similar to: OT: Polycom Firmware available (by accident?)

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

Hi, I'm getting some strange behaviour on Asterisk 1.4 running on Debian Stable (Lenny). I suspect it's something to do with my setup, rather than a bug, but I'm struggling to see it, and would appreciate any input. Setup: PC with two ethernet cards: eth0 goes to local network, including two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes to router and

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963] zttranscode: Loaded. [ 7.618969] wctc4xxp: tc400b0: Attached to

Has anyone figured this out yet? Lots of places say to add the following to sip.conf of an Asterisk 1.2 system (current production machine/Asterisk as root): tos=0xB8 (Hex B8 = Decimal 184 = Binary 10111000) or if you are running Asterisk v1.4 or newer: tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ;

This message is pointed directly to Steve Underwood. I tought it would not be nice to directly email him with a question that interests a good part of the Asterisk community, so here it is. :) Steve, remember a few days ago when we discussed about issues on Asterisk 1.6.1.13 and T.38 fax reception? Well I opened an issue on Mantis (https://issues.asterisk.org/view.php?id=16756) and turns out it

hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs at fpf.slu.cz --------------------------------------- Marek Cervenka =======================================

Hey All, AT&T is installing a PRI in a couple weeks and while I've been doing homework on PRI's for the last few weeks there's something I'm still confused about. After being asked how many digits I wanted them to send us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked her what that meant and all I got was the question repeated. Do any of you have

Hello everybody I am trying to connect my asterisk to a payment equipment trough PSTN. I have a TDM400P card with an fxs module an the equipment use modem to send data! I was thinking to implement a software data modem in asterisk, but I found out that there is just faxmodem for asterisk, Is anyone here know something about software data modem working with asterisk to help out? Thanks,

Hello all, I have two machines I'm connecting with TDMOE (dahdi dynamic spans) and I have a question about timing parameters. By my understanding one machine should be the source of the timing and the other a slave of that timing. So on machine A I have the following in system.conf: dynamic=eth,eth0/00:0C:29:55:89:7E,24,0 On machine B I have this is system.conf:

Hello all, I'm setting up a couple of test boxes and I'm running into a problem. What I need help with is determining whether I'm going something wrong or if I need to post a bug report. I have two asterisk 13.0-beta 3 machines set up with extensions connected to each as such: 3700 ----> AST-A <------> AST-B <---- 3800 & 3801 When I place a call from 3800 to

Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons. Atenciosamente, Vin?cius Fontes Gerente de Seguran?a da Informa??o Canall Tecnologia em Comunica??es Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunica??es Passo Fundo - RS - Brazil +55 54 2104-7000

I'm having issues with T.38 on Asterisk 1.6.2.8. A few lines are received OK and then I get only garbage. I'm using ReceiveFAX() provided by app_fax to receive the faxes. After talking to the engineers on the telco, they said Asterisk is sending a REINVITE to alaw after the T.38 reception is complete, and that could be the cause of the problems. I personally am not totally convinced of

Quick and dirty question: for the TDM410P I must use the wctdm24xxp driver? Att Vin?cius Fontes Desenvolvimento Canall Tecnologia em Comunica??es Ltda.

Hello everyone. I got a little problem in here: I want to set up a queue so that if anything of these happens: a) No agents logged in b) All agents busy Then the user gets diverted somewhere. I used this (for testing purposes only, of course): exten => 7080,1,Answer() exten => 7080,n,Queue(teste) exten => 7080,n,Goto(${QUEUESTATUS}) exten => 7080,n(ERROR),NoOp(${QUEUESTATUS}) exten

To the volunteers of R-Help. Hello, I am currently stuck on an RStudio assignment. The assignment involves creating a double bar graph with the provided info http://math.fullerton.edu/mori/data/introstats/pennstate3.txt My professor has only gone over the very basics of RStudio and we only learned how to make a simple bar graph and labeling x and y axis. The specific directions from my

Hi all, I''m a software developer from New Zealand. I''ve built a couple of rails sites and have some ajax experience. I''d like to develop on a rails site with the aim of making a business - is anyone else looking to do the same? The two rails sites I built recently are foopad.com and friendr.com. Regards, Ben

I'm playing around with an ALIX 2D2 board (http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an AMD Geode processor with 256MB of RAM. Also available are two network interfaces, two USB ports and one serial port (no keyboard or VGA). I'm using the Voyage Linux distro, which basically is Debian Lenny optimized for this board. Asterisk 1.6.1.12 runs fine on the

Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) <-> Sipura 3k (fxo) <-> Asterisk <-IAX-> remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the

I have a big headache. I have an Asterisk server connected to an Avaya PBX. Everything is working between those two. The problem is that I have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the Internet to the Asterisk server through a Fortinet firewall. When calling from a PAP2T I get one way audio, the remote site can hear me but I cannot hear them. If I do an "rtp

Hello all, I need the bootrom.ld file to set up some Polycoms I have Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A I've publiched on my FTP files downloaded from http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html (3.2.3 combined and split zips) but my phones are still showing the message: "error, application is