similar to: SIP transfer/REFER to voicemail problem

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>

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic

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 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

-----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 :

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

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

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

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 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

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

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 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, 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

ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1 7960). lots of bugs. when i press the speed dial button on either 7910, asterisk dies. also, if i dial from the 7910 to 7910, everything works fine. i can dial from or to the 7960 once, and then one of the 10's and the 60 die and try to reregister. if i take the 7960 out of the mix and remove its

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>

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox

I am following the instructions in https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice's phone (caller), but Bob's phone (callee) doesn't ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob

I'd like to know how everyone else is going about getting correct CDR information for calls. Right now I notice that if a call come in and gets parked the CDR info doesn't how the correct info on who picked that call up, also when someone transfer a call there isn't a new record being made so the duration of the call shows up for who received the call and transferred it. I started

Hello! I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") Thank you!