similar to: Troubles with Macro-stdexten and dial

Hi, As you asked, I have included my diff to what I did for the DIAL command. I probably didn't stick to some * pre-agreed standard of coding or something, so if these things offend you then I suggest that you close your eyes. :) The biggest thing to consider when you are doing a prepaid system is, what if the person with the same account in/out calls twice? I chose, for now, just to keep

Hello friends, I am a newbie to asterisk , please help. I am receiving a phone from a sip server and I want to route it to another sip server. The problem is that the target sip server takes a # in the argument . I am trying to dial as exten => s,5,Dial(12345#123456789@xxx.xxx.xxx.xxx) ;;;; dial the number at the ip address. When this gets executed, I get the following eror:-

I'm running into errors when using Enum lately. I can't figure out what the problem might be as I've had Enum up and running in the past. I'm running the latest CVS-Head compiled version. I've also tried using the new Enum function with the same results. When doing a lookup on a number that exists in the enum server I get the following results: -- Executing

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

hello how to solve these errors /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10 use Asterisk::AGI; vi /etc/asterisk/extensions.conf exten => _X.,1,agi,agi-rad-auth.pl|Routing=SIP&AuthorizeBy=SIP vi /etc/asterisk/modules.conf load => res_agi.so <---------------errors------------------------> *CLI> Can't locate Asterisk/AGI.pm in @INC (@INC contains:

I'm making a samba/pdc server with windows (2k e xp) clients. For now, that's ok, but I got some errors: - Some files (ex.: /home/silvana) don't show up in Windows Explorer. I've already checked on the server and all files of /home/silvana are into the server, but windows explorer don't show them. All the files are .doc. An interesting thing is that if we put the complete

You can play also with www.enum2go.com <http://www.enum2go.com/> or wap.enum2go.com Regards Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040727/6c42c39d/attachment.htm

Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom --------------

hello all, i got my htb running and controlling my bandwidth right, but i have a doubt. i configured HTB to limit my class 192.168.0.0/24 to 128kbit. but i want 128kbit for *each* machine on this class, and not for the entire class. when 2 machines in 192.168.0.0/24 are downloading one file, they share the 128kbit bandwidth between them, using 64kbit for each. why i can get all machine on this

Hello In the LARCT how-to subitem: 4.2.2. Load balancing the following phrase says: "" Instead of choosing one of the two providers as your default route, you now set up the default route to be a multipath route. In the default kernel this will balance routes over the two providers. It is done as follows (once more building on the example in the section on split-access): ip

Hi, I'm trying to build a vpn between two hosts with valid IP address. One maquerades the private network 192.168.1.0/24, the other masquerades 10.101.0.0/24. Configuration files and routes seem to be ok. I'm getting the following on beta: Could not set up a meta connection to alpha: Ago 5 17:56:24 Viatel01 tinc.nextel[16355]: Trying to re-establish outgoing connection in 245

Sorry, repost --------------------------------- Hi, I'm trying to build a vpn between two hosts with valid IP address. One maquerades the private network 192.168.1.0/24, the other masquerades 10.101.0.0/24. Configuration files and routes seem to be ok. I'm getting the following on beta: Could not set up a meta connection to alpha: Ago 5 17:56:24 Viatel01 tinc.nextel[16355]: Trying to

Buffer overflow detect in FC6-x86_64, see down. Juan. R version 2.4.0 (2006-10-03) Copyright (C) 2006 The R Foundation for Statistical Computing ISBN 3-900051-07-0 R ? um software livre e vem sem GARANTIA ALGUMA. Voc? pode redistribu?-lo sob certas circunst?ncias. Digite 'license()' ou 'licence()' para detalhes de distribui??o. R ? um projeto colaborativo com muitos

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Hello all, I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and I'm completely confused by the gosub/stdexten thing. I used to call the stdexten macro but I haven't been able to figure out how to use Gosub. I'm using the sample extensions.conf and added something like this: ========================= [home] include => stdexten exten =>

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten,8006,Sip/8006) [macro-stdexten] ; ; Standard extension macro: ; ${ARG1} -

Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten

I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489. When one phone calls another, I see the following on the console (here, 6223 dials 6123) -- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489", "stdexten|6123|SIP/6123&IAX2/6123") in new stack -- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489",

Hi! Is there a way to avoid being "at the middle" of communications between two SIP endpoints? So that we can avoid loosing bandwidth with it? Is there a way to "forward" the authentication to a IAX provider and "transfer" the call to it, avoiding using my own bandwidth? I've tested it with SER with some results, I was wondering if it is possible with

Hi! Are you also getting in trouble while trying to register in Broadvoice? Cumprimentos / Best regards, Helder Rog?rio __________________________________________ Microrede - Tecnologias de Informa??o, Ltd. http://www.microrede.pt *** ? There are only two types of people in the world, those who have lost data and those who will. ? -- Richard Nixon