similar to: Video phone settings???

Hi! I run a mongrel cluster with 6 mongrels in it. I want to monitor them individually for process hangs (and then restart them) and this is the solution I came up with: Here''s my configuration file for monit (/usr/local/etc/monitrc): [snipped relevant bits] ------ #check lighttpd process check process lighttpd with pidfile /var/run/lighttpd.pid start program =

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI>

Hi, We recently installed CentOS 6.2 on our cluster. During the installation/debugging of various secondary software, we had disabled iptables. When we re-enabled them, we found that the front-end would no longer X11 forward (although it does so when the iptables are off). What do we need to set in the iptables to permit X11 forwarding? Currently we're using iptables -P INPUT DROP

On Tue, 15 Feb 2005 03:02:45 +0100, Stefan Gofferje <stefan@gofferje.homelinux.org> wrote: > Hi folks, > > I have some trouble with the FOP and would appreciate if anyone could > point me into the right direction. There is a FOP user list, although not too active. http://www.asternic.org/ > Is there a way to define a button like Zap/g1/6000 and have it light up > when

I am using Asterisk and an X101P card as a glorified answering machine. We have a residential PSTN line with about six phones connected to it. Like an answering machine, I want Asterisk answer the line *only* when an incoming call is not answered after four rings. This mostly works. My extensions.conf is at the end of this message. The problem is that Asterisk will sometimes answer the line when

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3

Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>

Hi All, I am using "asterisk-11.12.0" version and I am trying to setup secure call (TLS + SRTP) between two extensions and while making a call, I got following error *CLI> == Using SIP RTP CoS mark 5 -- Executing [6004 at from-office:1] Dial("SIP/6003-00000000", "SIP/6004,20") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/6004 SSL certificate

Hi Lists, I have one box with two FXO and two FXS ports, it is running asterisk inside. I have one sinle POTS line connected to the one FXO and two phone sets connected to the FXS port. Extension 6003 is asigned to one fxs and 6004 is asigned to another fxs, the two extensions can call each other, they can both make/receive PSTN call, but they can't make PSTN call simultaneously. Is it

new warnings, I think lib/CodeGen/SelectionDAG/DAGCombiner.cpp: In member function ‘llvm::SDValue<unnamed>::DAGCombiner::FindBetterChain(llvm::SDNode*, llvm::SDValue)’: lib/CodeGen/SelectionDAG/DAGCombiner.cpp:6006: warning: ‘SrcValueOffset’ may be used uninitialized in this function lib/CodeGen/SelectionDAG/DAGCombiner.cpp:6006: note: ‘SrcValueOffset’ was declared here

Hello, Suppose a dialplan such as: exten => 6004,1,Answer exten => 6004,n,Wait(1) exten => 6004,n,SayDigits(1) exten => 6004,n,Festival(This is a test of Festival) exten => 6004,n,Hangup When watching in the CLI, I see this: == Using SIP RTP CoS mark 5 -- Executing [6004 at internal:1] Answer("SIP/505-00000004", "") in new stack -- Executing [6004 at

Hi friends, I am just starting use asterisk for our VoIP server. It works fine in LAN. But when it is deployed in public network(with a public IP), the SIP clients in different NAT fails to communicate with each other. I have set 'icesupport' to 'yes' in sip.conf and set STURN and TURN server in rtp.conf. It still fails! Hope someone to help me out! Thanks in advance:) This

Hi Guys, Just new to Asterisk and am completely stumped. I have created two accounts as instructed. Please see below for the config of the user accounts. [Peter] type=friend host=IP address disallow=all allow=ulaw allow=alaw callerid=Peter <6004> secret=XXXXXXX context=default port=9060 nat=force_rport,comedia deny=0.0.0.0

Hi, I have a difficulty on my Asterisk's database.How can I get the info about list of ringing agents on my queue In console : -- Started music on hold, class 'default', on DAHDI/77-1 *-- SIP/6002-00cc0f90 is ringing -- SIP/6004-00c23270 is ringing -- SIP/6005-00be6220 is ringing* -- SIP/6004-00c23270 answered DAHDI/77-1 -- Stopped music on hold on DAHDI/77-1

Hello, I am new to this list. We are trying to move away from our WDS server with PXElinux mods and are testing using iPXE on a linux box only. We mainly use HP but also have Dells, Panasonic, and Lenovo. On almost a dozen models tested now, everything works great. These were mostly laptops but a Dell Optiplex 360 and 380 booted fine. So far we have just had a problem with the HP 6005 Pro SFF

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

hi, exten =>6000,1,dial(SIP/6000,15,tr) exten =>6002,1,dial(SIP/6002,15,tr) exten =>6004,1,dial(SIP/6004,15,tr) exten =>6006,1,dial(SIP/6006,15,tr) exten =>6008,1,chanspy(SIP/6006 | wbq) when i dial 6008 ,it is connected ,but i can't able to hear the voice of the any one. when coversation between the 6002 to 6006. in my Console mode i got the following comment *CLI>

Hi all, I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything seems nice, but i'm not able to make calls nor to receive any. When I try to make a call, I keep receiven the "all circuits are busy now" message, and when I receive calls, asterisk doesn't seems to care (don't get anything on the CLI) I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's