similar to: Bind port

Roman: I created two bash scripts called Mail2Fax and Fax2Mail for use with the asterisk sever. They leverage the app_txfax and app_rxfax scripts, along with ast_fax. They make using these apps a lot easier, including being able to mail to fax@domain.ca for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules /

Hello Everyone, This is my first post, and this is my problem :-). I have a asterisk@home, work excellent (only internal users), but i need outbound calls. One person give me an access to his "Cisco 5300 Media Gateway", he give me a dial rule and the router ip address. I've created a SIP Trunk, and a outbound routing, with all the info (the rare thing, the

I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server was experiencing crackly audio on one circuit, accompanied by HDLC bad FCS messages. The telco recabled and moved me to another port on the DMS-100. The audio is better, but there are still bad FCS problems on the span. I have moved the PRI in question to the other server, and the problem does indeed move with the

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven

I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is

I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 x86_64 very good, but since FC keeps updating, I tried to follow newer kernel versions. I can't pass the zaptel compilation. Everything is OK, but when I finished, and tried to load it, allways got module not found when I run modprobe zaptel, and modprobe ztdummy. I already tried to modify is with the sed 1 option but

Hi all, I'm newbe with asterisk and i'm facing with this problem that i'm not able to solve. I've to put an asterisk box after an analogic pbx wich require a 0 digit to give the dialtone. So when a client ask asterisk to dial an extension it should 1) send the 0 digit 2) wait for the dialtone 3) dial the extension the client send. How can i obtain this result? Thank's in

Hi list! I have several boxes running asterisk as I want, no problems there but the one thing I haven't sorted out is how to properly start asterisk on boot time. This is probably a n00b class question but how do I properly set this up (I didn't find any docs on this). The zaptel script alone does not load the proper driver on boot time, I guess I need to do some thing with the

Yes, I can call the phones, they ring, etc, and call call out, just no outbound audio. Is their any difference in the inbound & outbound audio streams in Asterisk that could cause it, e.g., different ports, protocols, connection/discovery methods, etc? Thanks, Ross ---------- Original Message ---------------------------------- From: "Carlos Alperin"

Hi FOlks, I am trying to setup remotely an "AudioCodes Mediant 2000" MG Module 2 to work with Asterisk through SIP or H323. But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Thanks a lot, Isamar

I finished to try to install Asterisk Now 1.4.0 on an AMD 3800 dual processor machine. The install lookups on the search for the Sata drive, since however it loads the sata_sil driver it doesn't work. Did someone knows what version of Linux is using on Asterisk Now? Thanks, Carlos Alperin -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am looking at phones for my asterisk system and seem to have a problem. The only Power over Ethernet phones I can find that support the IEEE standard are 3com. Cisco uses its own proprietary ( and is expensive to boot ), snom has a different but equally non-IEEE method, and i'm havent found another phone that I'm confident can do the job for our office. Whats a good high quality ip

Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan,

I need to find someone to work with me in the Grand Rapids Michigan Area. Someone good with Linux and Asterisk would be ideal. Please get me contact info if you are interested. Thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/315eff3f/attachment.htm

This is all the reference to PTHREAD_MUTEX_RECURSIVE on lock.h #ifdef __APPLE__ /* Provide the Linux initializers for MacOS X */ #define PTHREAD_MUTEX_RECURSIVE_NP PTHREAD_MUTEX_RECURSIVE #define PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP { 0x4d555458, \ { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0x20 } } #endif #ifdef BSD #ifdef __GNUC__ #define

Hi, I'm new to asterisk, and have a uniden UIP 200 that I got off of ebay. I'm having trouble getting the phone to register with asterisk. I've tried a few different settings. I'd be extremely grateful if someone with a similar setting could give me the sip.conf block for the UIP and the settings you're using in uniden.txt. Here's what I have currently: IP of phone

Hello Everyone, Pardon me if im sounding like a total idiot, but im new to this and have to ask. Numerous people have been telling me that I will be able to somehow do long distance calling for free when I roll out Asterisk.. and yet none of them can explain to me how exactly that will be. So.. I'll ask the community at large.. is this total BS or is there actually a way to reduce my long

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

Before I go ahead and spend $40.000 on a Cisco 5400, just because my clients need T.38 faxing, I want to ask the community if there is any chance of having Asterisk receive G729+T.38 and sending the call via Zaptel to its final destination. Any answer will be appreciated. Federico

Hi all!, i search for some information about to setup my asterisk box with e-mail notification when a I call the voicemail application. Voicemail application works fine in the Dial Plan but nothing happens with email notification ...so what i need to know about this?...wiki pages did not help me ....thanks! G. ----- Original Message ----- From: <asterisk-users-request@lists.digium.com>