similar to: iptables -d fqdn instead of IP

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

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

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

Good Evening, I am trying to forward packages on an internal device using iptables: /sbin/iptables -A FORWARD -i eth0 -o eth0 -m state --state NEW,RELATED,ESTABLISHED -j ACCEPT but the packages are still blocked, e.g.: Feb 6 20:58:28 firewall kernel: DROP-TCP IN=eth0 OUT=eth0 SRC=192.168.100.177 DST=172.28.2.184 LEN=40 TOS=0x00 PREC=0x00 TTL=127 ID=16609 PROTO=TCP SPT=7166 DPT=3590 WINDOW=0

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

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

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

Hi, I just got a Xorcom Astribank with 8 FXS but it does not work. So I tried resetting and loading the firmware. But loading just times out. /usr/share/zaptel/xpp_fxloader usb --------- FIRMWARE LOADING: (usb) [1 devices] 'xpp_fxloader'[11561]: USB Firmware /usr/share/zaptel/USB_FW.hex into /dev/bus/usb/002/003

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come over the other lines. -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director

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

I'm running dahdi on the host system, and have added the /dev/dahdi/ devices to the guest vserver as recommended in Beave's "Virtual Private Asterisk" whitepaper (http://www.telephreak.org/papers/vpa/). I tried copying libtonezone.so and libtonezone.h to the guest, but I couldn't anything to replace zaptel.h in DAHDI souce (it seems dahdi.h was deprecated?). I need to

Hello everyone. I'm struggling to get T.38 faxing to work in Asterisk 1.6.1.13 with a SIP DID provider here in Brazil (GVT - Vox IP service). Here's my scenario: When faxes arrive by a specific DID, they are routed thru this simple macro: [macro-recebefax] exten => s,1,Set(DB(fax/count)=$[${DB(fax/count)} + 1]) exten => s,n,Set(FAXCOUNT=${DB(fax/count)}) exten =>

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

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 =======================================

Hi all, is there a way to combine iptables parameters like: iptables -A OUTPUT -p UDP & -p TCP -d $IP1 & -d $IP2 ? Best Regards Marcus

Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk:

Good afternoon gentlemen (and ladies). A costumer of mine has many servers and each one maps their SIP extensions to the others via DUNDi. It works like a charm. SIP extensions can only register at one server, the one they "belong" to. In case one extension wants to call other that is registered in another server, DUNDi takes care of that by calling the other server using IAX2 and G.729

I've spent a week playing with Asterisk 1.6 and I love it. What a brilliant piece of software! Progress and learning have been reasonably good. I have external SIP provider calls coming in and have put together a little call platform and I'm stunned at the flexibility. There is one issue for me. I took me a while to click that ZAPTEL now equals Dahdi, but now I'm there I have an

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <span class="postbody">Greetings,<br> I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium TE121B PCI express card with a </span>VPMADT032<span

----- "Chandrakant Solanki" <solanki.chandrakant at gmail.com> escreveu: > Hello > > I have successfully compiled OSLEC for echo cancellation for DAHDI > channel. > > Is there any way to do echo cancellation for SIP Channel. > > Is any, please suggest me.?? > > Thanks in advance.. > > -- > Regards, > > Chandrakant Solanki Short