similar to: debugging zap channel

Hello, Can anyone help with the output shown below? It?s running on RH9, recent CVS of Asterisk and with one X100P card (2 channels), a budget tone 102 and Xlite softphone. CLI> -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:24 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no

Hi there Background: I want to add DDI and voicemail to users on an existing analogue pabx.. It does not support ISDN. I have 10 DDI numbers via IAX which I am having sent to my Asterisk box. I have 2 X100P cards connected to 2 analogue extension ports of my main legacy analogue pabx. I have set up voicemail for each of my DDI numbers, and when a call comes in for the person at pabx

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Hi everyone, our setup includes a NEC PBX connected to our asterisk via bri lines. The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door. So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so. Setup: asterisk

I try to use a SIP trunk from a VOIP provider to make land to mobile calls. If I do these from a ZAP channel, using an analogue phone, after few seconds of silence (I don't like to generate fake [r]inging) I ear the ringing tone from the mobile operator along with any message the mobile operator decide to say me. If I try to use a SIP phone (or a IAX phone) attached to my asterisk box, I

Hi Guys, Currently, I have a Sangoma A400 installed with 20 ZAP PSTN analogue lines. The first line is giving me problems due to rain (probably coroded line). My server using FreePBX dials out with g0 (group 0 which includes all 20 lines) and it happens that the bad line is the very first line. Can I simply put ; in zapata.conf like this to seclude the first zap line from getting calls in or

Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GS102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels and the first

Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GrandStream102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels they end

Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks.

Hi there, I have a x100p card in an asterisk box. Does anyone know if it's possible to do a hook flash / recall on an active zap channel? On what I'm trying to do... >From an ordinary analogue pstn telephone I can call someone, press recall, call someone else, press recall 3 & presto we're on a three way chat, with me only using one line - using the telephone company's

Hi. I have the following line in the default context of all my internal extensions: exten => 9876,1,Transfer(125) When I dial extension 9876 from any sip phone, * dutifully transferrs it to extension 125, which is just what I want. Unfortunately when I dial 9786 from my Zap connected analogue phone, the transfer doesn't go through and the dialplan drops through to a hangup. debug

Hi List, I'm having a bit of an odd problem with asterisk and outgoing zap calls. Tzafrir has been kind enough to help me get the logging sorted out so I have some idea of what's going wrong, but I'm a little flummoxed. Essentially the symptoms are as follows; Make a SIP call from Cisco 7960 or 7940 to asterisk, where it is routed out on a ZAP (x100p) line. After

I am installing one asterisk, to establish connection with my PABX Siemens, in ISDN, link went up normally, also I obtain to internally call the branches the PABX, normally, but when I try to dial for the PSTN, through pabx with the command exten = _ 19xxxxxxxx, 1, dial(zap/g2/${EXTEN}, 30) asterisk, reports me the following error: -- Executing Dial("SIP/8110-a729",

Hi, I have a working Asterisk 1.2 server on kernel 2.6.22 with the OSLEC echo canceller on a Digium PRI card. I recently switched to kernel 2.6.23 with the MG2 echo canceller (nothing else changed). Each time I try to establish a call on the PRI line I get a congestion signal. in /var/log/asterisk/full: Feb 20 08:09:43 VERBOSE[10657] logger.c: -- Executing

Hi all, I've been trying to wrap my mind around this one for several days now. How can I 'debug' the CallerID reception on a Zap/POTS channel? I have a POTS line with CallerID and a Digium TDM11B card right now. I have my signalling set to ks for both sides, can make and receive calls just fine. But I never get CallerID on incoming calls. I get the following messages: Aug 11

TDMoE was it. Thank you!!!! Wiley ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Goodyear Sent: Friday, June 24, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Exposing Zap Channels on Server A to be UsedByServer B

I was looking at the sample "follow me" config (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me <http://www.voip-info.org/wiki-Asterisk+Tips+follow+me> ) which uses a dial modifier 'c' to enable Answer confirmation - "If the letter c follows, then "Answer Confirmation" is requested, in which the call is not considered answered until the called user

Ok here is one for you. I know we all do the this for 911: exten => _911,1,Dial(Zap/1/911) exten => _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a "least cost" routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is

Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the channel, i cannot hear nothing. If I change the configuration and i send the call to an internal sip

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the