are some timing issues. What happens is you pick up the payphone and the ATA grabs a line and goes offhook. While you monkey with putting money in and dialing the number, you are eating up the time before you get the offhook reorder tones (or howler tones I think). If you can put the money in and dial real fast, it works!! I have been screwing with the ATA configs for days now and can't come u...

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a predefined extension when it goes offhook?...

Hi, I have to develop a phone application using asterisk's chan_oss. When the phone is idle, i.e. the last command was a hangup, one hears a "toot, toot, toot, ..." But unforuntaly its use is in Germany, where one expects a continous "toooooooooooooooooooooooooooooooooo ..." before dialing. Is there anything to define the tone indicating "ready to dial"?

Hi, I'm having an interesting problem with my card. It seems to work fine, for the most part. When I first load the module and asterisk, it detects the line in the on-hook state. However, after the first phone call, zap show channel 1 lists it as being off-hook. During subsequent calls, the card is listed as being on-hook, and when it's not used -- off-hook. There are also some weird

...is the script I am using to make the call, and start recording the channel. I am starting to make a system the will use asterisk to become an automatic random quality monitoring system that will dial into an Aspect (or any for that matter) ACD that has CTI and do it's job. The "ZapDialOffhook" event says it's successful in the API window but does not seem to do any dialing (i hear no tones). Is there another command that works? I have tried using the CLI command SendDTMF to no avail. Everything below works except the -zapdialoffhook-. Thanks in advance for any assistance...

Does anybody know if there are any external telephone-keypads for sale anywhere? (containing the keys 0-9, *, # and onhook/offhook would do) I am looking for a keypad to control a softphone and would prefer the controls to be in the physical world instead of as a window. Sincerely, Markus Hakansson

...Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no1* Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x0000 Actual Confmute: No Hookstate (FXS only): Offhook When cable plugged in: [Oct 20 09:02:56] DEBUG[2359] chan_zap.c: Monitor doohicky got event No more alarm on channel 4 [Oct 20 09:02:56] NOTICE[2359] chan_zap.c: Alarm cleared on channel 4 When cable unplugged: [Oct 20 09:04:55] DEBUG[2359] chan_zap.c: Monitor doohicky got event Alarm on channel...

I am searching for a way to add a 2 second delay before calling out with Dial(). Sometimes I get the message "you must first dial a 1 to place this call". I presume the phone company is missing the first digit pulsed out sometimes. How do I put a 2 second delay after coming offhook and before dialing the digits? Thanks, jerry

...e Timeout bebop*CLMailboxCount Check Mailbox Message Count bebop*CLMonitor Monitor a channel bebop*CLStopMonitor Stop monitoring a channel bebop*CLChangeMonitor Change monitoring filename of a channel bebop*CLZapTransfer Transfer Zap Channel bebop*CLZapHangup Hangup Zap Channel bebop*CLZapDialOffhook Dial over Zap channel while offhook bebop*CLIAXpeers List IAX Peers bebop*CLIAXpeers List IAX Peers bebop*CLQueues Queues bebop*CLQueueStatus Queue Status bebop*CLI> show manager commands bebop*CLPing Ping bebop*CLLogoff Logoff Manager bebop*CLHangup Hangup Channel bebop*CLStatus Status...

I have made bri-stuff.0.0.2rc19 to work (I think) but I can not achieve any in-dialing nor I can dial out; this is what I have from "pri intense debug span 1" command ---------- *CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 -- Executing Playback("SIP/201-a862", "tt-weasels") in new stack -- Playing 'tt-weasels' (language

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come over the other lines. -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director