search for: noh245tunnel

...Linux I can make H323 call without any problem from X-Pro and from X-lite dead-air both end. My default h323.conf =============== [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=g723 allow=alaw allow=ulaw allow=gsm gatekeeper = a.b.c.d AllowGKRouted = yes noFastStart = no noH245Tunneling = no noSilenceSuppression = no Modified h323.conf ============== [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=g723 allow=alaw allow=ulaw allow=gsm gatekeeper = a.b.c.d AllowGKRouted = yes noFastStart = yes noH245Tunneling = yes noSilenceSuppression = no I can to he...

...ir. The configuration is GW to GW. This is my configuration from h323.conf: [general] port=1720 bindaddr=my.ipaddr dtmfmode=rfc2833 gatekeeper=DISABLE AllowGKRouted=no context=default [termination] type=peer host=some.ipaddr noFastStart=no noH245Tunneling=no noSilenceSuppression=yes progress_setup=3 progress_alert=8 progress_audio=yes disallow=all allow=ulaw Any Suggestion is welcome. Thanks! __________________________ Laurent TOSTAIN

Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance.

Hello every one, I have got a H323 gatekeeper for testing. The informations are something like this: account code: test01 gk ip address:192.168.10.12 I don't know how to set it in the h323.conf or oh323.conf, I have tried it for almost one day but I always got the error. Help me please. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...when i try to dial through h323, it segfaults.<BR>I'm using X-Lite as SIP client, i have set up my h323.conf:<BR>[general]<BR>port = 1721<BR>bindaddr = 0.0.0.0<BR>tos = lowdelay<BR>dtmfmode = rfc2833<BR>context = Out<BR>noFastStart = yes<BR>noH245Tunneling = no<BR>gatekeeper = 01.23.45.67<BR>AllowGKRouted = no<BR>allow = all</P> <P>[NET1-BG]<BR>type=h323<BR>prefix=0<BR>context=Inc<BR><BR>and in my extensions.conf there's a line like this:<BR>exten =&gt; _0.,1,Dial(H323/08898...

...working. Anyone out there have any clues they can share? ; ; Configuration file of OpenH323 channel driver ; [general] ; ; Port to listen to port=1720 bindaddr=XX.XX.139.59 gatekeeper=XX.XX.131.112 disallow=all ;allow=g729 allow=ulaw context=tbr-extn UserByAlias=no noFastStart=yes dtmfcodec=101 noH245Tunneling=yes ; ; Specify alias(es) of this host. ; It may be used multiple times. ; alias=tbrmtp alias=689 ; ; Set the context of H323 calls ; [tbrmtp] type=h323 context=tbr-extn prefix=689,688,687,686,685,684,683,682

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

...H.323 GK. I have installed asterisk using the ports. It seems to be running OK, but when i try to dial through h323, it segfaults I'm using X-Lite as SIP client, i have set up my h323.conf [general port = 1721 bindaddr = 0.0.0.0 tos = lowdelaydtmfmode = rfc2833 context = Out noFastStart = yes noH245Tunneling = no gatekeeper = 01.23.45.67 AllowGKRouted = no allow = all [NET1-BG] type=h323 prefix=0 context=Inc and in my extensions.conf there's a line like this: exten = _0.,1,Dial(H323/0889811777,20) which i think should dial the phone entered, no matter what number is dialed by the client (if...

...0 bindaddr = 0.0.0.0 tos=lowdelay ; amaflags=billing ; disallow=all ; turns on all installed codecs ;disallow=g723.1 ; Hm... Proprietary, don't use it... allow=gsm ; Always allow GSM, it's cool :) ;allow=ulaw allow=alaw ;allow=g729 ; noFastStart=yes noH245Tunneling=yes noSilenceSuppression=yes ; jitter=20 ; dtmfmode=inband ; gatekeeper = 192.168.0.207 ; AllowGKRouted = yes ; context=outgoing ; [CAC-IP] ;our computer. type=h323 prefix=9,7,*,8 context=outgoing ; -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lis...

Dear friends, I have compiled and installed h.323 in my asterisk. And I have a service from a H.323 VoIP provider who give me user, password and gatekeeper IP address. All configured. But when I start my asterisk i receive the following error and h.323 calls can not be making and/or receiving. [chan_h323.so]=> (The NuFone Network's Open H.323 Channel Driver) == Parsing

...303 * chan_zap: Implement confirmation for PRI * chan_mgcp: Re-listen after MGCP reload * chan_zap: Add udev support * chan_zap: new option to zaptel.conf, prilocaldialplan. This is used only for setting the caller number's PRI dial plan. * chan_h323: implement the setting of noFastStart and noH245Tunneling. NEW APPLICATIONS - none- Upgrade your Asterisk now and test all these new functions! * http://www.asterisk.org/index.php?menu=download *** Reporting bugs in the bug tracker ------------------------------------- Before you report bugs in the bug tracker, you have to read the instructions, you...