similar to: current CVS version

Hi, I have some trouble with the Monitor() application. I start and stop it via the management interface, giving no special parameters except the channel name. What happens is: - if I specify WAV as the format, the resulting files are exactly 44 bytes big and contain nothing at all - if I specify GSM as the format, the resulting files are of size 0. I did not request mixing of the files or

Hi, I have a big problem with EXEC in AGI scripts: I do, for example, "EXEC Dial SIP/phone1", Asterisk says -- AGI Script Executing Application: (dial) Options: (sip/phone1) Jan 10 14:33:20 WARNING[10567]: chan_sip.c:1389 create_addr: No such host: phone1 Jan 10 14:33:20 NOTICE[10567]: app_dial.c:743 dial_exec: Unable to create channel of type 'sip' I do "EXEC

Hi, I still have some trouble with the AGI interface: - I can use EXEC now, but it never gives me the error returned by the executed application, if an error occurs - I can use ANSWER, but I have to put something else behind "ANSWER". If I say "ANSWER", I get "510 Invalid or unknown command". If I say "ANSWER ''" or "ANSWER ." or

Hi, what happens if the dialplan contains something like exten => s,1,AGI(agi://10.0.0.1) exten => s,2,Dial(SIP/phone1|20|tr) etc. - if 10.0.0.1 isn't reachable or doesn't react on the connection? In my test cases, I always got a hangup and no further processing of the dialplan. Any hints? ( the call mustn't go into Nirvana if the AGI server isn't available!) Thanks

Hi everyone, I have a question concerning DNS SRV lookups. The situation is like this: - one central Asterisk server - many domains with SRV records, let's say we have bar.com and doe.com Now the question is: if the SRV lookup is done for foo@bar.com the call is mapped to foo@myasterisk.mydomain.net. Is that correct? If so, I have a problem: if somebody calls foo@bar.com, Asterisk

Hi, I'm having strange warning from asterisk when I try to dial GSM Gateway: -- Executing [1011501522xxx at sm:1] NoCDR("SIP/ibm-b2c52848", "") in new stack -- Executing [1011501522xxx at gsm:2] Dial("SIP/ibm-b2c52848", "Zap/R3/501522xxx") in new stack -- Requested transfer capability: 0x00 - SPEECH [Jul 13 11:58:50] WARNING[18208]:

Hey, whenever I try to load, I get these errors Jan 15 16:37:24 ERROR[7573]: chan_iax2.c:7486 load_module: Unable to bind to 0.0.0.0 port 4569: Address already in use Jan 15 16:37:24 WARNING[7573]: loader.c:345 ast_load_resource: chan_iax2.so: load_module failed, returning -1 == Manager unregistered action IAXpeers == Unregistered channel type 'IAX2' Jan 15 16:37:24 WARNING[7573]:

Hello. I have just arrived to Asterisk. I would like to know if Asterisk can perform some functionalities I am looking for. I want to allow voip over sip to some users. All of them must have their own user name and password to login to Asterisk so only allowed users can login. All calls started by users have to be redirected to one account at our voip provider. I think those functionalities can

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

Trying to compile asterisk (again) from scratch. I seem to be still experiencing the effects fro Jan 25 where I get no sip to sip audio. I have tried upgrading to 1.2.3 which has made no change in the problem. I am starting over and now trying to compile/install /trunk zaptel libpri asterisk following the instructions to grab the source trees: # svn checkout

Hi, I am a beginner in asterisk. Implementing it in my dept in India using TDM400b card with asterisk, zaptel, libpri version latest of CVS HEAD Callerid on my system is coming tough. Asterisk doesnot finishes the callerid spill and Cancells it. After going through code in Callerid.c and chan_zap.c I found that my line is providing caller id of length 8867. Flow enters in zt_call and

I am trying to help a guy out with his Atcom IP04. He has set it up like this. He has a handful of IP phones all connecting via SIP. He has two phone lines connected to the FXO ports one from telecom, another from vodaphone. He has set up the dialplan so that one of the trunks fails over to the other trunk. Everything seems to be working OK except for outgoing calls. He can call from

Here is a log from a recent call made out on a ZAP channel from a SIP phone inside my network. For some reason, CDR is billing time even though the "busy tone" was detected. It's also logging the call as ANSWERED. Is this normal behavior? Seems a little odd to me. I have this as the first 3 lines of my zapata.conf [channels] busydetect=1 busycount=3 CVS HEAD updated late

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

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

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

I'm trying to clean up some notices/warnings that are repeatedly logged in *.Any Help would be appreciated as I'm not sure of the cause /solution. Here are the errors: Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321 (zt_call): cidspill already exists?? +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ /* Don't send audio while on hook, until the call

Hello, i'm using a TE410P on some E1/PRI with EuroISDN and experiencing a few audio quality problems with current CVS (both zaptel and asterisk) and the following network ISDN public SIP/zaptel network ---- pri --- ASTERISK GW --- iax --- ASTERISK PBX --- PHONES w/ any codec the rx (public network to local

I recently purchased a TDM11B so I could hopefully hook flash the FXO from either the FXS (on the TDM11B) or a SIP device. From the FXS, I've tried hitting # then transferring to an extension that flashes the line then dials the FXS again (3020). This seems to send me to a busy signal and the console tells me no such host of 3020 (the number I'm on). The call on call waiting gets sent

I have put a box together using 2 X100P and 2 TDM400 4port cards. Using the simple setup that Martin posted a few days ago, I have asterisk almost up and running. /etc/zaptel.conf fxsks=1-2 fxoks=3-10 loadzone=nz defaultzone=nz (I have added NZ tone information to zaptel and Asterisk - I'll submit a patch soon). /etc/asterisk/zapata.conf [channels] context=incoming signalling=fxs_ks