similar to: How does Asterisk interact with an h323 gateway

Hi List, I would like create a H323 trunk from Avaya IPOFFICE to Asterisk, but i would like activate a "direct media path" for the RTP transit directly between the phone and the Asterisk. Now, - H323 Trunk is OK - RTP from the phone transit directly to Asterisk (activate "strictrtp=no" in rtp.conf, and "Allow Direct Media Path" option in Avaya Ipoffice) H323: Phone

Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If

On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi <shinacalypse at gmail.com>wrote: > Hi! > I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide > a conference bridge for an existing Avaya PBX. I have no control over the > Avaya system, but I am able to speak with the admin in charge when I need > stuff done. I am running all this in a VirtualBox

Hi, I am familiar with asterisk, though never actually tinkered with one myself ... so i don't know the full extent of its capabilities. I am facing a request to bridge a sip network and an h323 network. I would like to operate the sip with ser as the proxy and some gatekeeper on the h323 side (not required though). Actually, i have a few more points that may make it simpler - i do not need

Hi, I ve got a problem when I do this : usr/src/asterisk/channels/h323# make There are a lot of errors with ast_h323.cpp and .h. And at the end, I've got this: make ***[ast_h323.o] Error 1 In fact, I want a sample PRI/H323 gateway. Asterisk _______________ |___ ____|

Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable?

Hi, I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router with an FXO port. Asterisk talks to the router via h323 and opens a call and connects with no problem. At exactly 74 secs (timer on the phone) the call drops, and Asterisks displays this message: -- H323:29764 answered SIP/6000-9794 15:20.606 H225 Caller:80eea08 H225 Received connect PDU.

Hi there. I am trying to connect Asterisk to a local danish ip-telephony provider. But is having some difficulties. First I thougt they were related to the provider. But then i started debugging on the Asterisk (aix2 debug) When I make a call using AIX to the provider everything seems to work just fine: *CLI> -- Accepting AUTHENTICATED call from 192.168.1.150, requested format = 1024,

I have several SIP clients which connect to Asterisk. Is it possible to use a H323 client account (username/password combination) as a gateway/proxy for these SIP clients when they need to call someone on this H323 network, just like using an IAX2-gateway (registering an username/password combi). I took a look in the archives, but couldn't find anything related. But then again, I might not

hello all, i want to have ooh323 connection between asterisk and cisco. in my scenario, asterisk is gateway and cisco is gatekeeper. this is my ooh323.conf file: [general] port=1720 bindaddr=192.168.0.227 gateway=yes faststart=yes h245tunneling=yes h323id=gw10 at test.com settracelevel=10 gatekeeper=192.168.0.212 context=from-trunk disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833

Hi List, I have a Multitech H323 Gateway MVP400 box with 1 phone on port FXO 1. I have Asterisk ruuning in Fedora Core 3. Both are in the same network. But I can't figure what I have to do in Asterisk to make that box work. What files I have to configure? Can anyone help me? I will really appreaciate youu help. Luis. _________________________________________________________ Do You Yahoo!?

May be force be with you hello, We want to make H323 calls beteen asterisk and avaya(lucent) pbx. We create node-name,H.323 signaling group,trunk, but we can not make H.323 calls to asterisk. Also no warnings exist in debug. Instead of giving the IP of Asterisk ,i give my computer's IP and run SJPhone ith H.323 GUI. In this time, connection is established. SJPhone accepts H323 calls but

I have just started laying out the plans for my first project using Asterisk. I am very interested at this stage in getting much needed feedback, critiquing my approach. What are the ups and downs going to be if I develop this project as follows: -The client wants to connect some phone reps in India through a VoIP to their clients. -There will be 3 phone lines, and 1 broadband internet

HI, I have the latest RC2 of Asterisk on a RH 9 non-modified-load box. I have an Avaya IP phone that uses h323, so I am trying to compile h323 into Asterisk. Now, I downloaded pwlib and openh323 tar files and I have compiled this according to the instructions: pwlib: ./configure make opt openh323: ./configure make opt cd asterisk/channels/h323 make cd asterisk make clean make install I am

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

Hey All, Our upstream provider requires the use of H323 and after several months (6!) of having problems with OH323 I've decided it might be worth biting the bullet and getting a cisco device that can gateway up to approximately 50 calls from SIP to H323. Would a 2500 or 2600 series do the job? Once we get to the point of 50 simultaneous calls hopefully we'll be able to get something

I have posted my problem on the link below, but didn't get any answer. I am hoping someone here can help me with this issue. Here's my problem: I am using H323 to talk between Asterisk and Avaya IP Office 500. For some strange reason, when we are talking on a VoIP call, we get disconnected after 10+ minutes. We have two other Elastix box, but none of them are getting disconnected. From

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 ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --