similar to: Test my test-branch!

defines.h defines a bunch of IPTOS constants if they're not already available: #ifndef IPTOS_LOWDELAY # define IPTOS_LOWDELAY 0x10 # define IPTOS_THROUGHPUT 0x08 # define IPTOS_RELIABILITY 0x04 # define IPTOS_LOWCOST 0x02 # define IPTOS_MINCOST IPTOS_LOWCOST #endif /* IPTOS_LOWDELAY */ A few lines further down, it includes <netinet/ip.h>, which

In Sweden, where I live, it's snowing like crazy. The Stockholm area is covered in white stuff and there's really no reason to leave the computer and get out anywhere. More white stuff is coming down all the time. Boring. I am sure your weekend is no better - rain, snow or just another boring sunny day. Let's find something cool to do during this weekend! Join the cool crowd

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

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

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

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

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

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

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 --

I'm just one of those weirdos, who wanna make a powerfull queues shaper (not QoS but near) with ipfw2 on their freebsd 4.x-stable. My server is using frequently used configuration with NAT+FW ADSL router with one external ip on external network interface (we're using ADSL modem in bringe mode). I've configured single pipe, configured queues to use that pipe, add queues with different

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

Dear all, I ran PDC on smbpasswd auth. When i use following the command, i receive : ////////////////////////////////////////// debian:/usr/local/etc/samba_3# ./bin/net rpc join mylove -U root Enter root's password: Creation of workstation account failed Unable to join domain MYLOVE. debian:/usr/local/etc/samba_3# /////////////////////////////////////////////////////// my smb.conf is :

Hello, I''ve tried searching for this but I don''t seem to be able to find a way to search past archives in this list. Is there a way to get a per-process qos functionality in linux? At this very moment, I''m using with success a kind of ''workaround'' in my server, which involves creating multiple virtual ethernet interfaces with different IPs and binding

Anyone have any info on the date (or bug tracker number) of the change from incominglimit to call-limit, and is it call_limit or call-limit? Does it work with SIP friends? Running CVS head 8/24 (right before 1.2 release). -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'd like to be able to use my Snom 360 LEDs to view the status of parking slots, so I'm trying to install the "metermaid" patch (http://bugs.digium.com/view.php?id=5779). Can someone help an svn newbie figure out how to install this patch? I've done the following: svn checkout http://svn.digium.com/svn/asterisk/team/oej/metermaids cd metermaids make clean make but I get

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

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

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

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

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/