similar to: WG: Timeout when Dialing - HELP

Hi I have beronet BN8S0 isdn card in my asterisk and , card is working fine, but when I try to dial to special number 118913 ( telephone number information) from polish telecom TPSA, I always geting timeout . Bellow is isdn signaling dump : --> * CallGrp: PickupGrp: --> rxgain:0 txgain:0 --> * dad:118913 tech:mISDN/2-u25 ctx:default --> * Setting Context to from-tpnet -->

Hello, I am experiencing weird problem in Asterisk 16.2, possibly a bug. Same thing works fine in Asterisk 11. Here is the situation: I have 2 extensions on 2 phones. 4 extensions in total. phone 1: 8882 8382 phone 2: 8884 8384 And I have 2 SIP trunks for outgoing calls. I want to call via SIP1 when called via 8882 or 8884, and SIP2 when called via 8382 or 8384. And one last detail. SIP1

Dear Sir, I need to configure my Voice Mail on asterisk...I made the following configuration: * extensions.conf:* exten => _999.,1,VoiceMail(${EXTEN}) exten => _999.,2,HangUp() If the customer dial 9991234 then a prompt message should ask him to enter his voice message and this what is not happening *voicemail.conf:* [a2billing] 999123456 => 123456, 123456, michofr at mm.com The

For several months I have had frequent lockups using 2.4.x kernels. They began with about 2.4.10 and have continued down to 2.4.17. They always occur when I am online, downloading. The moden stops, the hard disk light is on, and no key works. They never happen with 2.2.20 or with earlier (i.e. pre-ext3) 2.4 kernels, and they never occur on my laptop with any kernel, including those using ext3.

Full_Name: Dan Freak Bolser Version: Version 2.0.0 (2004-10-04) OS: Linux 2.4.20-31.9 (RedHat 9) Submission from: (NULL) (62.253.128.15) The same problem I am having has been reported here

Estimados, Les consulto por el siguiente problema. Variable respuesta: Número de picadas de alimentación de los insectos (picadas) Factores: 1) Especie de insecto (especie), 2) Estado de desarrollo de los insectos (edad) y 3) Estado de desarrollo de frutos (estado). Modelo: MLG con distribución binomial negativa como componente aleatorio. Estrategia de selección de modelo: Evaluación del cambio

I have a Cisco 7970 that I am tying to get working. The phone powers up and registers but nothing else. I am using the development build. Any thoughts? The message I get when the phone powers on and registers. *CLI> skinny debug Skinny Debugging Enabled -- Starting Skinny session from 192.168.101.6 Recieved AlarmMessage Device SEP000F23FCA366 is attempting to register --

The same problem I am having has been reported here http://tolstoy.newcastle.edu.au/R/devel/04a/0344.html Namely that using par(mfg=...) with a postscript (eps) for inclusion with latex makes the figure appear upside down and back to front (flipped)! Converting the dvi to ps makes matters worse (the eps seems to be broken), however, it appears fine with gv. Here is (basically) the code I am

The same problem I am having has been reported here http://tolstoy.newcastle.edu.au/R/devel/04a/0344.html Namely that using par(mfg=...) with a postscript (eps) for inclusion with latex makes the figure appear upside down and back to front (flipped)! Converting the dvi to ps makes matters worse (the eps seems to be broken), however, it appears fine with gv. Here is (basically) the code I am

hi, i've a wirded problem, i try to dial out, using this dialplan [default] exten => _*7.,1,Macro(anrufextern-sip,${EXTEN:2}) [macro-anrufextern-sip] exten => s,1,SetCallerID(SIP-ID) exten => s,n,Dial(SIP/${ARG1}sip-out) exten => s,n,Hangup() when i use my analog telephone, everything is okay: - Starting simple switch on 'Zap/3-1' -- Executing

I'm running a setup with chan_misdn on a austrian PTP-line. When somebody dials in without an extension, he gets a busy signal. I don't see the call at all in asterisk. I *have* set immediate=yes in misdn.conf. And I *do* have an s-extension in my dialplan for the context used by misdn. Calls that provide an extension work fine. Attached is my misdn.conf and a verbose 3, misdn set debug

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. If 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

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

In Asterisk 1.8, I used the following line in extensions.conf to allow me to pass "*82" in front of a dialed number, to disable the callerid block that's normally on that POTS line: ; disable callerid block exten => _*82.,1,Dial(${POTS}/${EXTEN}) But this seems to have stopped working when I upgraded to Asterisk 11.7. I get the following debug output, with a "no

Hi, I need help finding why my system is dropping calls.. I enabled debugging on my box in the hope it would lead me to the answer as to why my system is dropping calls but unfortunately nothing is jumping out at me.. I have attached the portion of the messages file for two calls that were dropped.. (numbers names and ip's have been changes to protect the innocent) Can someone give me a

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. If 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

Greetings everyone, I'm having some issues connecting a Asterisk box to a old ISDN PBX. Everything works fine but the undetermined digits rules. For instance, if I have _00X. and I start dialing for instance 0035..., Asterisk just get the 4 first numbers and starts dialing 0035. I've tried adding: stop_tone_after_first_digit = yes append_digits2exten = yes Although I

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an

Hi all Does Digium make a quad BRI card? I can't see anything of the sort on their page but I thought they might call it something else in the States. Failing that, can anyone recommend a make/model that would handle 4 BRI ports? -- Cheers Wayne

Hi ! I have a single host which is connected via an 512kbit ADSL link to the n= et. I quite often use some realaudio application that uses some bandwidth = and I want to avoid other traffic to scale that bandwidth down. So I used = some classes (with htb) on imq0 device and applied some filters and it see= ms to work in most cases. However, there are some sites with which downloa= d is