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timeout to be set.<br> I&#39;m hoping to find an option along the lines of the Dial() ringtime,<br= > but no luck.<br> Gosub() looked interesting, but I don&#39;t think quite fits my needs eithe= r<br> <br> Could someone please offer a little insight on this situation and<br> point me towards the right command to be playing with?<br>

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ CLI Output : ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vicidialnow*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from

Situation: * New Install of Asterisk * 7960 w/ SIP 7.4 Image * 7912 w/ SIP040406A * 3 Lines Defined on the 7960 (5104,3100,2100) Questions (configs are below): * Why won't the MWI light on the Cisco? I've tried: * mailbox=2100 * mailbox=2100@default * mailbox=2100@wvlandsales-voicemail * Does anything look goofy overall? :-) Thanks, George Sip.conf [2100]

Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and 1.4.7.1 on FreeBSD 6.2) [general] priorityjumping=yes With n+101: exten => 1337,1,Dial(SIP/zytek,5,Ttj) exten => 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS}) exten => 1337,n,Hangup -- Executing [1337 at firma:1] Dial("SIP/113-087a3000", "SIP/zytek|5|Ttj") in new stack -- Called zytek

I installed asterisk, astguiclient, php and mysql. but when i dialled one number to another number my asterisk server give the following error: > /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi > install_driver(mysql) failed: Can't load > '/usr/lib/perl5/site_perl/5.8.8/i486-linux-thread-multi/auto/DBD/mysql/mysql.so' > for module DBD::mysql: libmysqlclient.so.15:

We have quadBRI configured 1 port in TE mode 2,3,4 ports in NE mode. We are trying to place a call from the phone connected to BRI card port #4 to city number through ISDN line connected to port #1. Number successfully dialed. Person on the other end answering the line. But conversation can't last more then 10 seconds. Below is a log of such call. Its not clear for me why we appear in

We have now got our new PRI line (10 channels, 100 numbers) connected and everything is working except the outgoing caller ID. Whatever SIP phone I'm using, the CID that's shown is the very first number... ----- s n i p ----- [default] include => outgoing include => priin [outgoing] exten => _NXXXXX.,1,Macro(dial,08${EXTEN},${RINGTIME}) ; Local number (w/o areacode) -

Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls. Can this be because I nowhere use the Answer() application in my dialplan when dialing out? -----Original Message----- From:

I have two sites running Asterisk PBX. Normally the inbound calls go through a 3rd (colocated) server and are routed via IAX to the site (the site registers with the main server) I created a macro that tries to ring one location and then another. Each site explicitly Answer() the call even though it will only ring all the sip phones at the relevant location. When fall back is in effect it goes to

Hi all, i'm trying to figure out if i'm being silly or if there is a genuine problem. Running on the notorious Redhat 7, 2.2.16-22 #1, X86. [user at host]$ ssh-agent -s SSH_AUTH_SOCK=/tmp/ssh-XXYFcFR6/agent.2101; export SSH_AUTH_SOCK; SSH_AGENT_PID=2102; export SSH_AGENT_PID; echo Agent pid 2102; [user at host]$ echo $SSH_AUTH_SOCK [user at host]$ echo $SSH_AGENT_PID [user at host]$

Jon, thanks for your help, but I'd rather not do it using agents and queues, ideally what would happen is it would simply play the message and wait for the person to press a button, if nothing is pressed, it just keeps going down the list. Any other suggestions? asterisk-users-request@lists.digium.com wrote: >Date: Wed, 15 Jun 2005 00:53:14 -0500 >From: Jon Gabrielson

Jon, thanks for your help, but I'd rather not do it using agents and queues, ideally what would happen is it would simply play the message and wait for the person to press a button, if nothing is pressed, it just keeps going down the list. Any other suggestions? asterisk-users-request@lists.digium.com wrote: > Date: Wed, 15 Jun 2005 00:53:14 -0500 > From: Jon Gabrielson

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Hi All, Asterisk 1.4.12 CentOS 5 My ISP account includes nearly 500 minutes of VOIP calls per month but the service is expensive for unbundled minutes. So I'm trying to find a way to keep an accumulated total of calls made through that trunk so that I can automatically switch to a lower-cost provider when my bundled minutes are used. The plan is to store the accumulated time in AstDB and

On 2024-10-14 14:48, Damien Miller wrote: > On Sun, 13 Oct 2024, Jan Eden via openssh-unix-dev wrote: > > When I connect to serverA (`ssh -v -o UpdateHostKeys=yes serverA`) > > afterwards, known_hosts on the client is not updated. The output of the > > ssh command contains this: > > > > debug1: Host '[serverA.domain.internal]:22' is known and matches the

/**************************************************************************** * * * Always Be Conferencing (ABC) * * * * Creator: chris @ Penguin PBX Solutions * *

Hi All, I need to stop the transfer feature on particular sip user. I am using linksys phone and it has set the forwarding enable to another user. I have three users 2101, 2102, 2103. 2102 is registered in linksys phone with forwarding enable to 2103. But is there any procedure in asterisk that we can not allow 2102 not to forward on 2103. and also i want to prevent the SIP/2.0 302 Moved

On Sun, 13 Oct 2024, Jan Eden via openssh-unix-dev wrote: > Hi, > > I created new host keys on serverA, updated sshd_config accordingly > (adding the line below) and restarted ssh: > > cd /etc/ssh > sudo ssh-keygen -f 2024_ssh_host_ed25519_key -t ed25519 -N '' > > sudo vi /etc/ssh/sshd_config > # added line: HostKey /etc/ssh/2024_ssh_host_ed25519_key >

I just can't get kphone to register with asterisk, i can make calls to the demos and even get into the mailbox but kphone cannot register. Here's my story. Can you help me?? Please I have installed asterisk on debian using apt-get install asterisk. I have configured an extension in extensions.conf as follows exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt) exten =>