similar to: Connecting two Asterisk servers with a framerelay connection

Hello all, have anyone an idea about converting an audio file (.wav, .mp3, .au,...) to a ".gsm" audio file to use it as a voicemail file with Asterisk. Thanks. Abdelkader Mosbah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070812/2ffc7135/attachment.htm

Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070809/ddaed76b/attachment.htm

Hello, I have the following situation: A firewall is blocking all SIP and RTP traffic in the side of some of my clients. My clients cannot change settings of the firewall. I need to solve this problem and I need some help from you. I have this idea: implement a SIP user agent which does not use well known SIP ports (uses http port 80 for example) and use other ports that are not blocked

Hello All, MOS and R factor are the two QoS parameters used to estimate VoIP call quality. I have found that they are calculated from other metrics like jitter, latency, packet loss,...etc. But, haven't found any formula or arithmetic rule to calculate them. Do you have an idea about their formulas or an open source that calculates them. Is it possible to interpret them from wireshark.

An attacker is scanning my Asterisk Switch to gain illegitimate access to VoIP call functionality. Using a sip scanning tool, *it* sends REGISTERs with random identities. And when it discovers one identity subscribed in my switch, it tries to authenticate with random passwords using this user name. For the moment, I have replaced this account. And also blocked the IP it has used but each time

Hello all, After installing Asterisk, i have installed the docs by "make progdocs". But i don't know where to locate this documentation. please Help. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070810/ceb95948/attachment.htm

The failregex statement in my jail.conf file is: * failregex* = NOTICE.* .*: Registration from '.*' failed for '<HOST>' - Wrong password NOTICE.* .*: Registration from '.*' failed for '<HOST>' - No matching peer found NOTICE.* .*: Registration from '.*' failed for '<HOST>' - Username/auth name mismatch

Is there anyone interested in developing an open source Asterisk / MS Exchange solution? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com <mailto:michael at highpoweredhelp.com> Attachment encrypted? click here <http://www.highpoweredhelp.com/tutorials/wincrypt/> . -------------- next part -------------- An HTML attachment was scrubbed... URL:

Will this work to match any number from the 770,404, or 678 area codes? _[404|770|678]NXXXXXX If this won't work, is there a pattern that will do this? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com <mailto:michael at highpoweredhelp.com> Attachment encrypted? click here <http://www.highpoweredhelp.com/tutorials/wincrypt/> .

I've found OpenVPN to be easy to configure and very robust. It has a zillion options, but they are just that - options. I haven't used it for VoIP, but I've put it to good use doing layer 2 bridging which has eliminated many problems with certain programs traversing NAT and load-balancing routers. I can't think of any reason why it would not work well with Asterisk. > On

We have an asterisk system and Polycom phones that were provisioned by someone else. They want to get call waiting to work, but for the life of me, I cannot figure out why the Polycom is returning a SIP 486 Busy Here when you call and the person is already on the phone. I have the feeling there is a configuration in sip.cfg or mac.cfg that I am overlooking. Any thoughts? Calls per line key

hello, I have installed asterisk 1.2.18 in suse 10.2. After typing asterisk in the terminal command line (i don't think that asterisk runs when doing this) i type "asterisk -r" but the response" is "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)". how to solve this. thanks. -------------- next part -------------- An HTML attachment was

Hello, Is it possible to transfer ss7 signaling over an FXO interface. I need to setup an ss7 test system composed by two Asterisk based IP-PBX systems with anlog interfaces only (FXO and FXS). I want to know if it is possible to connect the two IP-PBX as following: - FXS interface in PBX1 -----------------> connected to -----------------> FXO interface in PBX2 =============>

Hi, Thank you so much for your reply, I've a dedicated server in OVH, where I have done speed test for the server : *bkf at xxxxx ~> speedtest-cli Retrieving speedtest.net <http://speedtest.net> configuration...Retrieving speedtest.net <http://speedtest.net> server list...Testing from OVH SAS (x.x.x.x)...Selecting best server based on latency...Hosted by fdcservers.net

I am trying to use PHP to reload the extensions in an Asterisk installation. I keep getting this error: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) when I run the script by visiting the URL; however, if I run the script from the command line, it runs just fine (works perfect, actually). I think it is permissions related. Does anyone have any ideas? <php

Polycom Provisioning Tool Updated. I made a bug fix that was reported, which was causing the directory creator to not work when there was an invalid character in the filename of the csv. I have also posted an FAQ: http://www.wintrisk.com/ppt.html#FAQ Download the new one, and tell me what you think! It's free, and mildly useful! http://www.wintrisk.com/ppt.html Yours, Michael Munger,

In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here.... Any help is appreciated. Here is features.conf: ; ; Sample Parking configuration ; [general] parkext => 700 ; What extension to dial to park parkpos => 701-720 ;

Hello All, I have installed Asterisk 1.6 with openVPN in the same machine. I have set up a VPN connection between 2 SIP clients and Asterisk using x-lite. The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn tunnel. When attempting to make a call between the clients, the siganling part of the call goes well. But, when the call is set up, some RTP packets are exchanged at

This seems like something I should know... but.... I don't. How do you update zaptel / libpri on a Business Edition box running rPath? Tried running conary, but got 'Insufficient permission to access server conary.digium.com." Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com -------------- next part -------------- An HTML attachment was

I have a TDM800 that is installed and working. (TDM800 + 2 X QUAD FXO). Zttool says it is configured, ok, and there are no issues. Ztcfg -vvv shows that all the channels are configured. Zap show channels in the CLI show all 8 channels configured as they are supposed to be. When I plug in a pots line from the telco and make a call to that line, asterisk does not respond. (No Starting