similar to: Diagnosing codecs

For the benefit of everyone, having H323 Configuration problem due to H245 Tunnel, check the h323 Config embeded at the end. Comment the offending line as under: ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; -----Original Message----- From: Tola Ogunsan [mailto:tolaniye@hotmail.com] Sent: Wednesday, May 25, 2005 1:03 PM To: Kanuri, Seshu (Company IT) Subject: RE: oh323 problems

Hi, I've oh323 chan installed and working to make calls from SIP to H323 devices. The problem is can no hear sound with the H323 device. I think this is some related with codecs o nat, because the H323 have one public IP from a different subnet from the asterisk box. If I use netmeeting in gateway mode, the call can be completed and I can talk with a SIP device, but in gateway mode I can not

Hello all, It's me again. I would like play with calls between a H.323 client and a SIP client through * inside my LAN. For that, on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk; on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I register into * with a username, no password. The 3 files oh323.conf, sip.conf, extensions.conf are in attachment. In the same

Hi, I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular phone. Asterisk configuration is listed below. When I attempt to place a H.323 call, I receive the following errors: - Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20") in new stack Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path exists

Hi all, I have found the solution for getting Zultys handsets to accept paging messages. We need an app to send RTP packets with a codec of G711U to send to port 3771 on multicast dest of 224.0.0.1. There isn't any need to use SDP or SCCP packets to inform the phone to "listen" for those packets, they do it by default. Finally we may be able to get a solution going for Zultys

On oh323.conf I have: codec=G711U frames=20 But while connecting it gives me in log: ? 1:18.636 ? ? ? ? ?H225 Caller:8111de8 H245 ? ?Capability merge result: ? Table: ? ? G.723.1(5.3k){hw} <1> ? Set: ? ? 0: ? ? ? 0: ? ? ? ? G.723.1(5.3k){hw} <1> Which I don't have, so the connection is dropped. Any known solutions? (remote side has g711 u-Law) -- Witold Kr?cicki (adasi) adasi

I want to use Asterisk registering itself to a GK. SIP phones are registered to Asterisk H323 are registered to the GK I want to: 1. make calls from SIP (Asterisk) <--> H323 (GK) 2. use Meetme to make a conference call for both types of phones I got on the GK, login and password, IP of GK, and codex g711u How to set-up h323.conf and extensions.conf for that? (I am using ooh323c)

Hi all, i have big problems to make a h323 call over the gatekeeper from my provider. The provider demanded following account data: H323 ID: XXX-XXX-XX-X DetinationNumer: XXXXXXXXXXX I have configured the oh323.conf following: gatekeeper=XX.XX.XXX.XXX alias=XXX-XXX-XX-X Isx the alias equal to the h323id ? And how i have to make a call with the dial app ? I have following config: exten

Hello again. I'm stll struggling trying to terminate calls from SIP through Asterisk and throught my H323 gateways... Basically the call is accepted by GnuGK but then dropped with *reason = unreachableDestination <<null>>* I did a *debug trc 10* on GnuGK and looked at the sessions... one from X-Lite through Asterisk... and one from OpenPhone... The one from OpenPhone works

I wonder if anyone can help me? Am trying to terminate to H323 Vegastream. I'm using OH323 with little success. I can dial out and answer but voip end just keepings ringing and ringing. Thanks for any help. Neil Config file: [general] listenAddress=ALL listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no jitterMax=100

All, I've fixed my problem getting Asterisk ooh323 channel to stay registered with my Cisco IOPS gatekeeper, now I need to get dialling working. I have the following: [Asterisk with ooh323] ----h323---- [Cisco IOS GK] ----h323---- [Radio system OpenH323] 192.168.1.5 192.168.1.6 192.168.1.7 the Asterisk box has numbers

Hello, Thank you for the job well-done. I installed the chan_h323 of the asterisk-1.2.7.1 and with lib pwlib-v1_10_0-src-tar.gz and openh323-v1_18_0-src-tar.gz and I used licensed g729 from digium. However, I am having a very funny behavour. 1. If I send a call on its ringing at the called side but the caller didn't get the ringing tone. 2. if the called picks up the phone, I am

I have built Asterisk on SuSe Linux 7.3 on an Ultra 2 Sparc WorkStation. I am listing the modification I had to do for the benefit of anybody else who wants to use Asterisk This workstation is equipped with one 400 MHz RISC UltraSparc II CPU, 256 MB RAM, Two 9 GB 10,000 RPM UltraSCSI Disks. I have a gatekeeper running on this machine, I had to do the following modification to build * on Sparc:

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

All, I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...? I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form

Hi all, I tried to boot up a VM using two options (a) reading boot information from a config file (b) adding boot information directly to the command line. However, I got "IP-Config: Unable to set interface netmask (-22)" problem when using option (b). For (a), I don''t have such problem. The boot information for both (a) and (b) is same. Could you please let me know what I am

> Hi all, > I tried to boot up a VM using two options (a) reading boot > information from a config file (b) adding boot information > directly to the command line. However, I got "IP-Config: > Unable to set interface netmask (-22)" problem when using > option (b). For (a), I don''t have such problem. The boot > information for both (a) and (b) is same.

Hello, All! We was upgrade our Asterisk from version 0.7.2 to 1.0.7. And have big problem. When asterisk starts: ------------------------------------------------------- *CLI> h.323 show codecs Allowed Codecs: Table: G.729A{sw} <1> G.729{sw} <2> G.723.1{sw} <3> G.711-uLaw-64k{sw} <4> Set: 0: 0: G.729A{sw} <1> G.729{sw}

Send these questions to Asterisk-Users mailing list. h323.conf ################################################## ; ; Configuration file of OpenH323 channel driver ; [general] listenAddress=W.X.Y.Z ; local ip listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=yes h245Tunnelling=yes h245inSetup=yes jitterMin=20 jitterMax=100 ipTos=none outboundMax=100

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