similar to: Problems with H323 channels

Hello all, I`ve got an AVM c2 card instaled on my SuSE box. I?m having problems configuring its channels. I don?t know how to set up asterisk to use the CAPI channels. I don?t know how to call them. My capi.conf is as follow, [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] ;#########config de la primera interface CAPI##########333

Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk 1.4.x , but in 1.4.x the oh323 channel don't have more,

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

Hello all, I'm trying to setup a AVM C2 card. I have read the kernel requirements for this card. <M> CAPI2.0 support [*] Verbose reason code reporting (Kernel size +=7K) [*] CAPI2.0 Middleware support (EXPERIMENTAL) <M> CAPI2.0 /dev/capi support [*] CAPI2.0 filesystem support <M> CAPI2.0 capidrv interface support My problem is when I make a "make

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

Hello everybody, when I upgraded from RC1 to RC2 I didn't had any audio between my ATA286 and H323 EP (my post from 13/08/04) I checked further and discover that problem is with ATA286 who is unable to call. I always get an 404 error. Coming back to RC1 everything works fine again. I tried to modify my dtmfmode from rfc2833 to info but in change nothing. Local call to asterisk are

Hello all, Recently I bought a TDM02B digium card to conect to the PSTN. We pluged it on a Pentium IV 2,8 Ghz, Asus Motherboard, but when we try to start asterisk, the box hangs. Someone have the same card running with asterisk in a similar machine? Could you tell me your box hardware details? Thanks for your time, Ismael Gil.

Hello, I?m trying to compile the Fritz CAPI module for Debian stable, following the steps related in http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install But I always get the same error, debian-asterisk:/home/ismaelg/fritz# make (cd src.drv; make CARD=fcpci) make[1]: Entering directory `/home/ismaelg/fritz/src.drv' cc -c -DMODULE

Hello all, We have been testing Asterisk RC2 with the native H323 channel driver. We followed the instructions with the needed OpenH323 and PWLib versions and everything compiled ok. Operation of the driver seems ok, except from 2 main points: 1) Audio is passed between the two ends of the call only after the call is answered. This was not the case with previous versions of Asterisk (0.9.2

Hi, I'm on a RH 9 box, fresh install. When I try to compile channel h323, I get multiple compile errors. Can someone help? asterisk, ptlib, openh323 all are fresh from CVS. Thanks! Here's what I've done so far: All the source is in /usr/src/asterisk/ [root@localhost openh323]# ls -la /usr/src/asterisk/ total 164 drwxr-xr-x 8 root root 4096 Sep 25 00:59 .

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

Hi All, I have set up a box that will be used as follows: SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server 192.168.1.5 192.168.1.50 192.168.1.80 Asterisk is running the latest CVS and oh323 driver. The SIP phone is a Grandstream Budgetone 100. I have everything setup and running with G.711 ALAW and ULAW and i'm able to make calls through

Hi, I am using the asterisk-oh323 wrapper and I am looking to allow registration of h323 endpoints and allow Asterisk to act as a gateway. The idea is simple: H323 endpoints would register with Asterisk. They each would have their own internal extension (like SIP). If a H323 endpoint dials an outbound extension, then the h323 call gets routed to a H323 Gatekeeper which then terminates

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

Hello! I see many of you experience troubles with H323 stack. I am focusing on building H323-SIP Asterisk based softswitch with all codecs supported (including G729 and G723). I can setup Asterisk from scratch with H323 support or solve your h323 nightmare with existing asterisk system for for 100$. Contact me pls offline.

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

Hi, I'm trying to compile h323 channel driver on cvs Asterisk 1.0 but no success (I get a lot of errors - related to pwlib library). I read in docs that there is also 3rd party h323 channel driver (somehow both even share protion of code?). I wonder what are pros and cons of both drivers ? Should I try to compile native driver ? Thanks in advance, Robert.

Hi all. I can''t seem to get the h323 connection tracking configured correctly for Shorewall. I am using the Debian Shorewall 4.5.16.1 package. I am running a Debian 3.9 kernel like so: # uname -a Linux gw 3.9-1-amd64 #1 SMP Debian 3.9.8-1 x86_64 GNU/Linux My version of iptables is: # iptables -V iptables v1.4.20 If I add the following rule in the /etc/shorewall/tcrules file to

Hello everybody, First time installer and I need the lists advice. My plan is to use asterisk PBX with some hardware to terminate my calls coming from several operational gnugk gatekeepers. Do have RH9 and downloaded the latest asterisk from CVS. Compiled according instructions and is running fine. Could hardly find any info on h323 implementation untill the REAME in the channels directory.

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,