similar to: Call Pickup problem in Asterisk with SIP phones

If I have an IAX client (Firefly softphone in this example), and the client is not registered at the moment because they are not connected to the network and someone dial that extension, they get the user's "I'm on the phone at the moment" message vs. the "I'm unavailable" message. Is this by design? Here's the extension in question's dialplan:

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Hi .... I'm looking for thoughts/experiences when the following conditions are met ... I have 2 users .... say Sam Smith and Jeff Smith On UNIX (on the samba server) their logins are Walt Smith = "smith" Jeff Smith = "jsmith On the NT DOMAIN side, their logins are Walt Smith = "wsmith" Jeff Smith = "smith" In my smb.config file, I

Can the Manager Redirect command transfer a ringing SIP extension? I'm trying to implement a Camp On feature, and having failed to do it in Dial Plan logic, am trying to do it with manager logic. If an arbitrary 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:

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 all >> >> As of 5pm PST today (7/13), I pulled

I have a classic domain. The PDC and BDC are Samba 3.6.25 on Solaris 11. I have two domain members also Samba 3.6.25 on Solaris 11. I have two domain members that are samba 4.1.17 on Fedora Core 21. LDAP backend for unix and samba accounts. in smb.conf on member servers idmap config * : backend = tdb idmap config * : range = 5000-6000 idmap config MYDOMAIN

I've tried numerous times but cannot get Vampire to bring across all groups or add users to all groups that they belong to. Sometimes I get everygroup except 1, other times not so lucky. Vampire log is below with an example of what is going wrong. vampire.log Fetching DOMAIN database Creating unix group: 'skischool' Creating unix group: 'sales' Creating unix group:

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Hi List, I'm having a bit of an odd problem with asterisk and outgoing zap calls. Tzafrir has been kind enough to help me get the logging sorted out so I have some idea of what's going wrong, but I'm a little flummoxed. Essentially the symptoms are as follows; Make a SIP call from Cisco 7960 or 7940 to asterisk, where it is routed out on a ZAP (x100p) line. After

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

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

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

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

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

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