similar to: SIP signaling vs Media (Voice) Traffic

Hi, I'm having trouble compiling zaptel 1.4.4 on SUSE 10.1. I'm really only interested in getting ztdummy to work because this is a dev machine with no zaptel h/w. SUSE 10.1 is a 2.6 kernel: asterisk-dev:/home/hugh # uname -r 2.6.16.13-4-default It seems that my problem is related to autoconf.h - I cannot find that file: asterisk-dev:/home/hugh # find / -name 'autoconf.h'

Hi, I'm running a 1.1 version of Asterisk (a stable build from back in Oct-05) running on RedHat 9.0. Everything's been great but a couple of days ago, we all stopped receiving emails of our voicemail. There's been no changes to our configuration I bet I'm expereiencing a Linux problem rather than an Asterisk problem, but because I know only as much Linux as required to get

Asterisk 1.2.7 RedHat 9.0 Hi, I've run into some voice degradation problems with IAX2: I frequently have calls come in on a DiD provided by an ITSP. I often have to redirect these calls back out to the PSTN (i.e. to a cell phone). When this happens, I don't want my server in the media path, I want to hand it off to my ITSP instead and let them handle both ends of the call. I've

I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent at best. Sometimes it's great other times completely unusable. When it's bad one usually hears harsh static when the other party speaks or their voice gets "clipped" to static if they speak too loudly. Many of these users have migrated to Skype ? much

I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). Before I implemented Asterisk, some users were using Bell services to set-up 5 way calling: The user would set up a three way call on one line, switch to the second line, set up another 3 way call and then link the two lines together with the Flash key, thus establishing a 5 way call (the user, 2 others on line 1, another 2 on line 2). How

Hi, I'm installing * 1.4.9 and a couple questions have come up: 1) I read here<http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x>( http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x) that if you are using E1 cards you need to install LIBPRI. I'm not using any cards on this system, so does that mean I don't need LIBPRI? Asterisk built

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Curious whether the ChanIsAvail command is thread safe. By that I mean, if I use ChanIsAvail to determine which channel to use, can I be sure that it will still be available when I go to Dial it on the next line? It occurs to me that there's a possibility the channel could get used by a competing thread AFTER my thread has determined it is available and BEFORE my thread gets a chance to

Greetings all- Here's an odd question. Supposedly, IAX2 now has the ability to operate with signaling and media in separate streams, very much like SIP. I've read about this feature here[1] and there[2], but I have yet to see how to actually implement or test it. There are no options in the iax.conf sample configs with Asterisk. All suggestions welcome, except those telling me to jump

Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _________________________________________________________________ The

I've been asked to assess the cost of implementing Asterisk with a single T1 line in one of our offices. I've had plenty of experience w/ TDM400 cards, but T1 is new for me so a couple of questions: 1) Will I need a digital or analogue interface card? I expect digital is the answer, but the Digium web site said something about analogue cards being able to support "provider T1

I'm trying to add MySQL CDR recording in Asterisk 1.4.6. I'm following the instructions posted here: http://www.voip-info.org/wiki-Asterisk+cdr+mysql I have MySQL installed and it works fine - starts on stratup, I can create DBs, tables and so on and I can connect through php. rpm -qa indicates: MySQL-server-5.0.22-0 MySQL-devel-5.0.22-0 MySQL-client-5.0.22-0 However I still get XXX

Where is the correct place to control what traffic is masq'ed out? This is what I have, but I was told the Forward chain isn't the right place to do this? iptables -A POSTROUTING -t nat -o $WAN -j MASQUERADE iptables -A FORWARD -i $WAN -o $LAN -m state --state RELATED,ESTABLISHED -j ACCEPT iptables -A FORWARD -i $LAN -o $WAN -m state --state NEW,ESTABLISHED,RELATED -p tcp -m multiport

I have two Asterisk servers running on the same LAN. One starts fine, but when I start the other I get: pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use and Asterisk does not start. OK I thought, I'll just change the port in dundi.conf. I changed it to 4521 and indeed it started just fine, until I reboot that is. WHen I reboot, I get the same

Anyone know how to set up port forwarding of multiple ports on a DLink DI-604? I successfully portforward the SIP port on the Advanced|Virtual Server page. It works because I can register a SIP client, but it's a single port - 5060. The DLink doesn't seem to provide an obvious way of portfarding the 10000 - 20000 ports needed for RTP. Any ideas? Hugh

Just installed Asterisk 1.2 on a brand new clean machine running RedHat 9.0. I have a TDM400 card inside. When I boot, the card seems dead. When I do: modprobe wctdm modprobe Zaptel the lights come on and all seems fine, until I reboot that is... After a reboot I have to repeat the modprobe. I shouldn't have to do a modprobe every re-boot should I? How do you get the drivers to load

On Tue, Sep 16, 2008 at 2:21 AM, Michiel van Baak <michiel at vanbaak.info> wrote: > On 22:46, Mon 15 Sep 08, hugolivude wrote: >> I have two Asterisk servers running on the same LAN. One starts fine, >> but when I start the other I get: >> >> pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: >> Address already in use >> >>

Hi, I am trying to configure the SMTP service on DMZ host. Added the rule: ACCEPT wan dmz:66.58.99.84 tcp pop3 - ACCEPT wan dmz:66.58.99.84 tcp 25 - ACCEPT dmz:66.58.99.84 wan tcp 25 - ACCEPT dmz:66.58.99.84 wan tcp pop3 - issued shorewall clear, shorewall restart, but still couldn''t telnet to the mail server

I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked "Enter the number of packages, followed by the Pound key". I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that