similar to: Asterisk configuration for h323 calls

hi After compiling and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI> -- Executing Dial("SIP/2.2.2.2-086f5ac0", "H323/652#150388590962@1.1.1.1|60") in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No translator path exists for channel type H323 (native 4) to 256 Mar 28

hello all, do i must must see Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface when in install TDM2424E card i think so but i can not see this in lspci is there is software tools to make sure that my motherboard have pci 2.2 and see TDM2424E see it right b4 install zaptel driver and did this kind of card need extra work than other TDM card --------------------------------- Yahoo!

Hi Guys, I need your recommendation which card to buy?The TE411P or TE406P do they have any difference?I check their brochures they are differ only when it comes to PCI slot voltages.Which card best to use?Btw,Il be using supermicro board with 3.3V and 5.5V?Will il be experiencing problem during the implementation later on if i choose the wrong PCI voltage options?Will it depends on

hi guys, after installing centos on my ml350 i found 2 choices on the grub options. CENTOS-4 i386 (2.6.9-22.ELsmp) CENTOS-4 i386-up (2.6.9-22.EL) what choice will i choose ? --------------------------------- Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi guy, has anyone come across a good and nice tutorial on openvpn ? havent seen any of it so far. i want to setup a low cost vpn network. tia --------------------------------- Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Greetings! I have the following configuration: Two PDCs with Fedora Core 4: PDC1 and PDC2. PDC1 trusts PDC2, respectively PDC2 is trusted to PDC1. I join an XP workstation to PDC2. After restart i can see both domains in the login screen domain combo box. I can logon to PDC2 , but not to PDC1, since the PDC2's /etc/passwd

Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten => 111, 1, Dial(chan_h323, H323/111@phone2) (in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the

Hi Guys, Im planning to purchase Supermicro server but I dont have idea what Motherboard and chipset model will work with TE406P.I read one post here that the chipset should be Intel E7520.Is it true?Is there other recommendation?My requirement by the way is to handle 1000 simultaneous calls. Does dualcore xeon (single cpu) and 2gig memory can handle this w/o

Hi all, Is it possible to draw line segments in a 3d space plot? I'm interested to draw the errors from an observed value to the regression plane, for a textbook example in an Intro Stats handbook for multiple regression. I used the scatterplot3d package to draw the regression plane. Many thanks, Adrian ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Adrian Dusa (adi@roda.ro) Romanian Social

I am using Nu Fone Network's h323 drivers. I can place H323 calls using following in extensions.conf file, exten => _1732.,1,Dial(H323/${EXTEN}@192.168.1.2) If I need to use h323.conf to do the same I cannot configure h323 to do the same. I get everyone is busy message and I do not see IP packets being generated by * trying to communicate to 192.168.1.2. Can someone point out what I

Hi, I have about one year of experience with Asterisk, working with ZAP (digium, junghanns) ZAPHFC, SIP and IAX. These technologies are quite clear to me, the problem is that I have no experience with H323, but now, I need to use this also. The problem that I have is very trivial, so I think that this should be a very easy question for you guys whom know how it works. All I want to do,

Dear All. I have installed Asterisk-1.0.6 on FreeBSD-5.4 RELEASE via port, so the chan_h323.so modules already included. When i try the SIP channel, asterisk works fine. In this case, i use the XLite softphone as client, i can hear the voice transfered through clearly. Asterisk is good!! :D The problem is when i try the H323 channel, the voice cannot be transferred through. I have

I try to get H323 to run, but have so far only partial success: There is a Gatekeeper GK, where asterisk connects to. The Gatekeeper sees Asterisk, and Asterisk sees the gatekeeper. From the Network on the GK, asterisk is reachable via the number 070333333. I have an extension on asterisk 6002, which is reachable. I try to call a number attached to the gatekeeper (070168177) with the

Driver chan_h323.so ---- If extension is exten => 0119823,1,dial(h323/0119823@10.10.10.1) then dial is OK: Executing Dial("SCCP/goran-00000002", "h323/0119823@10.10.10.1") in new stack ---- But if extension are something like: exten => _011xxxx,1,dial(h323/10.10.10.1/${exten:3}) exten => _011xxxx,1,dial(h323/${exten:3}@10.10.10.1) exten =>

Dears; Now using Asterisk H323 (which coming with Asterisk, I just compiled PWLIB and OPENH323), now I am placing a call from the IP Phone, the call comes to Asterisk, and it goes to the default context, but did not hear any voice of the played wave file. 1) Why Asterisk accepted the call without authentication? At least, it should be added to the h323.conf. 2) In case we found the method to

Hi list, After several tries to understand the subtil description in the h323.conf to be able to make the next scenario I was presented the following error messages by asterisk. Can somebody tell me please what I am doing wrong. Scenario: Gatekeeper (h323) --> Asterisk PBX -->(h323) Gateway Endpoints are connected to Gatekeeper. Call does come in like 999931235650087 with codec g711

Hi, I'm using D-Link DG-104SH (H323 4 port FXS gateway) with analig phone connected to it and X-Lite softphone as endpoints with * When I calling from X-Lite to analog phone it's ok When I dilaling X-Lite from analog phone, X-Lite si ringind but whei I picked up X-Lite connection drops IP of DG-104SH is 192.168.1.3, H323 ID is GW1 X-Lite number is 233 Here is * output: -- Executing

Dear list members, I'm relatively new to this list; can anyone tell me how to declare missing values once a dataset has been attached? For example here: VAR1 1 1 2 2 3 1 4 3 5 2 6 1 7 3 8 3 9 1 10 2 11 98 12 2 13 97 14 99 15 NA 16 3 I would like values 97, 98 and 99 to be treated as missing values. I read

FC2 Asterisk 1.0 When I dial a H323 dialup to an existing OKI Voip Router (BV1250), I get an EndedByRefusal yet the OKI Gateway is setup with the corrent reply ip addresses etc etc, unfortunately its an existing multiple voip router setup with g723.1 and g729a, so changing the codec on the router maybe an issue. I have compile in the h323 as per the channels/h323 setup with the listed libraries.

What is the difference between H323 and OH323 in Asterisk? I need Asterisk to have basic H.323 support so we can offer some simple H323 termination for some of our Cisco and Quintim hardware. Our upstream provider uses SIP, so I figured I'd use Asterisk as the go-between. I already setup Asterisk so it can push calls out through our providers via SIP. I just need a good/solid/very simple H323