similar to: Call generator

Hello, Can someone explain me how to do caller based routing. Here is my example. I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture. My problem is when I want to place fax. The calls between the 2 sites are in gsm codec. So the fax doesn?t work! Is there any possibilities to do caller based routing in

Hi, sipp (http://sipp.sourceforge.net/) seems to be a good app. Take a look at http://www.voip-info.org/wiki-SIPP on the wiki to have more info about it... Basically, there is scenario which are describe there and I personnally generated about 3,000,000 calls before having to restart asterisk and i placed about 90 concurrent calls. Good luck! -----asterisk-users-admin@lists.digium.com a ?crit :

Hello, Does someone know how to setup snmp monitoring on asterisk. I?ve plan to deploy 50 asterisk, so I need some monitoring tools. I try with nagios as I read in the wiki, there is some project on it, but I can?t reach the end. Can someone help me? Thanks. GIBERT Fr?d?ric Ste VigiNetworks Mobile: +33 6 72 08 35 16 -------------- next part -------------- An HTML attachment was scrubbed...

I have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring

Thanks for this informations. Do you know where I can find the icd-snmp package for a redhat 9 distri? I can't find it. Thanks. Message: 6 Date: Fri, 09 Jul 2004 15:45:57 +0200 From: Andrea Fino <af@faino.org> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SNMP Monitoring Reply-To: asterisk-users@lists.digium.com Holger Schurig wrote: >>You could try

Hello, I had installed several asterisk, but I every time had a problem with callerID. On each phones I don't reveive the first digit. For example: Caller 0672083516 called an IP Phone 0123456789. The IP Phone see 672083516 as callerID. I think there is a patch for it, but I don't know where to find it out. Thanks. -------------- next part -------------- An HTML attachment was

Hi, I am working on Trixbox. I want to create my own dial() function (named specificdial()) and I want to know how I can create a module and integrate the module in the trixbox plateform. thanks a lot Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091006/b69d4103/attachment.htm

Hi, I'm planning to embed R into an application, with the following context: - This application is written in Java (and managed with maven). I plan accessing R using JRI. - This application must be installable on several plateform (linux, mac os, windows). - The R engine must embed library, some of them having native code in C or Fortran. Does this sound reasonable? I would be very

hello list i got the following pb: # xm create -f mac1/mac1.xen Using config file "mac1/mac1.xen". Error: Error creating domain: vbd: Segment not found: uname=file:/xen/mac1/swapfs but: # dmesg | grep loop loop: loaded (max 32 devices) # ls -l /xen/mac1/swapfs -rw-r--r-- 1 root root 268435457 2005-04-12 11:25 /xen/mac1/swapfs # xm list Name Id Mem(MB) CPU State

Hello All, What version of g++ is usable to compile the latest LLVM CVS snapshot on a Linux/x86-64 (AMD64) Debian/ Sid or Etch plateform? What compiler do LLVM dzevelopers use to compile LLVM on Linux/X86-64 systems? IKt seems that most versions of g++ fail to compile LLVM and that some others compile it wrongly (producing buggy code)? Why can't g++-4.1 be used? FWIW, the latest gcc

Le 11 août 08 à 22:29, Chris Lattner a écrit : > > On Aug 11, 2008, at 1:13 PM, Jonathan S. Shapiro wrote: > >> Delayed reaction. Chris's response that the emitted code relies on >> long >> long to be a 64-bit type just sunk in. >> >> Chris: is there some reason why the backend does not #include >> <stdint.h> >> and then emit [u]int64_t

Deal all, I'd like to use winbind capabilities on AIX servers (AIX 4.3.3 and AIX 5.2). In particular, I'd like to define share access based on NT group. I think I've successfully setup my Samba suite, I've entered my AIX box in my NT domain and playing with wbinfo (-t, or -a user%passwd) works fine. As told in the documentation, I've copied the WINBIND module under

Hi, Plateform: gentoo + qmail My deliver line is in a .qmail-USER file: |/var/qmail/bin/preline /usr/libexec/dovecot/deliver -f ${SENDER:-<>} -d user at domain.tld I notice that sieve implementation in Dovecot use double ">" for the envelop sender when there is a redirect command in sieve script. (i've tried preline -f, without -f ${SENDER:-<>},... but nothing

Hi, I'm writting an intrinsics for the X86 plateform that replace every 'call' instruction by a 'push ret_addr' followed by a 'jmp func_addr'. I'm doing this in the X86ISelLowering class with a custom inserter. So if I have something like this: 0x0 call foobar 0x1 ... the call will be replaced like this: 0x0 push 0x2 0x1 jmp foobar_addr 0x2 ... This works fine

Hello, I was trying to add metadata to my OGG stream (ices 2,0-kh59, icecast 2.0-kh51). It works... but the players are doing a small sound cut at metadata reception : - VLC (maybe 1/3sec cut) - JORBIS (java. maybe 1/10sec, because I can have set higher priority to sound processing and not to comment handling) (player plateform: Mac OS X 1.3.6) With VLC... I can say comments are unusable,

Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if that was all I could get. Steve -----Original Message----- From: Storer, Darren [mailto:starusers@comgate.tv] Sent: 01 July 2004 09:35 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve,

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Hi there, I have the lucky task to setup a Xen system. my plateform is a dell 2950 with 16Gb of RAM. I got as far as setting up debian sarge and xen following http://www.debian-administration.org/articles/423 and overcoming some specific difficulties (http://kmuto.jp/b.cgi/debian/d-i-2615.htm & http://wiki.ncsa.uiuc.edu/wiki/Dell_PE1950_NIC_Firmware_Workaround) But, my concern now, while

Hi, I have writen a simple JNI wrapper for speex so i could use it from Java, so far i managed to record from a raw pcm file and from the javasound inputLine and to decode the output in narrowband CBR mode and play it back still using javasound. And it sounds quite good. I haven't tried anything else because my project has no use for wideband and ultra wideband. VBR would be nice but i have

Has anyone heard any more info about the Virbiage FT201 VoIP phones? About 3 months ago I was told they were 6 weeks away, about 3 weeks ago I was told they were 2 weeks away, and now I am told they are 2 months away again! Are they EVER going to arrive? Can anyone shed some light on this? -------------- next part -------------- An HTML attachment was scrubbed... URL: