similar to: SIP call from one extention to another

Hi All, I know, IAX is NAT friendly, but... I have a problem running gnophone from a box behind NAT firewall. I can register gnophone with * through NAT, but when I try to make a call it instantly disconnects. CLI iax show peers command tells me that peer is unreachable. However this peer is registred. Gnophone also tells me that it is registred. It seems that registration handshake has

Hi what can be wrong with * that console does not show any stderr text printed from agi script? I am starting with asterisk -vvvvvvvvvvvvvvvvvvvvvvvvrc VERBOSE command does show text on console but printing of STDERR does not I tried it from Perl and from Java and in both cases almost the same result, except in Java more things do not work. In Java for, for example, SAY DIGITS 123 78# would

Does anyone know if this unit works with Asterisk? Thx. B.

Greetings... We've been having some interoperability issues between Asterisk and an AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 somewhere. So, I've been pondering using iptel.org's SIP server (SIP Express Router) as a "front end" for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

Hi everybody, I want to read to debug messages and try to interpret them but they happen too fast, how can I log these guys to a file, or is there a file like this already? I checked the /var/log/asterisk but there isn't much interesting there yet? How can i turn on logging for SIP,IAX and other things? Thanks, Umut

I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause

Hi all! I am looking for some free software to monitoring all the calls that are being done in my network. Which telephone are connected, how long are the calls, quality of service, bandwidht,etc. If someone knows about a good one, plesea tell me. Regards, Mireia

Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS

Here's a clip of comments lifted from a Cisco bug list. This will be perhaps useful to those of you who have just purchased a Cisco phone off eBay. JT ------------- (1) Short problem description: Documentation on how to load SIP image on phone with skinny software (2) Longer problem description (what happens): If the phone is loaded with the Cisco Skinny code, then there is a small

Hello Everyone, What kind of hardware setup would I need to do if I want a T1 connection to PSTN and have 48 users in office with analog phones. Will something work if I have a T410P card in asterisk and have one T1 going to PSTN and other two to a channel bank. I would then connect the 48 phones (FXS) to the channel bank! Thanks. Deepak

I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 & also soft hangup

Hi I am using inAccess channel driver. Compiled, installed. This is what I get when I am trying to start * --------------------------------------- [chan_oh323.so]WARNING[16384]: File loader.c, Line 226 (ast_load_resource): libh323_linux_x86_r.so.1.12: cannot open shared object file: No such file or directory WARNING[16384]: File loader.c, Line 394 (load_modules): Loading module chan_oh323.so

Hi I want to set up * as a conference bridge. I would like to be able to conference is SIP calls (up to 12) I am looking through all available documentation for * to get info on how it is done. No luck so far. Can somebody direct me to the info in this subject? Thank you Serge _________________________________________________________________ Protect your PC - get McAfee.com VirusScan Online

1st. - I'm from Poland, we don't have (yet, and hopefully forever) software patents. Is there any free g.729.1 implementation for asterisk? I want to use it for my private use (dialing into inet->PSTN gateway), and I don't want (now) to buy codec, as I don't know if I will be using this service in future (now I just want to test it). Any solutions? Maybe even

Hi I am trying Manager interface for originate a call. This is what I get --------------- Action: Originate Exten: 555 CallerID: test <6656> Context: local Timeout: 600 Channel: SIP/8782 Priority: 1 Response: Error Message: Originate failed ---------------- What do I do wrong? Thank you Serge _________________________________________________________________ MSN Premium with Virus Guard

Hi, I am trying to compile Asterisk under Yellow Dog 3.0 distributionn. I am getting an error gcc -shared -Xlinker -x -o codec_gsm.so codec_gsm.o -lgsm /usr/bin/ld: cannot find -lgsm May be I need packages that my distribution does not include? What do I need to download to get it compiled? Thanks Serge _________________________________________________________________ The new MSN 8: smart spam

Hi I am running firewall/router "brew" made of RedHat, Shorewall, Siproxd and GnuGK on a box that connects through PPPoE to Internet. I run Asterisk on another box behind of it and it seem to work fine for me. I am thinking of replacing the router box because hardware is getting flaky. I do not want to go through pain of assembling all this stuff together again. Does anybody know of

Hi, I updated kernel for RH8.0 and updated * from cvs. After that zaptel compile exits with error. Is it because of new kernel or zaptel source code change? In any case could somebody help me to fix this problem? Thanks, Serge ---------------------------------------- /usr/src/linux-2.4/include/linux/module.h:196: warning: parameter names (without types) in function declaration In file included

Greetings... I have a TE410P with four T1's going into it. Things look roughly like this: #1 Goes to PBX -- we're responsible for timing #2 E&M span to telco 1 #3 PRI span to telco 1 #4 PRI span to telco 2 If I set primary sync source to span 2, users report strange echo, distortion, and crosstalk problems, which sound remarkably like frame slippage on spans 3 and 4. If I set