similar to: BYE from Cisco gateway

I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y So far so good. But I want to setup VOIP sessions with local carrier. I added dial-peer 40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN

Dear list, I'm looking for a function to generate (simulate) a random Weibull point process. Can anyone help? Cheers, Torbj?rn Ergon, University of Oslo

Hello. I would like to run asterisk as an user other than root. I have seen some tutorials on the web, but I would like to know if there is some ?official? how-to for this. Is there? I looked at a thread on reviewboard regarding this (https://reviewboard.asterisk.org/r/654/). It was Paul Belangers work trying to make the installation process take care of this. But the conclusion seem to

Hello, We are setting up an asterisk system for voicemail with video possibilities. We are not using the voicemail app, but rather writing our own dialplan logic. The part of recording, and replaying, the voicemail works, and we receive both an h264 and an wav-file. What I now wonder is how to convert these into one file playable by a (standard) media player. I have not found any real good

Hey, all. I'm currently trying out GlusterFS 3.3. I've got two servers and four clients, all on separate boxes. I've got a Distributed-Replicated volume with 4 bricks, two from each server, and I'm using the FUSE client. I was trying out failover, currently testing for reads. I was reading a big file, using iftop to see which server was actually being read from. I put up an

I've made packages for Debian Squeeze for Gluster 3.4.0-alpha2, they are available on http://torbjorn-dev.trollweb.net/gluster-3.4.0alpha2-debs/. They built and installed successfully, and have been running nicely for a couple of hours, but your mileage may vary. The Debian packaging is on http://torbjorn-dev.trollweb.net/gluster-3.4.0alpha2-debs/glusterfs-3.4.0-debian.tar.gz. I took the

Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first

Hello! We are using the 1.2 branch, and upgraded to 1.2.26.1. We ran into some problems when using realtime for peers. We connect the PBX to a sip peer at an ITSP, and when we try to dial the peer we get: Jan 23 09:02:07 VERBOSE[2236] logger.c: -- Executing Dial("SIP/dev02-08c36f28", "SIP/3246 at 989800-out||W") in new stack Jan 23 09:02:07 DEBUG[2236]

[1.3.20]-Capture The Mouse Works Great. Suggestions... Hi, I've been crying for a long time about mouse issues while running a full screen game in an emulated desktop window, but with newest Wine these issues are gone. Would it be possible to have some hot key that the user can press to uncapture the mouse? For example, VirtualBox toggles mouse capture by pressing the right [Ctrl] key on

Hi guys I have met some troubles when I want to evaluate the glusterfs performance with small file workload. 1: What kind of benchmark should I use to test the small file operation ? As we all know, we can use iozone tools to test the large file operation, while for the sake of memory cache, if we testing small file operation with iozone, the result will not correct.

Dear All : I need to link my HQ to some Remote Sites - I need a Router which supports VOIP , and VPN Also the Router Should has 3 FXS ports and 1 FXO ... The call should be routed from the Remote Site to the HQ through a VPN tunnel ( 3DES ) ... Any Advise ? Mohamed Farid ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are

Hello I have a problem where I would like to be able to send an arbitrary SIP domain when sending a call to a registered friend. By default the from domain is set to the IP of the Asterisk server, but I would like to set it to something else. The case is that when a call from a foreign domain comes in to the Asterisk, it will connect it to the callee (but with the domain changed). When

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

Otherwise 'modprobe -r' on a module having a dependency on bridge will implicitly unload bridge, bringing down all connectivity that was using bridges. Signed-off-by: Jan Beulich <jbeulich@novell.com> net/bridge/br_if.c | 9 +++++++++ 1 file changed, 9 insertions(+) --- linux-2.6.23-rc8/net/bridge/br_if.c 2007-09-26 09:23:54.000000000 +0200 +++

> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > > the JITTERBUFFER function? > > You only need to use the JITTERBUFFER function. > > The jbenable option will enable a jitter buffer on every channel > created for that peer (or, if global, for every peer in the system). > Depending on the version of Asterisk, it will also place the

I have several 7960 phones with SIP image (7.3) and Asterisk 1.0.1 on FreeBSD. When I have 2 active SIP calls on the 7960 phone there are no available lines for additional calls. I tried to configure 2 lines to the same SIP server but it's still limited to 2 calls. How to utilize all lines? -- Called user -- SIP/user-acc6 is ringing -- SIP/user-acc6 answered SIP/x.x.x.x-09a9a000 --

Hi all, I'm new in using dovecot and I'm running in a problem, which I cannot solve without hours and hours of work. Maybe there is a solution, which I didn't find with old friend google :_) Here is the problem: We are using an old mailserver (it's running fine for 5 years) with qmail and the old uw-imap server. All users have their grown directories under a home/Maildir folder.

Problem: I have two computers (one at work and one home) and I want to keep them identical, but I cannot rsync them directly because when one is running then the second is turned off. I also have access to 500 MB of storage space on a server running continuously, which is not nearly enough to hold all data stored on the two computers. So my idea is to use the server for storing an rsync file

Hello! I am going to use the JITTERBUFFER function in a SIP (and local channels) only setup, but have some questions of how to use it: 1. Do I need to activate jbenable in sip.conf? Or is it enough to call the JITTERBUFFER function? 2. What is the preferred way to invoke this function? Say I have channel A which is not in need of buffering, while channel B do need it. If A

If I can get this application to run under Mandriva I can kiss Windows goodbye. Yeah. The application is called MB Navigator. It is a stand alone stock trading application linking me to my brokers servers. The application installed great. Upon executing the program is starts up just fine then I get this error: Can't create object:ADOC command I can't produce a log.txt file because the