search for: alperin

...ake install. Also, I have the kernel sources, and a symlink to /lib/modules/.... Also, I tried the make install-udev, since there was no zap device on /dev/zap but nothing. The error is that when I run modprobe the result is FATAL NO ZAPTEL MODULE FOUND. Any clue about this? Thanks Carlos Alperin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070220/7c3e069d/attachment.htm

...module but it never worked for me. Any other experience? I want to track the use of my PRI's and trunks using graphical as MRTG does each 5 minute, day, week & Year. But the option of the 5 Minutes I don't think is usefull, We need something more realtime. Thanks, Carlos Alperin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050705/b1e1456d/attachment.htm

...c, still I have no audio however the call is not destroyed immediately as before. I'm going to put a second Granstream box, and findout if between two boxes this happen too. I cannot believe that we cannot do 2 g726 on the same box at one time. Carlos -----Original Message----- From: Carlos Alperin [mailto:calperin@senecacom.net] Sent: Wednesday, December 06, 2006 11:16 AM To: 'asterisk-users@lists.digium.com' Subject: FW: G.726 on Asterisk 1.4.0 Importance: High This is what I found today googling on the Web, also I post it now in order to save time to others: The G726-32 codec:...

...out, just no outbound audio. Is their any difference in the inbound & outbound audio streams in Asterisk that could cause it, e.g., different ports, protocols, connection/discovery methods, etc? Thanks, Ross ---------- Original Message ---------------------------------- From: "Carlos Alperin" <calperin@senecacom.net> Date: Sat, 9 Jul 2005 19:33:31 -0400 >Some way you should have a udp filter between you box and your phones. I see >that before. > >Can you call those phones? > >Carlos Alperin > >-----Original Message----- >From: asterisk-users-boun...

...-level uhci_hcd:usb2, uhci_hcd:usb5 193: 0 IO-APIC-level uhci_hcd:usb3 201: 850522 IO-APIC-level t1xxp NMI: 0 LOC: 249006 ERR: 0 MIS: 0 Is anything wrong about this? I never get 100.00 % most of the times. Thanks, Carlos Alperin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060116/4c590775/attachment.htm

I finished to try to install Asterisk Now 1.4.0 on an AMD 3800 dual processor machine. The install lookups on the search for the Sata drive, since however it loads the sata_sil driver it doesn't work. Did someone knows what version of Linux is using on Asterisk Now? Thanks, Carlos Alperin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061220/112d071b/attachment.htm

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven

Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan,

Just installed Asterisk 1.2 on a brand new clean machine running RedHat 9.0. I have a TDM400 card inside. When I boot, the card seems dead. When I do: modprobe wctdm modprobe Zaptel the lights come on and all seems fine, until I reboot that is... After a reboot I have to repeat the modprobe. I shouldn't have to do a modprobe every re-boot should I? How do you get the drivers to load

...eir makefile and compile everything again? When I tried to install everything from svn, I got messages like my zaptel is too old, and also the same for libpri. By the way, if you download from trunk, everything is the same download following the web page instructions. Thanks, Carlos Alperin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060523/7c850b1a/attachment.htm

...more an Asterisk problem, since the modification was made for Sipura boxes, but even when other phones, after the 1.2.6 we have fast busy problems on those devices with that code) So at this point. if someone has any other experience that want to share I'll appretiate it. Thanks Carlos Alperin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061206/a094e615/attachment.htm

Hy, I have a network with WIFI communication and VHF/ HF channels. I have integrated asterisk in the network using SIP, ZAP and IAX2 channels for WIFI communications, but I don't Know How I could integrate the VHF/ HF channels. I have heard speaking about app_rpt project, but I don't Know very much about this. Could I integrate VHF/ HF channels with this application? if the answer is

Just installed fc5, installed correct kernel source, and trying to compile zaptel-1.2. Changed the link in /lib/modules/2.6.15-1.2054_FC5 to point to /usr/src/redhat/SOURCES. Like: lrwxrwxrwx 1 root root 23 May 12 15:21 build -> /usr/src/redhat/SOURCES A 'make install' still complains with: make -C /lib/modules/2.6.15-1.2054_FC5/build SUBDIRS=/usr/src/zaptel-1.2 modules make[1]:

Dear All, I need to bind two different ports at the same time for SIP. 5060 and another port number. Is it possible ? It would be something like port=5060,5062 Isamar

Hello Everyone, This is my first post, and this is my problem :-). I have a asterisk@home, work excellent (only internal users), but i need outbound calls. One person give me an access to his "Cisco 5300 Media Gateway", he give me a dial rule and the router ip address. I've created a SIP Trunk, and a outbound routing, with all the info (the rare thing, the

Hello, i have setup exten => 555,1,MusicOnHold(default) i can hear the music, so far so good. But when i hold an incoming call by pressing the HOLD-key on my snom telephone - nothing happens. No output at CLI that the MOH gets played. When debugging SIP on asterisk, in the moment i press the HOLD-key i can see some SIP-INVITE messages from the phones that holds the call to the other

Hi all! I'm using Asterisk as a SIP server with 2 ip phones and it works great, the only think that I can?t make it wotk is: 1. I leave the messages but when I receive them in my mailbox , and open them I hear only noises -------voicemail.conf------- [general] format=wav49 servermail=asterisk attach=yes maxmessage=180 pbxskip=yes fromstring=The Asterisk PBX [default] 123 =>

Asterisk/phones work perfectly within our LAN. Asterisk box has a public IP - no NAT or firewalls. When I take the phones to a remote location (again, public IP - no NAT or firewalls that I know of) the outgoing audio does not work. I can hear the other party, my phones ring, I can dial out, etc, but the other party cannot hear me (even if I dial #'s, etc). Any ideas? Thanks, Ross

If I'm doing a time include in extensions.conf, do I want 04:00-23:00 and 23:00-04:00 or 04:00-22:59 amd 23:00-03:59? I want to make sure that at no time are both or neither included. In other words, does the second time go to HH:MM:00 or HH:MM:59? Thanks.

I need to find someone to work with me in the Grand Rapids Michigan Area. Someone good with Linux and Asterisk would be ideal. Please get me contact info if you are interested. Thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/315eff3f/attachment.htm