similar to: Snom100 H.323 sample config

Displaying 20 results from an estimated 2000 matches similar to: "Snom100 H.323 sample config"

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 Mar 20
1
HELP: Failed start after install asterisk_oh323-0.7.1
Hi, ALL: I install my oh323 channel driver following steps of http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en I works my asterisk well before install the chan_oh323.so. But after I do "make install" the oh_323, my asterisk crash and show me the following message (asterisk -vvvvvvc). Does anyone have any idea about it? What's wrong
2003 Aug 18
3
Pops
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi. Using inAccess Networks chan_oh323, I'm experiencing some clicks or pops, how can I fix that? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/QMfF2TEAILET3McRAu9zAJwNWtv+QSpka0NGaVk9E/IDHyalhwCgkQME Gynfp5zF0SWZUQEjelp7sBI= =CSqT
2005 Mar 09
6
how to sip->h323 using asterisk-oh323-0.7.1
hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web
2003 Jun 03
2
Asterisk Works on Linux on Sparc
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:
2005 Aug 18
2
Asterisk (OH323) - gnugk connection
Hello there. Is there somebody with this connection working? I can't seem to make this work at all. Could someone please share some .conf files? Cheers, Vedran. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/71480bd9/attachment.htm
2007 Oct 05
1
[asterisk-dev] oh323.conf, extentions.conf
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
2003 Oct 16
2
AGI problem (crash)
Hi Every time I hangup on my AGI script Asterisk crashes if it is not running in console mode. (happens when using python and perl AGI scripts) I'm desparatly trying to get my employer to let me use Asterisk. So I must get this to work. I've posted about this before, I'm sorry, but I'm desperate. I'm running RedHat 9.0 (kernel 2.4.20-8 everything else updated) I'm
2003 May 18
3
SNOM100 GSM again
OK I did some researches and tests with it, and finally: I registered my messenger to the asterisk and called if from the snom. The audio from the snom to the messenger was PERFECT. By the time of the call This message was running on the asterisk console: WARNING[16400]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames My conclusion is that the snom100 utilizes MSGSM
2003 May 16
5
Snom100 GSM
Hi, there were some postings a few weeks ago telling that the GSM codec problem with snom100 will be fixed. But it still seems to be very quality. Will be any change in this subject? THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/2fa9d206/attachment.htm
2004 Aug 05
1
h323 gnugk to h323 asterisk and then to endpoint
hi, we are using a voip h323 switch. the switch sends all caals to our Gatekeeper (gnugk). gnugk musst send all calls to asterisk and asterisk must do his choice (sip endpoint or out to PSTN) Making calls to our h323 switch works fine over asterisk. what must i configure to get inboung h323 calls from our gnugk to asterisk? any hints for me? thx -- Thomas K?pper 01063 Telecom GmbH &
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
2005 Jan 28
3
reason 24 (Call ended with Q.931 cause)
Hi Michael and Everyone I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting this error "reason 24 (Call ended with Q.931 cause)" I've checked the Asterisk wiki and several other resources. Please can anyone give me a hint on what the problem is I reach my wits end. Thanks Tola my config and debug Configuration of OpenH323 channel driver
2006 Feb 22
1
Problema calling from elesign h.323 to iax device
Hi, i'm using an elesign voip gateway esc1700 to call to one iax sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when I make the call using the esc1700 the communication is dropped, this is the log portion: Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Call accepted by 200.93.220.21 (format ulaw) Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Format for call is ulaw Feb
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
2005 Sep 30
1
No ringback tone generated by Asterisk with OH323connections
are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users
2004 Sep 28
4
Gatekeeper registration failed
Dear friends, I have compiled and installed h.323 in my asterisk. And I have a service from a H.323 VoIP provider who give me user, password and gatekeeper IP address. All configured. But when I start my asterisk i receive the following error and h.323 calls can not be making and/or receiving. [chan_h323.so]=> (The NuFone Network's Open H.323 Channel Driver) == Parsing
2003 May 03
1
Failed calls from SNOM100 to *
Hi all, I've just been setting up such a phone with * and I'm facing some problems. At this time there is no customized extensions & others, so I use extension 8500 (voicemail) in default config. What I get when this extension is called from the SNOM is pasted below. The phone ignores *'s invites and * sends a "BYE" to it. Please note the IP in the "Contact"
2006 Jan 21
1
h323 configuration
Can any body give me an example how to configure h323 in Asterisk. Which files do I need to configure? just extensions.conf and h323.conf ? Thanks, Patricio _________________________________________________________________ Descubre la descarga digital con MSN Music. M?s de un mill?n de canciones. http://music.msn.es/
2004 Apr 13
1
SIP->h323 problem DTMF
I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --