similar to: Asterisk behind IX66

Can anyone help me with this: - root@foxy:~ > asterisk root@foxy:~ > asterisk -r Asterisk CVS-v1-0-12/28/04-11:37:32, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com> ========================================================================= Connected to Asterisk CVS-v1-0-12/28/04-11:37:32 currently running on foxy (pid = 1751) foxy*CLI> foxy*CLI>

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Hi, Has anyone got incoming IAX to work on the above router. I can call out, but incoming calls are not reaching the * box. Has anyone got this working? Could they give me some configuration hints. Thanks John

I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

Due to some failed hardware on another platform, I've had to move a T100P card to a different chassis. After this move was completed, I am seeing some strange results on the T100P card that do not display to me any failure mode with which I am familiar. The card comes up, and shows "good" carrier and status, but Asterisk refuses to bring the D-channel up. Calls of course do

My username is different on my Linux box than on my Windows box so I used the line username map = /etc/samba/smbusers in my smb.conf file and this file includes the line steve = steveb However when I'm logged on to my Linux box as 'steve' and use the smbclient command like this: # smbclient //<Windows computer>/Shared ... session setup failed: NT_STATUS_LOGON_FAILURE it

At time to time I receive some junk faxes from some advertising companies that play smart and don't provide any TSI number so I can not bock them by the number in Hylafax. Despite calling their Fax Removal Service 1-800-... number several time they refuse to obey my request. So I would like to setup a small script or context loop in extension.conf if possible and maybe run it overnight; maybe

Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any

Hola! I'm using asterisk as H.323 -> PRI gateway. First call goes thru ok, second concurrent call fails with: Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri] -- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to

Hi Everyone, I'm new to * and I'm trying to setup a small configuration of SIP clients. Eventually when I get this working I plan on expanding with a Digium developers kit to add analog phones and PSTN access. My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both peers seem to register with * but I cannot call to one another. When I dial the associated extension, the

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

Hey gang, There must be any easy solution for this but my mind is frazzled on compiling 2.4 with RTC as module. Bleh. Currently extension 9000 is our VoicemailMain(@company) line. Some employee's are complaining that the old system was better because you didn't have to enter your mailbox number and that instead the old system took you right to it. I figured there was something similar

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

I edited the calleridname.agi patch to only overwrite the name if it is missing. The asteridex option still overwrites the name since it is our master list for known numbers. -- Steven calleridname.agi.patch: --- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue Jun 13 14:37:09 2006 +++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13 14:37:09 2006 @@ -16,6

Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same

Help, Any ideas ? I guess I missing something. make[1]: Entering directory `/usr/src/asterisk/utils' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\"CVS-HEAD-12/27/04-21:28:39\" -DASTERISK_VERSION_NUM =999999 -DINSTALL_PREFIX=\"\"

Hi All, Thought I would let those that are interested know that I had success in running 24G on an Asus P6T with 24G kit of Kingston DDR3. While I was putting this together I saw lots of forum posts asking if anyone had tried it. Well we did here at our work and all looks great including running "memtest86" overnight. I have a fluid dynamics simulation running on it with 90% memory

Hi guies, I know I know this subject have been The most written subject about VoIP Right... but I just want to make clear, just one time ! If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP