similar to: H.323 - SIP gateway

Hi! I am in a H.323 network with a gatekeeper and some terminals. Asterisk is a gateway between this network and the SIP network. Now I can do calls from de foreign network (SIP) to the locla (H.323) but I don't know how to do the inverse. The H.323 terminals use NetMeeting, and when I try to make a call, it says that the number dialed must be registered in the gatekeeper. How can I register

Hi all! I am using ohphone (well, I am trying to) to make calls. I will make an H.323 - SIP Gateway but I don't understand the architecture of all this. What is the exact role of asterisk? It can be used as gateway, that I know, but what else can he do? Is it necessary to have ohphone to make calls or asterisk can also do that? So when the gateway it is going to be implemented how is it

Hi all! One easy question... I hope someone will answer me. I've installed asterisk with the samples. Somewhere in my network I have an H.323 Gatekeeper. What must I do to make that the gatekeeper talk with Asterisk? And I another little question... with the samples installed asterisk works ok? What must I install to see how it works? I am lost!!!!!!!!!! Please help me! See you. Mireia

Hi! When I try to make a call with ohphone, that is the message I get: Call to "06302" aborted, insufficient bandwidth Can anybody tell me a solution or a reason why this messages appears? Thanks a lot! Regards, Mireia

When I try to make a call, I have this error: dial 06302@gatekeeper -- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then kill it):

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia

Hi all!!! I am trying to stablish a interdomain relationship, but i am having serious problem making samba and nt seeing domains in another network segments. Which is the best and easier way to configure samba to stablish relashionships between domains in differents network segments? For example i have domain A in network 192.168.101.0 and domain B in network 192.168.120.0. They are both under

Hi, I want to configure * to use a gatekeeper for routing calls to H.323 endpoints. I imagine it will work like that: * (chan_h323) will query the gatekeeper where to terminate the dialed number and the gatekeeper will return the information for the h.323 gateway. after that chan_h323 will try to make the call to the gateway it has received from the gatekeeper. so instead of duplicating a

Hi, I have one FC5 server and one CentOS 4.4 server and I want to stablish a VPN between them, i have read that VPNd could be an option, anybody know where do I find the rpm for CentOS or with repository should i have to install in order to get it? Also if somebody has some other software that recommend me to stablish a VPN -- Guillermo Garron "Linux IS user friendly... It's just

Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to

Hi! I have been looking for a while for informatoin about how QoS is assured in Asterisk, but I haven't found a thing. Can someone give me some tips about that? Thanks, Best regards, Mireia

Hi! I am trying to know well asterisk. For that I would like to know the exact role for each config file. Can someone tell me what is the role of the next ones or a web where I could find this information? That will be very helpful. - alsa.conf - enum.conf - modem.conf - modules.conf - oss.conf: what is oss? - parking.conf: what is parking? - rpt.conf: what is radio repeter? - queues.conf -

hi all I'm trying to setup a dlink dph-100h phone (actually a dph-100m but with the h.323 software) with asterisk and chan_h323. AFACS, the dph-100h software can only be configured to point to a gatekeeper. I know I don't need to do this, but it's a test before I setup my Symbol Netvision phones (I don't have an access point for them now). First, this is what I have, and what

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i have a working Microsoft ISA firewall with buildin H.323 Gatekeeper.... So Far, i got registerd the asterisk on the M$ Gatekeeper... here is the h.323 configuration: ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 allow=all ; turns on all installed codecs dtmfmode=rfc2833 gatekeeper =

I've read through the archives and have picked up that * does not need a gatekeeper to talk directly with an H323 handset to send and receive calls. I'm trying to go PSTN----*-----H323 and all the examples that I can find use a gatekeeper. Are there any examples or hints for doing it without the gatekeeper? many thanks in advance Brian

Hi I am trying to stablish a relation between two different Samba domains. Each domain has two PDCs, all they are WINS servers, and the two domains are in different subnets. They are configured to use LDAP as the user/group/machine database. First, I will create the relation, being DOM1.CORP the trusted domain, and DOM2.CORP the trusting domain, so in a server of DOM1.CORP, I create a user this

Hi all! I am looking for some free software to monitoring all the calls that are being done in my network. Which telephone are connected, how long are the calls, quality of service, bandwidht,etc. If someone knows about a good one, plesea tell me. Regards, Mireia

Hi, I'm thinking of setting up Asterisk for H.323 video phone clients. Now, what is the difference between native H.323 that come with Asterisk and "Open H.323 for Asterisk" ? TIA Kuni -- Kuniyoshi Murata.........................iChat/AIM:macwebcaster English-Japanese Interpreter mailto:kuni@ej-interpreter.net Macintosh Webcast Specialist

Hi, Im trying to register to a SIP provider that told me that they only need to authenticate using IP. the following string generates response 405 register => asteriskIPaddress@SIPproviderIP:5060 doing the following is not alowed by asterisk register => @SIPproviderIP:5060 any ideas? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/

OK. So it would appear that my quest for FXO adapters unconvers more, and certainly more mature, H.323 based devices...not so many SIP devices. What would be the benefits of SIP over H.323 for a small office * server? All I need to do is bring 4 POTS lines into * with Caller ID, make outgoing local calls reliably without undo echo. FWIW, my * server is Fedora Core 1 on AMD XP2500 with 512 MB