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...who sees us as 10.138.3.2:4569 Why doesn't server A see server B as 192.168.2.44?? All other traffic going over these lines has no problems with this. The 192.168.2.x & 192.168.11.x networks are fully 'connected' to each other... Who knows the answer...? Regards, Evert Meulie

Hi all, I have been looking at various alternatives for a system that runs several Linux & Windows guests. So far my favorite choice would be OpenSolaris+ZFS+RAIDZ+VirtualBox. Is this combo ready to be a host for Linux & Windows guests? Or is it not 100% stable (yet)? Greetings, Evert This message posted from opensolaris.org

Hi everyone! I'd like to create the following: a user picks up the phone (gets a dial tone), dials '0' for an 'outside' line, gets a second (different?) dialtone, and is able to enter an external phone number. How do I implement this in extensions.conf...? Regards, Evert

...ot; -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 'asterisk' displayed on my Cisco 7960 & 7912... On Wed, 22 Sep 2004 14:06:51 +0200, Evert Meulie <evert@witelcom.com> wrote: > Hi! > > When I call a colleague of mine from my Cisco (via Asterisk), they get > on their display: > From Evert > asterisk > > How do I remove/change the 'asterisk' part? > > Regards, > Evert You need to set a v...

Hi everyone! I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf ***** [ip-incoming]

Hi! Problem with my 7960. Outgoing calls work, but incoming don't. A 'sip show peers' gives: Name/username Host Dyn Nat ACL Mask Port Status 105/105 192.168.2.175 D 255.255.255.255 5060 UNREACHABLE Is there something wrong with the config on that phone? If so, who can tell me what? Regards, Evert

Hi everyone! Who can tell me how I can access my voicemail? When I dial the voicemail on my Cisco 7960 I get access, but when trying to enter my mailbox number it seems that Asterisk doesn't 'get' any of the keys I press. DTMF problem perhaps? Any suggestions on how/where to fix this...? Regards, Evert

Hi! If I install a CAPI-compatible ISDN-card in my server, will that: a) enable me to connect that server to the public phone net b) allow me to connect an ISDN phone to the server and use it as a SIP-phone c) all of the above? Regards, Evert

Hi! When I call a colleague of mine from my Cisco (via Asterisk), they get on their display: From Evert asterisk How do I remove/change the 'asterisk' part? Regards, Evert

Hi all! We use a PHP-portal for management of our projects & contacts. Now I would like to make it possible to dial contacts directly from the portal. Since users have to log in, I can use that to determine which office phone the call should originate from. And the number-to-be-dialed is of course also listed. How do I commence here? I'm pretty sure others have done this already, so

Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 11111 -> ext. 1 913 - 22222 -> ext. 2 913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the

Hi all! I'm trying to have asterisk route all outgoing calls out via my VOIP provider. exten => _NXXXXXXX,1,Dial,SIP/BYEXTENSION@VOIP seems to have them to in the direct direction. However, debug shows that my asterisk doesn't identify itself correctly: Sip read: SIP/2.0 100 Trying From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa To: <sip:[dialled

Hi everyone! The following: Any calls coming in on extension 12121212 should get a message telling them to dial the extension of the person they are trying to reach, and then press #. The call should then go to the entered extension. This is as far as I got... *********************************************************** exten => 12121212,1,Wait,1 exten => 12121212,2,Answer exten

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden

...y just setting the email address to deliver notifications to. Yet every time I try to start the service I get this: # service nagios start nagios is stopped Configuration validation failed [FAILED] The error appears to be identical to that reported here: http://forums.meulie.net/viewtopic.php?f=61&t=6538&start=0 Like the original poster of the thread on that forum, I am able to start it with "nagios -d /etc/nagios/nagios.cfg" though various web pages continue to report an unspecified configuration error. Does anyone have any ideas? Regards, Ben -...

Hi all! Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building... The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and after about a minute the phone

When I do a 'asterisk -vvvvvc' I get following, but asterisk does NOT stay up: == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found

Hi everyone! Situation: when I call from cell phone to a asterisk-connected phone, all works fine. When I call from the asterisk-connected phone (a Cisco 7960 SIP) to the cell, the connection gets made, but there is no audio going in either way... Asterisk reports the following: Sep 16 08:27:41 WARNING[245775]: chan_sip.c:2679 process_sdp: Insufficient information for SDP (m = '', c =

Hi everyone! I was wondering... Does the musiconhold quality improve if the mpg123 processes run with a negative priority? If so, is there a way to make them start like that, so I don't have to renice them? Regards, Evert

Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 11111 -> ext. 1 913 - 22222 -> ext. 2 913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the