similar to: Call Quality Detail Record

hello list, last night i upgraded my asterisk box from 1.0.5 to 1.0.6 and my music on hold did not work anymore. my setup is ISDN (wct1xxp)->SIP (Audiocodes mp124) and reverse. the system refuses to activate music on hold resource... i returned to 1.0.5 and it works fine again... i'm i missing something? i can really make use of 1.0.6 bug-fixes and i'm sorry i can't use it :((

Hello, Are there any T1/E1 PCMCIA cards available on the market supported by zaptel drivers and asterisk ? I need to make some demos at my clients with asterisk and it's a pain to move around with a midi-tower computer just for that. Thanks, Calin.

Hello list, Is it possible to implement an application that satisfies the following scenario using agi and php? - user picks up phone - he wants to redirect all his calls to the cellphone - he dials *400 for example and all the calls addressed to him are diverted - he comes back to office next morning, dials *500 and his redirection is gone Note: my extensions.conf is entirely rebuilt at 5

Hello people, It seems that my * does not react to tos=<whatever> field in iax.conf. I am using latest CVS HEAD code. Can anybody help me with this issue? ps: if i go to chan_iax2.c and modify the initial definition of tos variable, it works fine marking packets with the value specified there: static int tos=16; if i put random text in iax.conf's tos=, chan_iax2 refuses to load

Hello everybody! I am having the following problem and since I am a beginner in playing with asterisk, i can't solve it: I am trying to integrate my existing H.323 network in real world telephony by ISDN cards. The problem is that i DON'T want to change all e164 numbers in my h.323 network and my ISDN provider doesn't accept those identities (CIDs). So, i have to spoof the outgoing

Hi everyone! I am using OpenSuse 10.3 with desktop effect compiz-fusion and wine 0.9.57! I have installed counter strike 1.6 with wine and everything went fine. When i run the game with no desktop effect works fine but when i run with desktop effects it doesn`t start and it tells me: Code: calin at DarkStar:~/.wine/drive_c/Program Files/Valve> wine hl.exe -game cstrike

Hi, my name is Calin and I''m new to linux, but I guess its the right place to ask this: what do I set on a linux RH9 box with 2.4.24 kernel to route a 10 machine private network (192.168.x.x) by 3 limited bandwidth, public IPs (193.231.x.x). The network uses a switch, the linux box has 1 ethernet card, the link is available trough a wireles ethernet bridge from my ISP. I begun to read

hello i want to use SIPGetHeader application in asterisk-1.0.9. Jul 2 00:04:33 WARNING[19575]: pbx.c:1293 pbx_extension_helper: No application 'SIPGetHeader' for extension (default, 2000, 1) Any one using this __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

I'm tring to use the function named sipgetheader in asterisk, but I downloaded the asterisk version 1.0.3 in which this function doesn't appear. What the simplier solution to my problem? May I download something else? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050113/761194e1/attachment.htm

I'm tying to use park and announce for call park on Asterisk 1.4.2 but I'm having trouble getting it to work properly. This use to work with an older version of Asterisk. A telephone on the PSTN calls an IP phone. The IP phone is assigned extension 3-8396. 3-8396 answers the call and attempts to perform a blind transfer to x700, the parking lot number. The transfer gets to Asterisk,

Hi kalinix Thanks fro your correct info. It now works as expected. I am really happy about your rules. Thank you very much indunil On 3/24/07, kalinix <calin.kalinix.cosma at gmail.com> wrote: > > On Sat, 2007-03-24 at 12:06 +0530, Indunil Jayasooriya wrote: > > > > Hi List, > > > > I want to bind an ip address to a username with squid by using squid >

Hello, I've been playing around with * for quite a while now, and have run into a problem that I just cannot seem to figure out. When using * and any IAX client (I have tested with GnoPhone and both clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the connection. What I'm running is a P3-1Ghz machine with 512mb ram for a server. The other end has been

I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1 (see config below) and with a bit of messing about using sample config, have been able to make the test call to device 1000, and also through to the IAX test number at Digium. However, operation is extremely flaky - I cannot reliably startup and use the system on a regular basis. I have several problems listed below

I''m trying to figure out the best way to display some summary information in my app (an article submission database for writers), and I''m not quite sure how to get ActiveRecord to give me what I want. Here''s what the relevant part of my data model looks like: ----------------------- class Market < ActiveRecord::Base has_many :submissions has_many :articles,

Can anyone share any links regarding packages to do Call Detail Record (CDR) analysis from the CDR Master file? Login-distance reconciliation, billback, and data presentation are three primary areas of interest. Thanks in advance for your help! --Nick -- Nick Eggleston Consultant Data Communications Consulting, Inc. 6320 Rucker Road, Suite E Indianapolis, IN 46220 317/726-0295 x18

Hi all, Now i want to store cdr (call detail record) to Oracle database but i don't know how to do .Can anyone help me ? Thanks and best regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101109/2fac8686/attachment.htm

We've had a regular Park function in the past but recently I found the ParkAndAnnounce() application and I love the idea behind it. Here's a snip from the wiki (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce) so that we're all talking the same language: || ParkAndAnnounce(announce:template|timeout|dial|return_context) || || Park a call into the

This is almost certainly not an Asterisk-specific posting, but due to my inability to find a VoIP-focused Cisco list, I'll post here in the hopes of finding a more diverse user community. I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and have been experiencing situations where the phone locks up. "Locks up" means that the bottom part of the screen

I just built it and now can no longer get incoming or outgoing service. It was working with CVS Head prior to my "downgrade". Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hi, I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch. When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk : In INVITE : Vm Phone Number ( to route the call ) In To : Person who has been called ! In From : Person who was calling ! Of course, I need to send the call into the "Called User" Mailbox (Thus To SIP header) ! So