similar to: How do I get caller's number in oh323 ?

Displaying 20 results from an estimated 4000 matches similar to: "How do I get caller's number in oh323 ?"

2003 Dec 03
2
"oh323 calling party number"
How do I get asterisk to populate the "Calling Party Number" field in an H.323 call? I have asterisk configured to accept a SIP call and connect it to an H.323 IVR system. The call goes through, but the caller id is put in the "Display" field rather than the "Calling Party Number" field. -----Original Message----- From: Skuse, Phil [mailto:Phil.Skuse@vicorp.com]
2003 Oct 13
1
oh323 inband dtmf - Possible bug?
I'm trying to use H323 for the first time so please forgive me if I've made a mistake here. I have downloaded and compiled the latest versions of pwlib, openh323 and asterisk. I have dtmfmode=inband in h323.conf, but the remote system is not hearing the DTMF. Running a trace reveals the following... 1:08.398 ThreadID=0x00022012 h323.cxx(4594) H323
2003 Oct 14
0
Has something changed with AGI recently?
I updated to the latest CVS yesterday, from a version several months old. On one of my extensions, I have an AGI script in priority 1. Previously, the AGI script would run and when it terminated, asterisk would move on to priority 2 and connect the call. But now, when it terminates, it starts all over again in a continuous loop and never gets to priority 2. Do I need to update the priority in the
2003 Sep 09
5
Xlite = no sound
What's the secret to getting sound through Xlite? The SIP messages all look OK to me, but the sound isn't coming through. It was trying to use GSM, so I searched the archive and tried: disallow=gsm allow=ulaw Now it says that it's using ULAW but I still get no sound in either direction. Phil Skuse (MBJEJPIEUI) <phil.skuse@vicorp.com>
2003 Apr 23
4
Grandstream BudgeTone 100
After reading about these $75 SIP phones on this list, I purchased a couple for evaluation. They do work with asterisk - and are good value for money, but as somebody commented: they are not yet perfect. I just wondered if anybody had managed to get either the message-waiting indicator or the conference button to work? Phil Skuse <phil.skuse@vicorp.com>
2003 Jul 17
0
Sip call question
There's something that I want to set up in our lab for testing purposes, but I'm not sure how to do it. I would like to be able to call an asterisk extension, and then enter a SIP address using DTMF, and then have asterisk make a SIP transfer to that address. For example: If I dial <extn> followed by 4444*192*168*0*10*5060 I would like to be transferred to
2004 Oct 28
0
Permission denied creating Clearcase view on Samba share.
If I try to create a clearcase view on a Windows 2000 Server share then it creates a subdirectory tree. If I try it on a Samba share, it creates the first directory correctly with the right user mapping and permissions and then returns permission denied. I am able to manually create files and directories on the samba share using explorer, so I guess that clearcase must be using some different SMB
2003 Apr 25
3
Internet Dial-in security questions
Hi, My company wants to put a SIP address on their website. The idea is that potential customers can call that address and will be forwarded to our main switchboard. It's fairly easy in theory because my asterisk server has a real IP address, so any calls to sip:<number>@asterisk-server.mycompany.com should connect just fine (except currently it will be blocked by the firewall). Our
2003 Nov 07
3
Unable dial out with the new Oh323 0.5.6
Hi all, i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then i've installed the new chan_oh323 (0.5.6). when i try to make a call with "netmeeting" through * ( * dial out with "Dial,OH323/${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
2004 Oct 08
0
problems with asterisk-oh323-0.6.3b
Hi guys, I've been trying to update my chan_oh323 from 6.1 to 6.3b. I built asterisk from cvs-head on the date Micheal said he made it compatible, pwlib-1.6.6 and openh323-1.13.5 (both with nothing more than the ./configure, make, well aplied patch on openh323) When I start * with my normal config I get this: [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing
2004 Sep 04
1
Oh323, Please Help Newbie ;(
Hi, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no
2006 Apr 08
2
oh323.conf problem
I have installed oh323 channel driver (finaly! :)). I head some problem starting * so I have put the smallest possible oh323.conf file to se what happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts but he also disables h323 channel because there are no available codec's (*2). When I put codec (*3) Asterisk doesn't start (*4). What have I done wrong? I
2005 Aug 29
2
Register Asterisk with Gatekeeper - oh323
I have tried everything. to register with this gatekeeper to make and receive calls These are the details I received from the voip provider: protocol H.323 Gatekeeper Address - AVS@210.21.118.XXX Port - 1719 RAS - 53 Q931 - 80 h245 - 1722 RTP - 1722 Username - H323 I have 2 phone number/accounts with this gatekeeper that I need to register to. ID - HMA0200.10szxn-xxxx e.164 - 22xx2912
2005 May 25
0
oh323 problems - Solved
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
2004 Nov 26
0
"reason 23 (Temporary failure)" when using Dial(OH323)
I've complied the OH323 .so successfully and can easily receive calls from my H323 gatekeeper (using 711u), however it seems that all outgoing calls are refused and I'm getting "reason 23 (Temporary failure)" as an error code which I can't find documented everywhere. My H323 gatekeeper needs a 001NXXNXXXXXX to dial out to the PSTN even if I'm in north america (Montreal)
2004 Aug 03
0
OH323 not dial Modem[i4l]/g1
Hello everybody, I have a strange comportment with oh323 and asterisk, I'start testing asterisk but with this I can't understant plesae help me ! Thanks Eltorio ---------------------------------------------------------- 1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a Modem[i4l] line ---------------------------------------------------------- Nothing happens
2005 Feb 15
0
oh323 question
I'm trying to connect an asterisk server via oh323 to a Lucent iMerge. I patched the code due so that Lucent can handle asterisk's ver4 h323 http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration I can now successfully dial in to our company over multiple lines. The issue is when I dial out The first outgoing call connects to an outside user A The second call drops the first
2004 May 04
1
Probs with oh323 driver: audio only in 1 direction
Hi, try to setup asterisk as an ISDN2H323-Gateway. The only problem i have after establishing a call is, that Audio works only from IP to ISDN-Phone but not from ISDN to IP-Phone. A known problem ??? Thanks in advance Michael i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus) and oh323-0.6.0 Here are my config's ############## # modem.conf # ##############
2003 Jul 08
2
oh323 problem (small one)
I have just compiled & installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's'
2005 Jul 07
1
Calls with oh323 with no sound
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