[Asterisk-Users] "Catch all" extension

2006-01-14 Thread kleis-asterisk-dev
Hi all, What do you think about having a single extension in the dialplan that matches everything and then delegates the next action to an external application through AGI? I mean something like this: exten => _.,1,AGI,catchall.agi,${EXTEN} Then, catchall.agi could EXEC dialplan macros in functi

[Asterisk-Users] Callware VoiceOne released: a new, easy web GUI

2005-12-19 Thread kleis-asterisk-dev
We proudly announce that VoiceOne has just been released as an ALPHA version at http://www.voiceone.it Main features: * Client/Server architecture based on Web services * Relies on Asterisk Realtime Architecture (ARA) * SIP extensions management (support for Zap/IAX/mISDN soon added)

[Asterisk-Users] A simple network environment: a configuration issue or a bug in Asterisk?

2005-11-16 Thread kleis-asterisk-dev
My Asterisk box is installed in the DMZ of an IPCop firewall. The RED interface of IPCop has a static public IP address, and all traffic directed to that address is forwarded to the PBX in the DMZ. The IPCop also routes traffic from LAN (192.168.2.0) to DMZ (172.16.0.0), so Asterisk is reachable fr

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2005-06-04 Thread kleis-asterisk-dev
>-- Messaggio Originale -- >Date: Sat, 04 Jun 2005 16:16:04 +0200 >From: "Olle E. Johansson" <[EMAIL PROTECTED]> >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Subject: Re: [Asterisk-Users] chan_sip + MD5 encryption: WARNING Format for > authentication entry is user[:[EMAIL

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2005-06-04 Thread kleis-asterisk-dev
>-- Messaggio Originale -- >Date: Sat, 4 Jun 2005 14:52:42 -0400 (EDT) >From: Steve <[EMAIL PROTECTED]> >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Subject: Re: [Asterisk-Users] How to quickly replace ', > ' with '|' in dialplans? >Reply-To: Asterisk Users Mailing List -

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2005-06-04 Thread kleis-asterisk-dev
>-- Messaggio Originale -- >Date: Sat, 04 Jun 2005 12:09:04 -0700 >From: Michael D Schelin <[EMAIL PROTECTED]> >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Subject: Re: [Asterisk-Users] How to quickly replace ', > ' with '|' in dialplans? >Reply-To: [EMAIL PROTECTED], >

[Asterisk-Users] How to quickly replace ',' with '|' in dialplans?

2005-06-04 Thread kleis-asterisk-dev
Finally I decided to rewrite my dialplans according to the right sintax, that is exten => someexten,priority,application(arg1,arg2,...) should be exten => someexten,priority,application,arg1|arg2... Isn't there anybody skilled enough in regular expressions that could write a quick Search 'n' R

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2005-06-04 Thread kleis-asterisk-dev
>-- Messaggio Originale -- >Date: Sat, 4 Jun 2005 14:31:07 +0200 >From: [EMAIL PROTECTED] >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] chan_sip + MD5 encryption: WARNING Format for > authentication entry is user[:[EMAIL PROTECTED] >Reply-To: Asterisk Users Mailing List - No

[Asterisk-Users] chan_sip + MD5 encryption: WARNING Format for authentication entry is user[:secret]@realm

2005-06-04 Thread kleis-asterisk-dev
Hi all! So far I've always used plaintext passwords for SIP, but now I've decided to use MD5 encryption. For each client I edited its section as follows, then: auth=md5 md5secret=hashed_passwd ;secret=plaintext_passwd where hashed_passwd is the output of echo -n "user:realm:plaintext_passwd" |

RE: [Asterisk-Users] Realtime+IAX2 and RSA

2005-06-03 Thread kleis-asterisk-dev
>-- Messaggio Originale -- >Date: Thu, 2 Jun 2005 03:35:26 +0200 >From: [EMAIL PROTECTED] >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] Realtime+IAX2 and RSA >Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > > >Anyone had Realtime working with IAX2 and RSA au

[Asterisk-Users] Realtime+IAX2 and RSA

2005-06-01 Thread kleis-asterisk-dev
Anyone had Realtime working with IAX2 and RSA authentication to connect two PBXs, please? It seems that inkeys/outkey fields are not read at all and the following warning is logged when dialing: Jun 2 02:41:36 WARNING[6299] chan_iax2.c: I don't know how to authenticate to XXX.XXX.XXX.XXX

[Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'

2004-07-26 Thread kleis-asterisk-dev
Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaph