similar to: Asterisk on Solars?

Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: <sip:4000@serve.address.com;user=phone> (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have

Hi, Has anyone installed Asterisk on FreeBSD? i need help/steps on this task -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061122/054eb688/attachment.htm

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of J. Oquendo > Sent: Thursday, April 26, 2007 6:47 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Asterisk brute force watcher (was FYI) > > Steve Totaro wrote: > > I suspect that

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

Long story short... Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Useragent : snom360/6.5.2 Function: F_RETRIEVE [root@pbx ~]# asterisk -rx "show version" Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on 2006-11-17 16:35:22 UTC [gateway] exten => 201,hint,SIP/201 exten =>

Hi guys, I need your help ... I have a couple of DIDs that reach my Asterisk box .... But I'd like to set my DIDs automatically via Macro or other routine based on the number called by my agent ... Ex: My agent called 954-111-1111 ... So I'd like to set the callerid as 954-222-2222 (That is my DID) Thanks in advance, Marcelo

Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360), I noticed my client is having some form of feed back on the phone. Because of Snom's "inner oddities" this is how I got it to work. Plantronic --> RJ11 --> SnomHandset Port (on Snom Base) Handset --> Plantronic jack (bottom base in the front) If I placed Plantronic(RJ11) --> Snom's

Hi folks, I'm having a problem with installing R on Solaris. Has anyone seen a similar issue? I don't find any hits on the search engines. Thanks. Environment: SunOS xxxx 5.10 Generic_118822-25 sun4u sparc SUNW,Sun-Fire-280R During the installation/compilation of R-2.10.1 I get the following error: begin installing recommended package boot ERROR: cannot extract package from

Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails -- #Joseph

Has anyone gotten this configured to show all extensions vertically instead of filling up the window. If so would you mind sharing your configuration Yes I have tried searching terms like +asternic +op_panel +vertical and a slew of others. Unsucessful though. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g'

Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16

Hey all, I have a question for those administrating/building out systems with over 30 users on them. How often do you experience the dropped call phenomena. Would you care to share your experiences including what versions of * you were using, what kind of connectivity was present (T1, Fractional T, Intergrated T, DSL, Cable). Echo? Solutions? (e.g. we bought an X_Brand Echo Canceller). Also,

Hi all I have installed 1.2.12.1 in FC5 with libpri.1.2.4 when i start iam getting the following error and it quits == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325 __load_resource: libpri.so.1.0: cannot open shared object file: No such file or directory Oct 23 16:16:07 WARNING[11084]: loader.c:554

So I have whose autoattendant is colliding with their extensions... Quick fix anyone? Second someone presses say a person's extension (101) ... Autoattendant sends them to the first context... [companyx-main-aa] exten => s,1,Background(companyx/companyx-main) exten => s,2,Background(silence/10) exten => s,3,Background(companyx/companyx-main) exten => s,4,Background(silence/10)

Hello everybody, I am running 1.2.10-BRIstuffed-0.3.0-PRE-1s with florz-patches on Linux 2.6.16 with 4 HFC-Cards in TE-mode connected to 4 PtP ISDN-"Anlagenanschluesse". There are about 40 SIP-clients connected (mostly Sipura/Linksys PAP2, and some SNOMs and softclients) to this server. Everything works fine, except that my CDR reports some very long _concurrent_ calls from one sip

Dear list, following problem, i have some users, who are supressing their callerid. This setting is adjusted at the sip phone. So if these guys are calling internal persons nobody sees the callerid. I am looking for the following resolution: User has set his phone to anonymous, user calls somebody internal, Asterisk initials a lookup on the channel and generates a new callerid for the

Victor Hoodicoff wrote: > > > I think your impressions of Aastra are outdated. Install the latest > firmware, download the latest documentation and test and THEN give an > opinion! Did you miss the part when I wrote I have Asstras sitting on my desk collecting dust. I program on average about 5 per month, deal with about 40+ per day. They're as impressive as that Hyundai in

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

problem - occasional garbled calls, mostly remote users. T1 connected to PSTN, SIP over local LAN and internet to "remote users". NAT at local firewall and at remotes. There is no traffic shaping in place, no QoS. Most are Polycom phones, two Aastra's. Start with QoS on LAN switches? No 2x4's please, start with 1x4's. joe a.

Anyone running caller id spoofing applications in the USA running asterisk? Then it's time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-t o-be-outlawed.html Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial).