similar to: SIP Absolute Timeout

To be clear I meant using Chan)_h323 with Call Manager where CM is configured with * as a H.323 gateway, not client. CM supports H.323 to direct calls through gateways, and in fact until recently that is all they supported. They now also have MGCP, but only to their IOS platforms, and SIP is coming soon. There are NO sccp-based gateways, from Cisco anyways. Dan -----Original Message-----

Hi! I use the following configuration to register my asterisk server to my SIP provider: register => 12345:passwd@sip.provider.com/12345 sip.conf: [sipout-test] type=peer username=12345 fromuser=12345 fromdomain=provider.com secret=passwd insecure=very host=sip.provider.com qualify=yes context=test-incoming extensions.conf: exten => 12345,1,Dial(SIP/10) exten =>

Hi guys, When I try to get ring tones when dialing out with the command Dial(SIP/sipout/${PHONE},15,r), I get the error message indicated in the subject. I've checked my indications.conf file using the sample file provided with asterisk 1.4.10 (the version I'm using) and it's not better. Any idea ? Regards. -- Cyril SCETBON

Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ

Hello All I am trying to answer incoming SIP calls, first, by dialing an extension, thence into voicemail, which works; and secondly by going straight into voice mail which does not. The extension.conf that works is like this; [incomingSIP] exten=>_.,1,Dial,Zap/2|1 exten=>_.,2,Voicemail,u5152 exten=>_.,3,Hangup the extension.conf which does not is like this; [incomingSIP]

Hi all, Is it just me and not reading the docs right, or has anybody else had problems with the AbsoluteTimeout application and the 'T' extension when used inside a macro? [macro-attended] ; ARG1 is the device to dial out on, SIP or Zap, or whatever ; ARG2 is the extension to dial using 'attended' dialing exten => s,1,AbsoluteTimeout(30) exten =>

I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session

Hi, Have a problem in my extension.conf: I have: [sip] exten => _333.,1,wait,3 exten => _333.,2,Answer exten => _333.,3,AbsoluteTimeout,7 exten => _333.,4,Hangup I wanted to test if * is executing this dial plan by calling 3335254255 for example. The problem is as follow: It waits, it answers but it does not seems to see the Absolutetimeout: call goes forever. What's wrong? Am

Greetings, I am running Asterisk with a T100P and a Zhone channel bank for over a month now. For the most part it works fine but from time to time (about once a week) the system will not let go of a line and will play the greeting over and over. Anyone calling gets a busy signal. If I reset Asterisk everything works fine. Has anyone seen this problem before and fixed it? If so what did you do?

Hello Conference again. Meetme can now limit number of users in a room. Can it also limit how long a conference session? Someone ask the same question (from achive) but doesn't have a solid answer. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030821/c1ca1383/attachment.htm

In some PBSx you can limit outgoing call that you cannot speak longer the 15 minutes. Is it possible to do with Asterisk ? Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/6eb392cc/attachment.htm

Hi everyone, I'm new to Asterisk, but I like it ;o) Have a question to you; How can I limit the incoming call duration? -- Suich

I'm using the M() option to Dial() and having problems. In the following dialplan example ANY digit exits the macro. When the callee presses 1 the Noop(Reset AbsoluteTimeout(0)) does not get run. Does anyone have any ideas as to what I'm doing wrong? Asterisk 1.2.x [extensions] exten => 2998,1,Dial(Zap/1/5551212,,wM(answer-confirmation^20)) [macro-answer-confirmation] exten

When I attempt to start asterisk with my modem setup listed it will not start attached are the error messages i get and also the modem.conf that i am currently using. Any assistance would be greatly appreciated. running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run with it (from http://www.opencall.org) just e-mail me privately if you need more info Thanks in

This probably came up before, but I have a faxing question for everyone. I have a simple extension setup to use rxfax to receive faxes sent to asterisk. It is: exten => s,1,Answer() exten => s,n,AbsoluteTimeout(300) exten => s,n,Set(FAXFILE=/var/spool/asterisk/fax/${ARG1}_${CALLERIDNUM}_${UNIQUEID}.tif) exten => s,n,rxfax(${FAXFILE}) exten => s,n,System(/usr/bin/mailfax

Hi, I still cant dial out on Zap and I really have no clue why. I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4 ports, 31 channels each and able to receive incoming calls and fax perfectly. I've done this in my dial plan. exten => 111,1,Answer() exten => 111,n,Ringing() exten => 111,n,Wait(2) exten => 111,n,AbsoluteTimeout(30) exten =>

Hi, I'm trying to setup a small BRI ISDN <-> voip gateway. The ISDN card is based on Cologne chipset, so I try set it up with zaphfc. The versions i'm running: kernel-2.4.27 Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e zaptel modules 1.0.7 zaphfc is from bristuff-0.2.0-RC8e When I'm doing the insmod on zaptel, zaphfc, zaprtc: Zapata Telephony Interface Registered on major 196 PCI:

Hi , I''m getting random hangups on zap channels with long calls. It seems that the hungup happens after 10 minutes or so. AbsoluteTimeout is set to 0. Any other thing I should be configuring? Thanks! PHM

Hi all, I'm new to the forum. Oh no....newbie question coming, I hear you all cry! I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and