search for: duanec

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2004 Sep 30
1
Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp]
-------------- next part -------------- An embedded message was scrubbed... From: Deon Rodden <drodden@webunited.net> Subject: Re: [Asterisk-Users] Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp Date: Thu, 30 Sep 2004 09:05:39 -0400 Size: 5509 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/289c69cc/dsp.eml
2004 Jul 30
1
cisco ubr924
Hey list, Does anyone have a current working config example of a cisco ubr924 and * ? I think the 924 only supports MGCP. I want to get VoIP on this device, I was wondering if anyone has already tackled the problem, if not, I'll go in blind :) Thanks Duane Cox -------------- next part -------------- An HTML attachment was scrubbed... URL:
2004 Jul 28
4
MS SQL & Free TDS
Help! I've been using mysql for cdr storage, I need to switch to MS SQL. I must be stupid or something but I cannot figure out how to setup the cdr_tds. I have FreeTDS configured properly, but my unixodbc is not working properly either... I'd be happy with either solution, but I'm in need of assistance. Luke Catranis
2005 May 20
2
MGCP 1.0 / NCS 1.0
I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 / NCS 1.0 -- is the channel_mgcp compat with NCS -- anything I can do to make it compat?? This is what happens - below *CLI> mgcp reload Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found Use EXIT or QUIT to exit the asterisk console == MGCP Listening on 10.1.22.39:2427 == Using TOS bits 0 mgcp
2004 Jun 25
0
ATA186 (sip) in * dynamic mode
I've got my ATA's setup with DHCP, host=3Ddynamic, and turned the registration on in the ATA All is well as long as my sip.conf is configured using the phone number like so... [1231231234] type=friend host=dynamic context=main canreinvite=no username=1231231234 secret=worknow disallow=all allow=ulaw allow=alaw It appears to me that the Cisco ATA (v3.1) will register the device using the
2004 Jun 29
0
MGCP and call waiting, doesn't work.
Hey guys, can you shead some light on this? I will copy my mgcp.conf and post below, but here is the problem. I can't get call waiting to work with my MGCP device. I already have one call going, and I can hear the second call come in, I flash over to it, but all I get is a dial tone, * puts the 1st call on mute/hold, but I never get the second, and it terminates. I flash back over and pick
2004 Jul 22
2
MSSQL ODBC CDR
Hello I am having trouble setting up the cdr_odbc with MSSQL server. Has anybody done this? I have followed what little documentation I can find. Asterisk errors on a connection error, but tcpdump doesn't even show an attempt. Does anybody have some examples for MSSQL ODBC CDR? I've downloaded unixODBC and the FreeTDS project, recompiled asterisk.... Thanks Duane Cox --------------
2004 Jul 27
0
Touch Tones not working after call is made
I'm running the latest CVS of asterisk. If I make a call to an automated attendant system that has a directory tree (don't we all hate those), I can't get my touch tones to be recognized at the far end. In debug mode, asterisk sees what key is pressed, but the far end system doesn't recognize it. Any ideas what to check? This is MGCP channel. Thanks Duane Cox --------------
2004 Sep 23
0
asterisk-1.0.0 woes
Is this a new bug, or am I doing something wrong here... ? Same config files as I have always been using, maybe there is a config file change needed? I have been running CVS for a while and just upgraded to 1.0.0 Now I get an RTP error with MGCP, MGCP dies afterwards. Sep 23 11:28:33 WARNING[98310]: rtp.c:711 ast_rtp_offered_from_local: rtp structure is null -- MGCP
2004 Dec 06
0
strange caller id and caller name with SIP and ATA186
Hello list: I'm not sure what is going on, but I am using: asterisk -stable cvs (november cvs download) SIP channel Cisco ATA186 Zaptel 4-port PRI for PSTN Caller ID is enabled on the cisco ATA and seems to work fine. We do not get caller name at this time over the PRI ... Caller name works fine VoIP to VoIP and for PSTN calls, it displays the calling number like caller ID. But for some