similar to: freebsd 4.10 and port misc/zaptel

has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a

hey all, am wondering if anyone has successfuly done a SIP attended transfer using the REFER method (after an INVITE obviously) and the Replaces: header. we're writing our own SIP UAC and the asterisk code seems to support it, but we're not really sure if this is so. we plan on the following call flows: 1. incoming call from exten 1111 is sent to our UAC with Dial() 2. our UAC makes

we're evaluating the use of a Lucent APX8100 E3/SS7 to SIP gateway for use in conjunction with asterisk, serving something like 4000+ lines. does anyone have experience with the APX8100 and it's integration with SIP on asterisk ? does the APX8100 handle SS7<->SIP signalling well enough to be used ? any anecdotes would be well appreciated. -- Regards,

i have quake v1.06 installed and running fine under windows. however, running it in wine gives a bunch of errors. see below: prophet% wine --winver win98 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD. Try --winver nt40 or win31 ! prophet% wine --winver nt40 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD.

the attached log with verbose=6 and debug=6 refers. we've got a sangoma A104 (no hwec) with PRI ports 1 & 3 loopbacked to each other. we're trying to have txfax send out on one of those pri ports with rxfax listening on the other side, hence having asterisk send a fax to itself. we however have bad, and i mean really bad (<10%) success rates. we're currently using asterisk

in a simple dialplan like follows: [firstcontext] include => secondcontext include => thirdcontext include => fourthcontext [fourthcontext] _03X.,1,Goto(${EXTEN:2},1) _X.,1,DoSomething() _X.,2,Hangup() the Goto() for exten _03X. seems to start the search for the jump within firstcontext, thus possibly matching an exten in secondcontext or thirdcontext first before hitting the

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type

I posted about a winbind/PAM issue earlier and discovered that the auth portion of pam_winbind is what was causing my failures. I receive the following message when trying to authenticate sshd or login with the auth pam_winbind module: Jun 3 20:43:04 gonzo pam_winbind[14850]: request failed: No logon servers, PAM error was 4, NT error was NT_STATUS_NO_LOGON_SERVERS Jun 3 20:43:04 gonzo

Greetings from Hack in The Box -- We are pleased to announce that the Call for Paper (CfP) for HITBSecConf2006 - Malaysia is now open! Set to take place from September 18th - 21st 2006 at The Westin Kuala Lumpur, this years conference promises to once again deliver an International deep-knowledge security conference. HITBSecConf has been described as "the most intimate of hacker

https://bugzilla.samba.org/show_bug.cgi?id=13147 Bug ID: 13147 Summary: inconsistent behaviour regaring vanished files information Product: rsync Version: 3.0.9 Hardware: x64 OS: Mac OS X Status: NEW Severity: minor Priority: P5 Component: core Assignee:

hi, i facing a problem here. in my sip.conf, i specify a account like this, [1234] type=friend context=from-sip username=1234 secret=1234 nat=no canreinvite=yes dtmfmode=info mailbox=1234@default disallow=all allow=ulaw so i am able to login with username 1234 and password 1234 but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus

Hello everybody, I am quite new to asterisk. I have tried to connect two asterisk boxes with the iax2 protocol, but although the web is full of documetation for that, I was quite unsuccessful. The last try was Example 2 from http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers I am running two machines, "gonzo", my home system and "laptop", a machine wit a

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration

https://bugzilla.netfilter.org/show_bug.cgi?id=1329 Bug ID: 1329 Summary: flush set not working: "Could not process rule: Invalid argument" Product: nftables Version: unspecified Hardware: x86_64 OS: Debian GNU/Linux Status: NEW Severity: major Priority: P5

This evening I posted a new set of Digium G.729 codec modules to our FTP server and web site, for Linux x86 and x86-64 processors. They were built using GCC 4.0.1, and they now report the processor they were optimized for when they are loaded. The previous x86-64 module required a non-standard Asterisk binary configuration, so this was corrected. In addition, there was only a generic version

hello i am trying to work with radiusclient form portaone. but i have some problems in installation. when i am trying to use example from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth error sip debug Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

https://bugzilla.netfilter.org/show_bug.cgi?id=1424 Bug ID: 1424 Summary: v0.9.0: segfault when using nft -f <file> and issuing "ruleset flush" twice Product: nftables Version: unspecified Hardware: All OS: Debian GNU/Linux Status: NEW Severity: normal

Greetings everyone, I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy allows you to communicate with multiple Asterisk boxes from a single point of contact using a variety of I/O formats, now including support for XML, HTTP, HTTPS, SSL, CSV, and the Asterisk-native standard format. Astmanproxy is

is where anyone who knows what is needed to get the pickupexten (*8) running ? gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff I've activated it in features.conf (default *8) and also tested other extensions res_features.so is loaded show features says: Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind