similar to: How to use OpenVPN with Asterisk

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, 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 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

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

What modules do you want on it? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of MOSBAH ABDELKADER Sent: Saturday, August 04, 2007 3:16 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Connecting

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 =============>

I downloaded xen-3.2.1.tar.gz from xen.org, installed all dependencies and make world on ubuntu hardy. It compiles fine until the errors below.... ------------------------------------------------------ ... ... install -m0644 -p include/public/foreign/*.h /usr/local/src/xen-3.2.1/dist/install/usr/include/xen/foreign install -m0644 -p include/public/COPYING

Hi all! After that good thread "DGD patch not detecting dead gateway" I was able to set up a Load Balancing with ping based DGD (without Julian Anastasov patch). But now I''m facing a new problem and tried some options, with only partial solutions. I made a script based on http://www.mail-archive.com/lartc@mailman.ds9a.nl/msg16257.html (Thank you Manish Kathuria),

Hello all! I applied http://www.ssi.bg/~ja/routes-2.6.8-10.diff patch to kernel 2.6.8.1 and it works fine, or almost fine. It does the load balancing well, but when one link is dropped it continues to try it. At the end of http://www.ssi.bg/~ja/nano.txt it is said to ping gateway 1 and gateway 2, for the kernel to know if that route is working, but since my linux is connected to the links

How they differ? Does the to_xml method of the classes array/hash/object is based on ROXML lib? As I see in http://roxml.rubyforge.org/#quickstart, roxml needs we define the class (class Book in the guide) before to convert a object to xml. Moreover, the method to_xml (when not using "include ROXML") just accepts any existing object, what is much suitable. I need to compare and tranfer

Hello, What is the maximum number of simultaneous calls supported by asterisk. thks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090703/0794c554/attachment.htm

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

(Dovecot 1.0.rc15) Hello I'm tring to configure postfix+dovecot on debian etch. In dovecot I use passdb pam and userdb passwd-file. I can connect pop3 clients and reveice mails and connect smtp clients and send mails, but it is delivered to /var/mail/%u instead the configured mailboxes in /var/local/dovecot/%u. I`ve configured postfix as say http://wiki.dovecot.org/LDA/Postfix.

Hi all! Is there a way to asterisk-gui to allow underline (as such cpd_tom) in Names? It allows to [di]enable alphanumeric, but not underline noway. Why such restriction in asterisk-gui if even asterisk users.conf allows (and works fine) it? Thank you, Tom Lobato

Hi all, First of all, thank you for supporting us. I've installed Libretime (https://github.com/LibreTime/libretime) which user Icecast2 on Ubuntu Server 16.04. I've also developed a mobile application to listen to stream myhost:8000/mount. My problem is when the number of listeners increases up 500 (On live streaming with butt), the application blockes, and I hardly get info from

could someone explain some of the uses for a web service. i''m reading about how to make them, but i''m still not quite sure what they''re good for. i have multiple rails apps that i would like to pass user information back and forth to. i''m not sure if something like this could help me or not. thanks -- Posted via http://www.ruby-forum.com/.