search for: jerjers

hi I recently spoke to mr McNamara on IRC, and he mentioned there was a "far better way to do realtime-stuff than the usual realtime in asterisk, and that this was GPL". He failed, however, to ever mention how this could be done, so I just wonder if someone else might know... ? roy -- Roy Sigurd Karlsbakk roy@karlsbakk.net --- In space, loud sounds, like explosions, are even

a) jerjers been doing a lot commendable work for * b) support is not mandatory, and i agree with royk it should not be withheld based on political viewpoints, that's pointlessly draconian c) choice is always good, so people should have the option of oh323 or h323, let them decide, and not limit them, un...

Oops, I had some problems and was offline unable to remind you about the conference yesterday. LISTEN to recent recordings: http://x2z.eu/astusers.htm (Flash player, will autostart) THIS WEEK: Stephan Winterberg and Stephen Boche tell us more about the new book, whick looks like a great effort. A surprise visit from Jeremy, one of the pioneers of our community who started Nufone when someone

Okay, it's 15 minutes late, but it's out, thanks very much to all the people who worked so hard this weekend to make this possible! Mark p.s. there was no 0.6.0 release.

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some problems with proper g729 codec recognition t...

In conjunction with Mark Spencer's visit to our Paris office, we'll be kicking off Asterisk TV (http://asterisktv.com) live during the weekls Asterisk Users Conference. I believe someone from Lumenvox will be back with us on the conference, now that I've had a chance to play with their speech recognition product. I think we're starting to get some great info from the user

I have found some references to the "hint" (or HINT?) variable and method in the source code, but quite a bit of Google-ing has not turned up any extensive answers as to some real-life examples of how to use this perhaps very useful tool. I understand the point of the tool, but I need to get some actual configs to look at before I think I'll figure it out. Even if my

Does anybody know how to do an international call using NuFone. I realise this isn't really the place to ask, but NuFone appears to be closed for the weekend and would like to have a try at this before tomorrow. I assumed it would be '011' for an international line followed by country code but that doesn't seem to work. I am getting: -- Executing

Hi, Can anyone confirm that if I want to do h323 proxying that I do not need codecs installed? For example if the codec being used is g723.1, I don't need the codec installed locally because there is no compression or decompression being done on my server; the incoming traffic is simply being sent out on another h323 channel (h323 in->h323 out). Is this correct? Thanks, Chetan

For outbound only, I have traditionally recommended VoipJet. They just recently has a spat of issues that seem to have resolved though. I am still using them via their east coast server and it seems to work quite well again. Cost is around 1.3 cents minute I believe. Use IAX and g711 for best quality to VoipJet. Thanks, Wiley -----Original Message----- From:

Hello I like to make calls to an h.323 device. I'm using Nuphone h323. Compiled everything okay "I Guess" When I make a connection * SIP > h323 device, the phone is ringing and then * tells me "No one available....." and disconnect Thinking this is a codec problem and check in CLI h.323 show codecs and * shows nothing. I try many combination in the h323.conf like.

Hi Is there any work being done on implementing IM/SIMPLE support for SIP on Asterisk? Like a presence server? rdgs, /Staffan Kerker

Hi, Where can I find information on H.323 for Asterisk and/or integration with Cisco CallManager in particular? <http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration> I have oh323 working on Asterisk. Since the CallManger I am working with is running 3.3.3 I cannot use SIP... Thanks, Adi

I'm trying to get two * boxes to talk.... no matter what variation I try I get No Authority Found and connection refused from 192.168.1.5 I've googled, I've site searched.... to no avail. Here is the server a configs (192.168.1.5): iax.conf [general] port=5036 bandwidth=low disallow=all allow=gsm jitterbuffer=yes tos=lowdelay register => pbx:test@192.168.2.2 [pbx] type=peer

Hi All, I am using PWLIB-1.6.6-1 and Openh323 1.13.5-1 and running a RH7.3 machine and I am unable to compile asterisk due to these errors. say.c: In function `powiedz': say.c:1633: parse error before `int' say.c:1636: `i1000E6' undeclared (first use in this function) say.c:1636: (Each undeclared identifier is reported only once say.c:1636: for each function it appears in.)

>> Follow-up post to the one I sent last week regarding bad calls between SIP >> and ISDN. >> >> On Fri, 14 May 2004, Vic Cross wrote: >> >> > To me, it looks like a variation of the SIP RTP timestamp problem (yes, my >> > 7960 is at 6.3 code), but the problem exists on the ATA-186 too and I >> > don't have any other issues with that

I have been using Asterisk 1.0-RC2 successfully with a channelized T1 circuit for quite a while now but after upgrading to 1.0.0 callerid no longer works properly. Debug output from a channel shows what actually is received through DTMF from the carrier: << [ TYPE: DTMF (1) SUBCLASS: * (42) ] [Zap/9-1] << [ TYPE: DTMF (1) SUBCLASS: 2 (50) ] [Zap/9-1] << [ TYPE: DTMF (1)

During debugging of H323 trunk side (using Jeremy Macnamara's H323 driver in ~/channels/h323) a couple of things come don't quite work as advertised... 1/ the following line in extensions.conf explicitly sets the outgoing caller ID (required in my case for downstream GK processing..) exten => _1NX.,1,SetCallerID,6400047602100 exten => _1NX.,2,Dial,H323/${EXTEN:1} what

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

G'day all, I've been googling myself silly looking for help on this one but have come up blank. I have an AVM Fritz!Card PCI, and I'm using chan_capi v 0.3.1 with * from CVS-HEAD-05/08/04-22:48:00. I can start * and make and receive calls on ISDN fine but after a few hours of * uptime, on any ISDN call I make or receive from my SIP handsets (7960 or ATA-186) I get bad audio: on the