similar to: REMOTE CRASH FIX

Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call

We have decided to allow our tech's to do support for non-clients of voicemeup.com You can head to http://support.voicemeup.com/ and one will be in touch 8 to 6pm business hours. 3 levels of support are offered for Asterisk/compiling Trixbox , Ivr's etc. -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -------------- next part

I'm trying to get an IAX trunk to failover to a local trunk it the trunk is down. This is what I've been working on: [macro-forward1]; exten => s,1,Dial(IAX2/192.168.1.1/${ARG1},20) exten => s,2,Goto(call-${DIALSTATUS},1) exten => s-CONGESTION,1,Dial(LOCAL/${ARG2},20) exten => s-CHANUNAVAIL,1,Dial(LOCAL/${ARG2},20 ;end macro-forward1 exten =>

Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___________________________________________________________ Yahoo! Answers - Got

at http://wiki.openpbx.org/tiki-index.php?page=Easy+route+to+building+OpenPBX.org there is the following adduser command: adduser --no-create-home --ingroup openpbx --disabled-password --disabled-login openpbx This does NOT seem to be the right format for Centos. So far, using man, I have come up with; adduser -M -g openpbx What else do I need?

I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s -- Abra?os Luis Claudio Mobile + 55 21 9215 2888 Mobile +55 15 9141 8402 Office +55 15 2102 5859 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/b8593cb1/attachment.htm

Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? ? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006

Kinda confused by this... I have a Cisco 7960 configured with a couple SIP extensions configured on the phone. Just trying to dial one extension from the other on the same phone, but when I do, I get: -- Remote UNIX connection -- Executing Dial("SIP/2001-ffd4", "SIP/2002") in new stack -- Called 2002 -- Got SIP response 486 "Busy here" back

Hello, I created a new field named pre_dst of type varchar(80) exactly like dst field in cdr table. In the dialplan I put: exten => _7.,1,Set(CDR(pre_dst)=${EXTEN:1}) and when I call, all goes fine except that pre_dst has always NULL value in cdr. Do you know why? Is something wrong I did? I know that original fields in cdr are only readable, but in this cas pre_dst is one I created

Hello, What do you think of this project www.openpbx.org ? Something like ser and openser ! Kinds Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

Ok here is a real geek question, I building my own linux kernel for my asterisk system and came across the kernel setting for the timer frequency. I have one of 3 hardcode choices 100Hz, 250 Hz and 1000Hz. From what I understand the default Freq was changed from 100Hz in kernel 2.4 to 1000Hz (1KHz) in kernel 2.6. Timing is a BIG issue in asterisk with all the TDM and zap channel stuff.

This is slightly OT in that it isn't specifically *-related, but I was wondering what the members of the * user community felt about these two subjects. I've been perusing the OpenPBX.org mail list and the current hot topic is the fact that their project has come to a grinding halt. They are concerned that they don't have enough people working on their project. They feel that * has

Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ"): The main points listed on Asterisk's "CONS" that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT

Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate

Hey, i'm trying to patch asterisk to work with ncs. I get this error: root@fox:/Kit/openpbx/asterisk-1.0.0# patch -p1 <../conf_patches/ncs_dtmf_combined_v03.diff can't find file to patch at input line 3 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -------------------------- |--- channels/chan_mgcp.c 16 Jul 2004 21:22:55 -0000 1.60 |+++

Hey, i'm trying to patch asterisk to work with ncs. I get this error: root@fox:/Kit/openpbx/asterisk-1.0.0# patch -p1 <../conf_patches/ncs_dtmf_combined_v03.diff can't find file to patch at input line 3 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -------------------------- |--- channels/chan_mgcp.c 16 Jul 2004 21:22:55 -0000 1.60 |+++

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator

I am interested in getting Openpbx up on Centos. The install information is at http://wiki.openpbx.org/tiki-index.php?page=Getting+Started And it is quite a bit beyond my current skills, so I am looking for some one that has done it and can help step me through it. I want Openpbx over AsteriskNow or Trixbox to get T.38 support.

I've got a brand new Eicon Diva Server BRI card and I want to configure it with Asterisk. I managed to get asterisk and zaptel to compile and install, I've compiled and installed the drivers for the Diva card and now I need to compile and install the chan_driver for chan_capi. Unfortunately this fails miserably. I get the following messages: I'm using: Kernel 2.6.16.37.4,