search for: comvoz

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

...> cdr_odbc: Query FAILED Call not logged! > cdr_odbc: Connected to SQL1 > cdr_odbc: Reconnecting to dsn SQL1 > cdr_odbc: Trying Query again! > cdr_odbc: Error in Query -2 > cdr_odbc: Query FAILED Call not logged! -- Pablo Endres <epablo@comvoz.com> ComVoz Comunications

...9.10.5 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 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199

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

...ect in it's place. > > Now how do I summit the code to de CVS? Maybe just make a patch in the > makefile.. so when you compile for freetds you can use this > version. > > Just an idea. > > Please let me know > > Pablo > > > -- > Pablo Endres <epablo@comvoz.com> > ComVoz Comunications > > _______________________________________________ > Asterisk-Dev mailing list > Asterisk-Dev@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailm...

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

...econd 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 Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199

...f the cons, $$$) Any ideas on how to set this up. I know must of the features work very well in asterisk, the rest I could cook up with some AGI scripting. But the main question is, do you recomend doing this with * or should I point another way? Thanks in advance -- 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

...-- Goto (marcela,5913020,1) -- Sent into invalid extension '5913020' in context 'marcela' on SIP/10.0.0.5-08420070 -- Executing Playback("SIP/10.0.0.5-08420070", "invalid") in new stack Any Ideas why? Thanks in advance -- 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

...ome guys on the IRC said the problem could be transcoding. I also have other phones and ATA's (Cisco 186, Sipura SPA-2000, etc) using the same route but the work perfectly. My * is CVS-HEAD-05/14/04-18:04:48 running on a FC2 box. Any ideas on what this could be? -- 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

...stener:9ddacf8 H323 Awaiting TCP connections on port 1720 0:00.947 H323 Listener:9ddacf8 TCP Waiting on socket accept on ip$*:1720 0:00.985 GkMonitor:9da8818 RAS Background thread started Segmentation fault Any ideas? Thanks in advance -- Pablo Endres <epablo@comvoz.com> ComVoz Communications USA: +1 954 343 2085 Ext 199 Venezuela: +58 212 771 3100 Ext 199 Colombia: +57 1 325 6900 Ext 199

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

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

...-- Executing Hangup("SIP/test1-6e3a", "") in new stack == Spawn extension (test, h, 1) exited non-zero on 'SIP/test1-6e3a' Attached is the trace of the asterisk-oh323 library Any ideas on what the problem could be?? Thanks in advance -- Pablo Endres <epablo@comvoz.com> ComVoz Comunications -------------- next part -------------- 0:01.160 OpenH323 Wrapper H323 Created endpoint. 0:01.160 H323 Cleaner H323 Started cleaner thread 0:01.161 OpenH323 Wrapper H323 Started listener Listener[ip$*:1720] 0:01.161 OpenH323 Wrapp...

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"