similar to: plainvoip - down ???

Hi: Is there anybody familliar with www.plainvoip.com voip provider. I sent them money through paypal and they didn't add the money to my account and they didn't respond to my request to send the money back to paypal. Is there anything I can do besides disputing the charge with paypal? Regards; Chawki ____________________________________________________ Start your day with Yahoo! -

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

Has anyone gotten Asterisk to compile on Solaris 10? I have tried both 1.2and 1.4 and I get errors about editline. Actually it seems that 1.4 goes through more of the process, but thats not good enough.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061116/82464f01/attachment.htm

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, 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 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

Hi, here what i have: [2001]--[Asterisk]---[ISDN-Trunk]---[PBX]--[8004] Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 on RedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL. Dialing from Astersik extension 2001 to PBX extension 8004 via ISDN Trunk gives me the following error, -- Executing Dial("SIP/2001-8a8e", "Modem/ttyI0:998004|20|r") in new stack

Hi Friends, At present, I am using VoIPJET.COM provider for make calls to USA. I have two doubts. 1) I am unable to make call to UK Mobile phone. Why? 2) I want to make calls to "Turkey" country from "India". With VoIPJET, I am unable to make call to "Turkey" and unable to find VoIP provider for Turkey. Please tell me VoIP Provider for Turkey from India.

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

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'

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 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

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)

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

Hey all. We have a DS3 circuit with GBLX split off into 7 systems with a 4 port sangoma card (A104D) in the first 2 systems, and digium T410P cards in the other 5. GBLX numbers their spans from 0 to 3 instead of 1-4 and we have a NFAS configuration with the d-channel on chan 96. All of our systems are running 1.0.7 for stability reasons (and no good time for maintaince, the entire platform