similar to: Website callback

When attempting to build zaptel I get the following error: configure:2184: error: C compiler cannot create executables vi config.log configure:2066: $? = 0 configure:2073: gcc -v >&5 Using built-in specs. Target: i386-redhat-linux Configured with: ../configure --prefix=/usr --mandir=/usr/share/man --infodir=/usr/share/info --enable-shared --enable-threads=posix --enable-checking=release

hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. waht do you think? thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Since i cant get latet beta of zaptel installed on the latest test version of Debian with kernel 2.6.17-2-686 can someone who is using debian give me some tips on how to get it working and installed? Many thanks, Christian

Hi list! I'm totally fed up with bristuff (or it's instability with a simple HFC-S card), 2 out of 3 times when people try to call they get the information tone that the number is not connected. I would like to try vzaphfc and I am looking for information on it. >From previous posts I found that the only place where the sources seem to be maintained and available is at the debian

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 >> >>

I am having trouble with chan_zap.so not loading. When I load it from modules.conf, Asterisk bails out without any error message. When I load it from the console, it just says "Unable to load module chan_zap.so" no matter what verbose level I am using. dmesg says: Zaptel Version: 1.4.4 Zaptel Echo Canceller: MG2 Freshmaker version: 73 Freshmaker passed register test Module 0:

Hi, I'm stuck writing a Web GUI because nothing out there is exactly what I need. I'm not writing something as extensive as what _is_ out there, but just something that allows users to change where their calls are forwarded and other small things like that. What I wanted to know is what is recommended by those you successfully wrote their own UI : 1) Modifying the config directly in

Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a x86_64 box (I guess nobody is using x86_64 platform or is able to fix this themselves?) First of all when bristuff is downloaded and compile is started it appears that the bristuff Makefiles are badly broken. The asterisk Makefiles all do see to find the kernel sources on a RHEL4 box in the proper directory, the pure

Any potential testers eager to build imap storage support using proper Debian packages: Resonably up-to-date packages of c-client (uw-imap) 2004/2006 are by now only availble from experimental: http://packages.debian.org/experimental/mail/uw-imapd On my test Etch system I simply downloaded the sources of those packages nd rebuilt them (debuild). IIRC with that installed the configure script had

Is it possible to start asterisk in the foreground ("asterisk -fc") and later detach from the terminal but leave asterisk running? thanks, James

I'm presently working on a project to build a scheduling system accessible by both web and phone. on the web side one can query what items are available when by using the time or the item as a key then reserve for an available time slot. reservations may also be modified by the user that made them or an admin. Where may I find examples of doing similar things with asterisk? all I've

Hi, Has anybody seen a non commercial, or freeware, or GPL, or even CHEAP... POP/IMAP to Text-to-speech? I have a working version for POP3 using festival. It DOES work... it even cleans the email contents to get the actual content. It works great with Outlook emails and similar, and skips non multipart/alternative (that would be mainly SPAM, where the email is just

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, First of all, hello again! Been a while since I've posted to the list, but I've been here lurking and watching ;-) Anyway, I wanted to pose a general question to the list to see if it turns up new suggestions for everyone/me. What is your preferred editor when coding in the dialplan? This is mainly aimed at those of you who write

I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061028/87014bd9/attachment.htm

Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on Fedora Core 4. I am making automatic call-out campaign with this setup on 4 PRI. The scripts for this: ==================================================================== caller php script write this to outgoung folder: fwrite($outfile,"Channel: Zap/g1/$phonenumber\n"); fwrite($outfile,"MaxRetries:

Hello! I have an Asterisk@home instalation with 7 users working OK, and I'ld like to implement either a -- Web dial feature, where the user would fill one form field with a phone number and a connection would be created between his extention and the entered number. OR -- Dial using an URI (callto:xxxxx link in a web page), having AstTapi installed and configured in all workstations.

Hi all, I've just being wondering if Dundi has the same purpose as ENUM. I don't know much (actually almost nothing) about these technologies. As far as I know they are a kind of DNS resolver used in the VoIP context. For example, user ronaldo at doamin.com has the extension namber 1001. This way nobody has to know the ronaldo's extension number. I'll appreciate if someone

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998