similar to: Zap Analog Line Hangup Problem

Hi... I have the next problem.. I have a FXO card with i can make calls but i cant recive calls. At the consol, i get the next error: -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook in strange state 6 on channel

Hello list, I have recently deployed Asterisk as the phone system for my office. So far, everything has been going really well, except for one little thorn in my side. I have a set of 6 analog lines that are connected to a TE411P via a Rhino FXO Channel bank. If I call the analog number, Asterisk answers the call, and routes it appropriately. The problem is, when I hangup, Asterisk never

hi all (again). i have this problem. when a people call to meetme and join a conference when this people leave and hangup your phone asterisk can't detect the hangup. all people use analog lines to connect the meetme is any way to tell asterisk to hook when these people leave? -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [s at from-pstn:3] Hangup("Zap/5-1", "") in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17

I have the following dialplan. Everything seems good except for one thing. If the background message is playing and the user hangs up and does not press a digit how do I run an agi on that event. I tried an exten => h,1,agi(smvoice,-digium_failed) but that was never called. I am using 1.4.10 thanks, Jerry --------------------------- [smvoice-analog] exten => s,1,Wait(1) exten =>

I'm having many troubles with x101p (orginal from Digium, wcfxo kernel module) on analog line simulated by a Cisco Router I'm experiency random hung up while zaptel doesn't recognize call progress (Italy signalling) Signalling is simple ignored as if someone hangs up on the other end of the analog line, on the asterisk end is it possible to listen the busy tone as generated from

I have a customer that is complaining that any call coming in from Nextel gives a fast busy. We are running Asterisk 1.4.7.1 with Zaptel 1.4.3 and all the MFC/R2 patches and libraries. All other calls go out and come in, just Nextel seems to have this problem. The phone company technician connected a PBX emulator on the line and that one could receive the calls from Nextel. The E1 is provided

And if that ain't confusing I don't know what would be. I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago and ended up never using it. Passed it along to a friend who is having some problems with it. (He isn't on this list.) We've both tried searches using Google but haven't been able to find anything that helps. So this is more a question of

I have had to install a TDM800 in a site, as the telco has held off installing ISDN indefinitely.. It's all fine except for the fact that it takes ages to hang up the line (6 or more rings), and sometimes doesn't even bother. This is only on incoming calls - outgoing calls work perfectly. Is there any good tricks for a fast and accurate hangup detect in this situation?

Hi list: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. THanks. In the example i'm dialing from extension SIP/112 My DialPlan Secction: [macro-callonlyiffree] exten => s,1,ChanIsAvail(${ARG1}|s) exten => s,n,NoOp(${AVAILCHAN}) exten

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Hi, I have an FXO card connected to my phone line which works in Asterisk as Zap/1. Is there any way of detecting whether something else is on the line before picking up on this channel? For example, I dont want to pick up and dial out on the line if someone is on it using another phone (which is connected directly to the line, rather than through Asterisk). Also, when an incoming call comes

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

Hi, We've got IAX softphones, GrandStream VOIP phones and zaptel connected analog phones. Caller id, internally, works just fine (as long as I use numeric only callerids) for IAX and grandstream. Is there a way to have the analog phones' LCD display show the caller id? These are plain old regular analog phone, that if I had callerid from my telco would show on the screen. thanks

Is there any way to fill in the DID information for analog Zap channels? I've got a TDM422P and since I know the phone numbers associated with each of the 2 FXO channels I'd like to set that so that future extensions contexts can use it and the caller-id info in the form mydidnum/callerid like I can with VOIP DIDs. I haven't found if there is a variable that can be set with this info.

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We have 4 analog line and 2 analog trunks. On the trunks we have all the DIDs coming into the current phone system. Trying to get everything moved over to Asterisk but having issues picking up the calls on the analog trunk. We can receive calls on the plain analog lines and we can call out on all analog lines and analog trunks. When a call comes in on the trunk line the ZAP channels don't

I've tried this before, with no luck. I've got to try again this evening, and I'm looking for some help. Here's my configuration -- pretty simple, really. Asterisk box -> T100P -> TA750(20FXS/4FXO) -> phones and outgoing lines I have an analog modem (Ok, it's a TIVO) that I need to be able to dial out. Right now, I have the modem connected directly to an outgoing

Hi all, i have noticed that i can't hear a ring tone if i make a call from my TDM40B to a chan_modem_i4l endpoint. I had a look in the code from chan_modem_i4l and there is a function calling "i4l_handle_escape" that gives a AST_CONTROL_RINGING frame back. But this seems not work ...(or i4l is not signaling it ?) Til now i have used the Dail app like Dial, Zap/g1:XXXXXX|60|r so it