similar to: Replacing a Cisco Call Manager

Hi, I'm trying to detect and or log the duration a a conference (Meetme). I need it in order to do some billing for theses services. Any ideas on how to do it? I googled around but found nothing. Thanks in advance epablo -- Pablo Endres <epablo@comvoz.com> ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199

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

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, 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

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]

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

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:

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, 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

Dear Mailinglist-User currently we`re working with an IP-PBX, based on Asterisk, with SIP, H.323 and ISDN-Capabilities. SIP and ISDN works fine, but H.323 not. In our first test, we started to connect Asterisk to an Cisco IOS-Gatekeeper with the "chan_oh323" (version 0.6.5). We successfully tested in/egress calls without any problems. But when we started to connect our Asterisk

Hi, After a few extra days of hard work, debugging, and many coffees, I am pround to announce that Asterisk::LCR has been released on CPAN. Asterisk::LCR is an open-source, Perl-based collection of tools to help you manage efficiently multiple VoIP providers with your Asterisk installation. It is capable of importing providers rates from multiple providers, comparing these rates, and

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}@??????) ^^^^^^

hello sir im working on text classification using java and r programming i start with exporting a document term matrix (tdm) from my java programme using corpus and now i try to apply knn algorithem using the matrix and r but i cant do that any help sir here my data here my script https://mega.nz/#!Q6J2ibAA!4PadiOKbP7rLodyiRrVsdKl-D2ZP7LYm0gaz94uBmF8 itry to post put icant whay!!

hello there, I am facing an issue with misd/misdnuser/lcr in the system I am running debian 7 and I managed to install from git misdn/misdnuser but in lcr I am getting: chan_lcr.c: In function 'load_module': chan_lcr.c:3520:24: warning: assignment makes pointer from integer without a cast [enabled by default] make[2]: *** [chan_lcr.po] Error 1 make[2]: Leaving directory

Hi there, I am trying to register asterisk to a Lucent's MVAM GK. It is not registering to the GK with both h323 channels(chan_h323 and oh323). The problem is that if I set the GK(in asterisk) through the GK's IP, the GK answers with GCF without it's GK ID, and after that it does not answer to the RRQ, because the RRQ message has no GK ID. And when I set the GK through