search for: h323

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to be using it .. connection is not established, I have pasted a dump file below .. anyone knows what's wrong ? i'm beyond my level of asterisk knowledge at this point :( thanks Dav...

I'm trying to use a Howlink CL-100 ip phone with * It's h323 phone with very limited protocol support. But it's enough that I can use it to dial netmeeting client and artisoft pbx just fine. When I try to dial my * with it using either chan_h323 or oh323, it seems to fail on negotiating H245. Maybe this phone doesn't support it? I've used all...

Hi All, I have set up a box that will be used as follows: SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server 192.168.1.5 192.168.1.50 192.168.1.80 Asterisk is running the latest CVS and oh323 driver. The SIP phone is a Grandstream Budgetone 100. I have everything setup and running with G.711 ALAW and ULAW and i'm able to make calls through Asterisk between the SIP phone...

...to the gateway * Rings a couple times on destiny * Call gets hungup. On the CCM I get the following error: MediaManager - ERROR wait_AuConnectErrorInd On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not available) On asterisk: -- Executing Dial("SIP/test1-6e3a", "oh323/2668#011582129783294@IP|50|tr") in new stack -- Called 2668#011582129783294@IP -- OH323/L31594 is ringing -- H.323 call 'ip$localhost/31594' cleared, reason 1 (Cleared by local user) -- Hungup 'OH323/L31594' == No one is available to answer at this time --...

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 th...

Hi Friends, Hope you would help me out here, I have searched the asterisk user list for hours and also read the readme and test files that comes with the driver. I need a very simple scenario. I have SIP clients and want to use oh323 to dial out to PSTN using a h323 gateway. a)If I set the extention.conf like this: exten => _87.,1,Dial(OH323/16.52.153.206) oh323 dials out (I can ring a netmeeting client at 16.52.153.206). (b)But if I set it like this, oh323 will not dials out ? exten => _87.,1,Dial(OH323/${EXTEN:1}@16....

Hi, I have downloaded files and also local versions of pwlib oh323 (both Janus patched). Both libraries compile fine, but I get following errors on asterisk-oh323-0.6.5. Readme is a bit confusing since it doesn't mention which local libraries should be downloaded from inaccess to get everything working OK. I've also tried with/without patching oh323 with...

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the...

We have an odd problem, where inbound H323 (chan_h323) calls will sometimes cause a Ringing Congestion that appears to keep the channels open and never release it until we kill and restart asterisk. These "Ringing Congestions" start to pile up, which eventually crashes Asterisk. H323 Gateway -> Asterisk (chan_h323) -> To...

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...

Hello all, I just installed Asterisk with H323 support (chan_h323 from Jeremy McNamara). But experience problem while connecting OpenPhone to Asterisk Here is h.323 trace: 5:37.444 H323 Listener:9c86de0 transports.cxx(1504) H323TCP Started connection: host=10.120.160.15:3172, if=10.120.160.99:1720, handle=27 5:37.444...

Hello! I see many of you experience troubles with H323 stack. I am focusing on building H323-SIP Asterisk based softswitch with all codecs supported (including G729 and G723). I can setup Asterisk from scratch with H323 support or solve your h323 nightmare with existing asterisk system for for 100$. Contact me pls offline.

Hello everybody, my * box is connected to gnugk with H323 channel. If I call from an H323 EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio start but noisy (scratch) , then became ok for callee (SIP EP) but still scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 EP and it's ok. And from now, it's also...

...that the altigen is sending a ReleaseComplete and dropping the call before it gets routed to the snom190. Here's an h.323 trace 9: 0:26.489 H225 Answer:9a33350 transports.cxx(1136) H225 Incoming call, first PDU: callReference=1471 0:26.492 H225 Answer:9a33350 h323caps.cxx(1942) H323 Added capability: G.711-uLaw-64k <1> 0:26.496 H225 Answer:9a33350 h323caps.cxx(1942) H323 Added capability: UserInput/hookflash <2> 0:26.499 H225 Answer:9a33350 h323caps.cxx(1942) H323 Added capability: UserInput/dtmf <3...

Hello, Users. I loaded module chan_h323.so, chan_vpb.so. I have met a message : "No one is available to answer at this time". I don?t know what I do.. My 'h.323 trace 5' result is : == vpb/1-8: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR] -- Executing Dial("vpb/1-8", "h323/192.168.1....

...er and I need the lists advice. My plan is to use asterisk PBX with some hardware to terminate my calls coming from several operational gnugk gatekeepers. Do have RH9 and downloaded the latest asterisk from CVS. Compiled according instructions and is running fine. Could hardly find any info on h323 implementation untill the REAME in the channels directory. I am an experienced openh323 gatekeeper user so do have good compiled libraries of openh323 and pwlib on my box. Version 1.5.0 and 1.12.0 to be exact. Ran the make in the h323 directory and compile seems to run fine. Than I ran the m...

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have lic...

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 * tryin...

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,S...

...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 -- Executing [73133 at internal:1] Dial("SIP/59000-08203708", "H323/73133 at Avaya") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 73133 at Avaya -- H323/Avaya-1 is making progress passing it to SIP/59000-08203708 -- H323/Avaya-1 is ringing -- H323/Avaya-1 answered SIP/59000-08203708 == Spawn extension (internal,...