Displaying 20 results from an estimated 300 matches similar to: "reason 24 (Call ended with Q.931 cause)"
2004 Jul 22
1
Sip -> H323 using oh323 and G729
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
2003 Jul 21
4
anyone with X100P & Callerid working outside US ?
I'm just curious if anyone has the X100P & Callerid receiving working
outside US.
Replies are appreciated. Also if it's not working for you in a certain
coutry you can respond too.
regards
Martin
2003 Jul 10
2
OH323 + G729 + Go2Call
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
2003 Nov 27
6
Help for oh323
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
2005 Jan 06
0
H.323 to SIP extension
Greetings All-
I have an * box with the NuFone H.323 channel driver installed.
I also have an Altigen VoIP system with a PRI to the PSTN.
I can sucessfully make a call from a SIP extension (snom190)
to an H.323 extension (altigen phone)
The thing I can't seem to make work is a call from a H.323 phone
to a SIP extension.
Here's the layout:
2003 Sep 12
2
problem with * and Howlink CL-100 ip phone
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 different versions of
2004 Aug 04
5
H323 Call Dropping
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 gatekeeper, however
the gatekeeper drops the call
2006 Nov 23
1
asterisk 1.4 chan_h323, help please...
Hi,
My configuration is SipPhone<-->*1<--->*2.
My asterisk version is 1.4beta3.
I installed pwlib,openh323,chan_h323.
When i call from
SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2,
there is no audio.
Using 'rtp debug', I can see that rtp packets are
being received.
Rtp packets are being exchanged.
I also tested chan_ooh323, but to fail.
Can anyone recommand best
2003 Nov 07
3
Unable dial out with the new Oh323 0.5.6
Hi all,
i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then
i've installed the new chan_oh323 (0.5.6).
when i try to make a call with "netmeeting" through * ( * dial out with
"Dial,OH323/${EXTEN}@xx.xxx.xxx.xx" ) the call will be blocked.
Before, there was chan_oh323 0.5.5 and pwlib(1.4.11) and openh323(1.11.7)
installed, and it worked.
Is here
2009 May 20
8
fixme:heap:RtlCompactHeap (0x110000, 0x0) stub
Hello!
I'am a new Linux-User and I try to install ELOOFFICE on my Ubuntu 9.0.4. For this purpose I loaded the Package of Eine 1.1.21 (but I tesed the official Version too).
The Installation of ELOoffice runs without Problems. But when I run ELO ist crashes after the Login-Screen.
When I start the Application tyoing on the ConsoleI get the followin Error Messages:
wine
2004 May 18
0
problems with asterisk-oh323
Hello,
I've been trying to send traffic to a Cisco Call Manager 3.2, but with
no luck.
Here's whats happening:
* Call gets to CCM
* Call gets 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:
2005 Mar 03
0
I have met a message : "No one is available to answer at this time".
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", "h323/192.168.1.107") in new stack
1:21:34.936 ThreadID=0x06f2bbb0
2004 Aug 11
7
H323 call dropped when answered
Hi All.
I'm using RedHat 9
I configured the chan_h323 and asterisk from CVS.
This is the scenario SJ_lab_phone(sip) ---------------> Asterisk
-------------> H323 GK --------------> PSTN
I have tried all codec's and always the same result, the called phone
will ring without dropping for how ever I allow it to but as soon as it
is answered it immediately gets disconnected.
2005 Jan 27
0
Problem with OpenPhone->Asterisk
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 H225 Answer:9cc1250 transports.cxx(564) H225
2003 May 31
1
oh323 problems
i am trying to make calls between two workstations using netmeeting and
asterisk.
i get the popup on both when i call the extensions 665 and 667 but when
accept, i get this error
*CLI> 0:18.190 H225 Caller:8112978 H225 Received connect
PDU.
0:18.288 H245:810b388 H245 Read error: Bad file
descriptor
0:18.318 H323 Cleaner H323
2003 Dec 17
1
PSTN to h323
Hi,
I start to be a little confused so I am asking to the list.
I want to make with * a gateway from PSTN to H323, and to send all
incomings call to a predefined IP, which will treat the h323 calls.
let's assume that all my incoming numbers starts with 00
here is my extensions
[incoming]
exten => s,1,Answer
exten => _00.,1,Answer
exten =>
2008 Oct 18
1
strange h323 delay issue
Hello,
I have a strange h323 issue. After executing command
"Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18
22:32:23. Meanwile I have sniffing traffic on port 1720. The call was
established just at Oct 18 22:33:03 (New H.323 Connection created.) and also
packet sniffer grabs the h323 invites at this time also. So my question is
what
2009 Jul 06
1
TOSHARG-DomainMember.xml translate finish and some bug found
Now, TOSHARG-DomainMember.xml translate to Japanese finished.
and Some bug found.
<procedure>
<title>Server Manager Account Machine Account Management</title>
-------Domain?
<step><para>
From the menu select <guimenu>Computer</guimenu>.
</para></step>
When the user elects to make the
2004 May 06
4
asterisk-oh323, new version 0.6.1
Hello all,
This new version (0.6.1) of asterisk-oh323 fixes the "one-way audio"
problem of the previous release.
Download from the usual location:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Regards,
Michael.
2003 Jul 23
2
h323 gateway call lost after 74sec always
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.