similar to: (no subject)

Hi all, i have big problems to make a h323 call over the gatekeeper from my provider. The provider demanded following account data: H323 ID: XXX-XXX-XX-X DetinationNumer: XXXXXXXXXXX I have configured the oh323.conf following: gatekeeper=XX.XX.XXX.XXX alias=XXX-XXX-XX-X Isx the alias equal to the h323id ? And how i have to make a call with the dial app ? I have following config: exten

Hi all, i have tried to connect to a clarent gatekeeper. I have used both of h323 drivers chan_h323.so and chan_oh323.so. But no one can register to this gatekeeper. Our ip is activated on this gatekeeper. Maybe, i do wrong anything.... I have only set the "gatekeeper" option in the h323.conf or oh323.conf to the ip address from the gatekeeper. gatekeeper=x.x.x.x But no one of the

Hi all (Michael), how it is possible to get the ip address of the calling party ? (i know by using h323... but there're a few unknown segfaults...) and so i want to use oh323, but i have to get the ip from the caller to permit or deny the call with AGI. Is it possible at all ? Thanks, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing. Thomas H?ger

Hi all, i made a test scenario with two windoze machines: On the first one callgen323 is running in listening mode On the second one, callgen323 strarting 25 calls to the asterisk pbx, and the asterisk calls the first windoze machine. But after the second one make a few calls (mostly after 11 - 14) asterisk crashes with the only message : Segmentation fault. Are this to many calls for oh323

Hi all, i don't know how often someone ask for this, but i ask agian: Is it possible to use G723.1 with * or not ? I tried to use G723.1 from * over OH323 to a gatekeeper from my provider. The situation is following: Zap/analog ---> IAX -----INTERNET-----IAX--->OH323---->GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas.

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

Hi all, i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then i've installed the new chan_oh323 (0.5.6). when i try to make a call with "netmeeting" through * ( * dial out with "Dial,OH323/${EXTEN}@xx.xxx.xxx.xx" ) the call will be blocked. Before, there was chan_oh323 0.5.5 and pwlib(1.4.11) and openh323(1.11.7) installed, and it worked. Is here

Send these questions to Asterisk-Users mailing list. h323.conf ################################################## ; ; Configuration file of OpenH323 channel driver ; [general] listenAddress=W.X.Y.Z ; local ip listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=yes h245Tunnelling=yes h245inSetup=yes jitterMin=20 jitterMax=100 ipTos=none outboundMax=100

On oh323.conf I have: codec=G711U frames=20 But while connecting it gives me in log: ? 1:18.636 ? ? ? ? ?H225 Caller:8111de8 H245 ? ?Capability merge result: ? Table: ? ? G.723.1(5.3k){hw} <1> ? Set: ? ? 0: ? ? ? 0: ? ? ? ? G.723.1(5.3k){hw} <1> Which I don't have, so the connection is dropped. Any known solutions? (remote side has g711 u-Law) -- Witold Kr?cicki (adasi) adasi

I have built Asterisk on SuSe Linux 7.3 on an Ultra 2 Sparc WorkStation. I am listing the modification I had to do for the benefit of anybody else who wants to use Asterisk This workstation is equipped with one 400 MHz RISC UltraSparc II CPU, 256 MB RAM, Two 9 GB 10,000 RPM UltraSCSI Disks. I have a gatekeeper running on this machine, I had to do the following modification to build * on Sparc:

Hi, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no

Hi, what's the difference between h323 and oh323 modules? which one should I use? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030722/3d3edb73/attachment.htm

Hi, ALL: I install my oh323 channel driver following steps of http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en I works my asterisk well before install the chan_oh323.so. But after I do "make install" the oh_323, my asterisk crash and show me the following message (asterisk -vvvvvvc). Does anyone have any idea about it? What's wrong

Hello I had this codec negotiation with oh323 call. i used G723 codec and the provider had G729 as first priority. In this situation what ever number i dial i used get "No one there to answer the call". As soon as i changed my codec to G729 the call went through but had other problems, which i got away by dowloading the latest code for oh323. Has anyone seen this problem? or it is

Folks! I would appreciate if someone could send me a simple working h323 configuration file oh323.conf that is part of asterisk@home installation. I have tried to use the oh323.conf content listed on WIKI but it is just not working as my H323 endpoint ( PA168 based ATCOM Phone) cannot register. I need a working example of this file for similar phone. Seshu

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

Hi all, i have read in the * whitepaper the following: "s: The "start" extension. A call which does not have digits associated with it (for example, a loopstart analog line) begins at the "s" extension." I think this means the s extension will be execute when the phone is picked up. But when i pick up the phone the s extension will be never executed. Whats wrong

Hi, I've oh323 chan installed and working to make calls from SIP to H323 devices. The problem is can no hear sound with the H323 device. I think this is some related with codecs o nat, because the H323 have one public IP from a different subnet from the asterisk box. If I use netmeeting in gateway mode, the call can be completed and I can talk with a SIP device, but in gateway mode I can not

Hi all, if i make a call between one SIP soft-phone to an other soft phone over asterisk, i get a Segmentation fault after take up. The extension is : exten => _00.,1,Dial,OH323/${EXTEN}@<myip>|60|r This means, if a SIP client comes with 00* then dial to <myip> over H323. If the H323 client takes up, a Segmentation fault occures. But, if the extension is exten =>

Hello everybody, my * box is connected to gnugk with H323 channel. If I call from an H323 EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio start but noisy (scratch) , then became ok for callee (SIP EP) but still scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 EP and it's ok. And from now, it's also ok when H323 EP call SIP one's! No