similar to: Any ideas why I can't dial this SIP phone (sometimes)?

hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology Inc Info System SIP Provisioning Regional Line 1 User 1

Hey, I'm currently using the GotoIf application to set it so if certain caller ID's call my number, it will transfer it to my cell phone, here is the code I have so far. I get an error message that states "call rejected by 198.22.67.70: No such context/extention." when I call the number from my house number. Anyway, here is the code I have. [inbound] exten =>

Hi I'm having trouble # transfering queue calls. in extensions.conf I have: [macro-queue] ; ; Places caller in queue ; ${ARG1} - Queue name to place caller in. ; ${ARG2} - Voicemail Extention ; ${ARG3} - Caller ID to Set. exten => s,1,DBget(temp=nm/on) ; Get Night key, if not existing,goto 102 exten => s,2,Playback(custom/500/10) exten => s,3,SetCallerID(${ARG3}) exten =>

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4 FreePBX = 2.11.0.4 I am trying to understand flow control in Asterisk dial plans and not having very much luck. I have read the Asterisk book from O'Rielly, or at least those parts I believe might apply, but that has not helped me much on this particular issue. What I wish is to set three distinct ring tones on our Snom phones for

Hello How do I fix this.... [IPComms-in] exten => ${IPCCIDN01},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN01},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN02},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN02},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN03},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN03},2,SetCallerID(${CALLERID}) exten =>

Hello, I have agents / queues working to the extent that agents can login, logout and I can send a caller into the queue and the logged in agent's phones will ring. Maybe I've spent to much time googleing and reading and my eyes are crossing now, but what I am trying to do is this but cannot find any reference to it. 1. Xfer the caller into the Queue... If Noone is logged into the

I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: ----------- [some-context] Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten => s,1,SetCallerID(${ARG2}) Exten => s,2,SetMusicOnHold(${ARG3}) Exten

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 We use Snom870 handsets with firmware v.8.7.3.19. I am trying to develop a custom dial plan to invoke a distinctive ring-tone when an external call is transferred internally. Based on an earlier solution I discovered I am attempting this: [from-internal] include => set-alert-if-local [from-internal-original]

Hi, there: I have installed xen(2.6.18.8-xen) and had successfully started domU guests (one or two) from dom0. However, I am not able to start the network for domU machines. Following is the information from my linux box. After the it is started: $ ifconfig eth1 Link encap:Ethernet HWaddr 00:A0:C9:D6:65:DB inet addr:XXX.XXX.10.133 Bcast:XXX.XXX.10.255 Mask:255.255.255.0

Here is the current dial plan section: [custom-michael] exten => _900,1,Playback(custom/extn-xfer) exten => _900,2,SayDigits(${EXTEN}) exten => _900,3,MixMonitor........... exten => _900,4,Dial(SIP/${EXTEN}|${DEFRT}) exten => _900,5,Playback(custom/extn-xfer2) exten => _900,6,Goto(custom-michael,901,4) exten => _901,1,Playback(custom/extn-xfer) exten =>

Run /usr/src/zaptel/ztmonitor 32 -v And adjust your gains in /etc/asterisk/zapata.conf accordingly. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May 07, 2004 3:35 AM To: asterisk-users@lists.digium.com

i'm posting this to the bind-users and rsync mailing lists in order to get this archived for the next person who needs to do this :) comments and suggestions are welcome, but please cc: to me because i'm not subscribed to either list. craig ----- Forwarded message from Craig Sanders <cas@taz.net.au> ----- Date: Fri, 27 Jul 2001 10:30:20 +1000 From: Craig Sanders

I'm using FreePBX 2.5.2.2 with Asterisk 1.6.1.4. If I make a call and then decide to blind transfer them using ## my side of the call is not hung up. Instead it sends me to voicemail. If somebody calls me and then I blind transfer them with ## I am hung up on as expected. I called from 8678 to 28688. I then transferred the call to 8532. Asterisk acts like it wants to hang up, but then

https://bugzilla.samba.org/show_bug.cgi?id=4320 Summary: Daemon should send stdout, stderr of {pre,post}-xfer command to client Product: rsync Version: 3.0.0 Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P3 Component: core AssignedTo: wayned@samba.org

Hi, all I'd like to configure Asterisk to receiving call from PSTN. After PSTN phone call in, Asterisk will prompt user to enter a number, then Asterisk will transfer the call to a SIP phone by this number. Please help me check the following extensions, is that OK? thanks! [from_pstn] exten => _.,1,Answer() exten => _.,2,GoTo(Xfer,s,1) [Xfer] exten =>

Hourly I have an rsync job backup /home to /home/backup. I have 24 directories (one for each hour): home.0 ... home.23 Here is the script I am running via cron: #! /usr/local/bin/bash dest=`date +%k | sed 's/ //g'` linkdir=`date -v-1H +%k | sed 's/ //g'` chflags -R noschg /home/backup rm -rf /home/backup/home.$dest rsync -ahHP --numeric-ids --delete --stats --link-dest=../

Guys below are my rsync options i am using :- rsync -PaRz --delete --delete-excluded --copy-unsafe-links --numeric-ids I have omitted the -v and --stats i was using previously. Now when i look in my log i still seem to get a list for each file transfered. I only want to have the problem files listed here such as permission denied etc. Any ideas? log output receiving file list ...

https://bugzilla.samba.org/show_bug.cgi?id=4130 Summary: Does not delete partial files upon completion of transfer Product: rsync Version: 2.6.8 Platform: Other OS/Version: Windows XP Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org

https://bugzilla.samba.org/show_bug.cgi?id=4131 Summary: Does not delete partial files upon completion of transfer Product: rsync Version: 2.6.8 Platform: Other OS/Version: Windows XP Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org

Hello! I have an Asterisk@home instalation with 7 users working OK, and I'ld like to implement either a -- Web dial feature, where the user would fill one form field with a phone number and a connection would be created between his extention and the entered number. OR -- Dial using an URI (callto:xxxxx link in a web page), having AstTapi installed and configured in all workstations.