search for: vieri

...How can I match an extension "ending with 3" (just an example but applicable to any other digit, including * or #)? exten => _ZX.3,n,... exten => _ZX.#,n,... (the above does not work) Can regular expressions be used in the standard dialplan (end with: "$")? Thanks, Vieri

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/

...aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited

...me of the LOWBATT event. My ups.conf contains: [INF-APC] driver = apcsmart port = /dev/UPS-1 desc = "UPS Monitor - ATEN port 2 - APC UPS" If I were to update to nut 2.6.5 (I can't do it for now) would I solve this issue (ie. is it a known issue?)? Thanks, Vieri

...r message. Personally I am not using wctdm24xxp but other modules such as wcte12xp and wctdm. The latter modules load fine and are compiled with pci_register_driver as expected. The only module that seems to require the deprecated function pci_module_init is wctdm24xxp. Is this normal? Thanks, Vieri ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

...oth softphones and hardphones use GSM and usually work fine (this kind of issue is not too frequent). The LAN isn't dedicated to voice but has QoS prioritizing VoIP. Could the cause of the distortion be network-related? And only on "my" side? Should I consider other causes? Thanks, Vieri

...i.ko] undefined! And "modinfo dahdi" shows that the driver was built for a 2.6.17 kernel, SMP mod_unload 586 4KSTACKS gcc-4.1 If I "modprobe -a dahdi", I get the following in dmesg: dahdi: Unknown symbol kasprintf Could this be a gcc/glibc or kernel headers issue? Thanks Vieri

...uction" server)? What can be causing the kernel panic? BTW, the last "log entry" I have in /var/log/asterisk/full is: [Nov 18 10:08:20] VERBOSE[6627] logger.c: -- dialparties.agi: Checking CW and CFB status for extension 6169 (this system runs freepbx) Help greatly appreciated. Vieri

...t; but it doesn't say anything else about it (how to connect, how to change config, how to reset the device, etc). There's absolutely nothing regarding RS-232. If someone has this or a similar device and accessed it via serial port then I'd greatly appreciate some quick tips. Thanks, Vieri

...t server2 knows that they are actually "on-line" so calls can be routed to them. How can I minimize this time lapse? Can Asterisk "notify" all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? Thanks, Vieri

..."third overlapping noise"). This is happening between Asterisk 1.4.31 and a 1.2.40. I'm wondering if there's something I can tweak in IAX2 to eliminate this artifact. Could the IAX2 jitter buffer between 1.2 and 1.4 be an issue (I believe it's enabled by default)? Thanks, Vieri

I already have ilbc installed on my system. The files are: /usr/include/ilbc/iLBC_decode.h /usr/include/ilbc/iLBC_define.h /usr/include/ilbc/iLBC_encode.h /usr/lib/libilbc.a /usr/lib/libilbc.la /usr/lib/libilbc.so -> libilbc.so.0.0.0 /usr/lib/libilbc.so.0 -> libilbc.so.0.0.0 /usr/lib/libilbc.so.0.0.0 However, if I do a "make" in asterisk-1.4.19, it will not detect that libilbc.a

Hi, My initial setup for receiving faxes worked as follows: fax call arrives on ISDN BRI connected to a BOSCH PBX, signal sent to ALCATEL PBX via PRI QSIG then finally sent to ASTERISK via PRI EUROISDN. The Asterisk server then forwarded the call to a iaxmodem and HylaFax received the data. All worked fine. Now I got rid of both BOSCH and ALCATEL in the "fax path" and it's as

Are shorewall-shell and shorewall-common required at compile time even if one only wishes to use shorewall-perl (4.0.9)? ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

...f I try a test call then nothing ever reaches peer2. However, if I remove #TEST from DUNDIVAR in dundi-outgoing and Goto(local-extensions,${EXTEN},1) in dundi-incoming then the call is established correctly. I guess the _X. pattern match is wrong? How can I match an alphanumeric string? Thanks, Vieri ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

...isdn v2 - use misdn v2 as a seperate package (disable misdn in the kernel) - use dahdi's support for misdn with asterisk 1.6 (not sure if this is true) At a first glance I think I may be better off with the latest misdn v1 but would like to know what other B410P users think about it. Thanks, Vieri

...nd happens to fall on pbx2 and will also have position 1. The worst case would be if there are, say, 10 callers in queue 1000 on pbx1 and the 11th call arrives on pbx2 with position 1. Is there a way of coherently setting up a clustered queue? Does anyone have examples/workarounds/links? Thanks! Vieri ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

...? next-server 10.215.144.7; ??? filename "/pxe/syslinux/pxelinux.0"; ? } ? host 10.215.145.94 { ??? hardware ethernet?? 00:24:54:D9:D4:2F; ??? fixed-address????? 10.215.145.94; } } option option-150 code 150 = text ; Note: the PXE client that I'm booting is 10.215.145.94. Thanks, Vieri ----- Original Message ----- From: Gene Cumm <gene.cumm at gmail.com> To: Vieri <rentorbuy at yahoo.com>; For discussion of Syslinux and tftp-hpa <syslinux at zytor.com> Cc: Sent: Tuesday, March 4, 2014 10:08 PM Subject: Re: [syslinux] Cannot chain to another PXE server on the...

Hi, I'm probably missing something trivial but I don't understand what. Asterisk is loading fine but when I connect to the console (asterisk -vr) and type "stop" I get a no such command reply: *CLI> help (...) skinny show lines Show defined Skinny lines per device soft hangup Request a hangup on a given channel unload Unload a

...ve both sources defined, can I *automatically* move listeners to the "announcements" and back to the main stream when all ogg files in dir2 have been played? If so, how can I trigger it? Or, can I merge/mix "announcements" stream with main music stream? Thanks for your ideas. Vieri