search for: nmartin

...kupgroup=1 ;the ringing SIP phone: [wsmith] type=friend host=dynamic nat=yes canreinvite=no qualify=1000 ;defaultip=192.168.30.108 dtmfmode=inband mailbox=103 context=Outgoing callerid="Walter Smith" <103> username=wsmith secret=****** pickupgroup=1-4 ;The phone attempting the *8 [nmartin] type=friend host=dynamic insecure=no nat=yes canreinvite=no qualify=1000 ;defaultip=192.168.30.100 dtmfmode=inband mailbox=105 context=Outgoing callerid="Nik Martin" <105> username=nmartin secret=****** pickupgroup=1-4 callgroup=1 The SIP debug: pbxMobile*CLI> -- Starti...

...ion in question's dialplan: ;extensions.conf exten => 106,1,Dial(IAX2/nikko,20,tT) exten => 106,2,Voicemail(u105) exten => 106,3,Hangup exten => 106,102,Voicemail(b105) exten => 106,103,Hangup And here's the CLI debug: pbxMobile*CLI> -- Executing Dial("SIP/nmartin-aeca", "IAX2/nikko|20|tT") in new stack pbxMobile*CLI> Jun 14 10:28:26 NOTICE[4997140]: app_dial.c:554 dial_exec: Unable to create channel of type 'IAX2' pbxMobile*CLI> == Everyone is busy at this time pbxMobile*CLI> -- Executing VoiceMail("SIP/nmart...

...Voicemail(u101) [pbx_config] 102. Voicemail(b101) [pbx_config] '102' => 1. Dial(SIP/dli|20|Tt) [pbx_config] 2. Voicemail(u102) [pbx_config] 102. Voicemail(b102) [pbx_config] '105' => 1. Dial(SIP/nmartin|20|Tt) [pbx_config] 2. Voicemail(u105) [pbx_config] 102. Voicemail(b105) [pbx_config] '600' => 1. VoiceMailMain() [pbx_config] '601' => 1. MeetMe() [pbx_config] '800' => 1. Dial(Zap/25) [pbx_co...

From: "Nik Martin" <nmartin@radiancetech.com>> To: <asterisk-users@lists.digium.com>> Subject: RE: [Asterisk-Users] asterisk compile problem Date: Wed, 14 Jul 2004 09:22:38 -0500 Organization: Radiance Technologies, Inc. Reply-To: asterisk-users@lists.digium.com Fletcher Bonds wrote: >> Hello al...

...rary Sip extension is ringing, I need the ability to pick up that extension from any other phone. What little docs there are on Manager commands shows Redirect takes these parameters: Action: Redirect Channel: Zap/1-1 Context: transfer Exten: 5555555555 Priority: 1 If I'm at Sip channel Sip/nmartin (extension 101), and the phone ringing is Sip/Foodaddy (extension 100), It doesn't seem possible to get that call transferred TO my extension, using the example context that accompanies the Redirect sample on the WIKI: [transfer] exten => _.,1,Dial(Zap/g1/${EXTEN}) It doesn't look dooa...

Of course, big brother wants his say in the matter. http://www.wired.com/news/politics/0,1283,63884,00.html?tw=wn_2polihead

Is there a clever way to camp on an extension in asterisk? What I need is a way to answer my extension (not just a ringing ZAP channel) from any other phone. If I'm in another office and hear my phone ringing, I want to be able to quickly pick it up from that extension. The list revealed the pickupgroup parameter, but that looks like it will pick up any zap channel that's ringing. This

What is the best way to upgrade a production asterisk box? make upgrade? I don't want my configs messed with, and need the process to go as smooth as possible. I fetched and built a new kernel last night, but haven't rebooted into it. I'll do that tonight, and then want to quickly upgrade to the latest asterisk (mainly for zttest.) Does make upgrade fetch head? Thanks Nik

Has anyone ever thought configuring asterisk on a pair of pc's to act as remote broadcast terminals for the broadcast radio industry? Seems like a stripped down asterisk on a laptop with a PCMCIA ISDN modem connecting to another asterisk instance on a PC at a radio station would work nicely. Use one of the higher quality codecs, interface the remote mixer to the sound card on the

In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd like to do something possibly unique with the formatting of extensions in my dial plan, and am having trouble. We use VoicePulse connect, which gives us local DID for inbound and outbound calls (even though DTMF tones are not working in Voice Pulse Connect at the moment). To dial local numbers, you have to

I upgraded to the latest HEAD version of asterisk, and all IAX calls started sounding choppy. It was suggested on the IRC channel that I go back to asterisk -stable to determine if that fixes it. Is downgrading as simple as upgrading? Because now, -stable builds fine, but I get an error on the asterisk console when starting, something about "ast_get_txt" not found. Recompiling and