similar to: Meetme + Billing

Hi, I've installed dovecot-0.99.10.5 using the rpm from Dag Wieers, but when I start it all I get is nothing (but none of the processes are running). I checked the config file and set it up (I'm trying to use it with mysql support, but it doesn't work with traditional config) Any ideas -- Pablo Endres <epablo at comvoz.com> ComVoz Communications USA: +1 954 343-2085

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

I agree, any platform suffers when it is extremely difficult to implement. What we need is an interface that does everything we need and shows what asterisk is capable of, a lot of features will go unused because you might not know the exist unless you hunt them down in the source or conf files. I trained on an Avaya INDeX switch it had a complex console but was laid out in a structured way a

Hi, Has anyone made a successfull instalation of cdr_odbc?? I've install unixODBC-2.2.8 (made my own RPM) and then built the module. I'm trying to send the cdrs to a M$ SQL server. The sql connection works because I can do any query via isql. When I do the calls I get the following output on the asterisk console: -- Executing Hangup("SIP/test1-a5e1", "") in new

Hi people. I have 2 cisco 7940, the first is behind a NAT and the second is behind 2 NATs (The one for the LAN and one for Wireless). The two phones worked perfectly until Friday. For some reason the second one stoped registering (chan_sip) and the today none would work. Any ideas on what could be happening? Thanks in advance. Pablo -- Pablo Endres <epablo@comvoz.com> ComVoz

Hi, This post may be a little off topic but I've seen lots of good ideas com from the list, so here it goes: I'm in the need of replacing Cisco Call Manager 3.2 that I have working as my primary GK. I thought I could replace it with gnugk, but it lackes some of the functionality that I need: 1.) Route retrying. If a call placed on a route fails for some reason try other routes 2.)

Hi, I've got a strange behavior on one of my new extentions. This is what I got: extensions.conf with to #includes: extensions-manual.conf & extensions-db.conf I have around 50 users and 100 contexts in these files. I reacently included one: (created from db) [marcela] include => Common include => VoiceMail include => marcela-extensions [marcela-extensions] ;; =======

Hi people. I've been having some audio problems with some of my cisco 7940 phones using firmware 7.1. The sound gets gargoled up, I can't understand much of what is said (listing on the IP phone). My setup is the following: 7940 - * - Cisco 2621(GW) - T1 I'm using SIP and g729 for all stages of the communication. I was using ulaw for the leg between * and the Gw, but some guys on

Hi guys, I've been trying to update my chan_oh323 from 6.1 to 6.3b. I built asterisk from cvs-head on the date Micheal said he made it compatible, pwlib-1.6.6 and openh323-1.13.5 (both with nothing more than the ./configure, make, well aplied patch on openh323) When I start * with my normal config I get this: [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing

Hi, i'm having another problem I can't work out - make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target `ccflags'. Stop. make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.1/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 g++ -Wall

All, I have access to a Cisco AS5300 w/ 4 T-1's and a Cisco 3600 with no boards. I was wondering if it would be possible some how to Have one of these Ciscos in-between our sip phones and the asterisk server so that we could use G729 Codec. Sip Phones (7960's & ATA's) via G729 -->Cisco Gateway-->Asterisk via G711. Any ideas? Has anybody done such an implementation or know

for inbound calls, i can register context = from-sipphone register => 1747xxxxxxx:passwd@proxy01.sipphone.com but how do i configure to make outbound calls to them? exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1) .... [dial-sipphone] ; ; SIP to sipphone.com ; exten => _X.,1,Dial(SIP/${EXTEN}@??????) ^^^^^^

Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message ----- From: "Panny Malialis"

http://asterisk.bkw.org/diff/cdr_odbc.diff Now this should fix it the proper way. bkw PS: Thanks for the info klasstek! > -----Original Message----- > From: asterisk-dev-admin@lists.digium.com [mailto:asterisk-dev- > admin@lists.digium.com] On Behalf Of Pablo Endres > Sent: Wednesday, May 19, 2004 3:30 PM > To: asterisk-dev@lists.digium.com > Subject: RE: [Asterisk-Dev]

Does anyone know about a good documentation on the oh323.conf file. There are a bunch of paramters that I don't understand. Thanks in advance Pablo -- Pablo Endres <epablo@comvoz.com> ComVoz Comunications

Hello, I've been trying to send traffic to a Cisco Call Manager 3.2, but with no luck. Here's whats happening: * Call gets to CCM * Call gets to the gateway * Rings a couple times on destiny * Call gets hungup. On the CCM I get the following error: MediaManager - ERROR wait_AuConnectErrorInd On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not available) On asterisk:

Hi, I have an Asterisk CVS-HEAD-08/29/05-13:21:43 built on a Redhat 9. I have problems to receive faxes with spandsp-0.0.2pre11 and libtiff-3.5.7-11. I'm trying with a fax machine Panasonic KX-FT25, 14.4Kbps modem speed. I have tested sending a Fax document from Asterisk to the fax machine, it is working fine, but when I try to receive with asterisk, I receive transmission error on the fax

> The documentation mentions that the Digium channels can be split into some > voice channels and the remainder of the channels used for routing IP > traffic. > > Does any one have this in use in conjunction with Asterisk? Does it work > well? Would you recommend it for a production server? > > Obviously, if this works, this makes for a cost effective platform where

What types of ISDN BRI cards work well in Europe (Guadeloupe, Martinique and France) ? For example: AVM C2 or AVM C4 or eicon Diva server 4 BRI? Any others? Which driver is appropriate? Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -------------- next part

Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-us&page=byoa/plus&ST=1