similar to: Speed Dial Application in *

Hi all, I have just successfully configured a Cisco 30VIP to work with my Asterisk server. I have seven of these phones new and would like to deploy them. I am wondering if anyone has this phone deployed with Asterisk and can suggest configuration of the various buttons, etc. (Bare with me as I am new to Asterisk.) Thanks, Chris

Hi all, I'm working on a script to check the status of streams from multiple streaming sources (ie. icecast, shoutcast, wms, etc.). With Icecast I am looking at the HTTP header returned by the server when I do a GET on the mountpoint I want the status of. It appears that a HTTP/1.0 200 indicates that the mountpoint is up while a HTTP/1.0 404 indicates that it is down. Two questions: 1. Is

What are best practices for allowing connection by remote SIP extensions over the internet? I'm thinking of putting the SIP inside a VPN connection. Kind Regards, Chris

Hi all, I'm new to rsync and have a problem I cannot resolve even after plowing through the lists. I have an rsync server setup with an rsync client. I am trying to backup a directory structure that is over four levels deep. Rsync does great until it hits that fourth level. Then it errors out for each file in that fourth level similar to this: rsync: recv_generator: mkdir

During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if

Samba: 3.6.6 PDC Client: XP Pro SP3 Background: 1. Started with a clean installation of XP Pro SP3 2. Joined the client to the samba domain 3. Logged in as user 'root' the first time after the join. 4. Added "Domain Users" group to the "Local Admin" group on the client (forget about the security implications for the moment) Now when a user (any user) logs on to the

I'm starting a new thread because the issue now is different from the one I was originally experiencing. Here are two level 10 debug logs one from each of two servers: http://w4fbc.org/samba/DC.log http://w4fbc.org/samba/FS.log Each has a test share setup like this: chgrp -R staff-faculty /test chmod 0770 /test chmod g+s /test setfacl -m g::rwx /test [test] path = /test read only = no

I am running Samba 3.6.6 on a Ubuntu 12.10 Samba domain member server. Users are authenticated against a Samba DC running 3.6.9 over an LDAP backend. I have a share configured as show below. Members of the 'staff-faculty' group can browse the share, but cannot write files to any subdir for which they are not the owner. It appears that the only reason they can read/traverse is because of

Hi, I'm running Samba 3.6.6 on Ubuntu Quantal. I have a need to manually assign some of the UID mapping on a samba domain member file server. I have used tdbtool to add the correct mapping record to winbindd_idmap.tdb. However, I am at a loss as to how to force that change to "propagate" so as to show in the permissions structure of the file system and in the output of such

Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is

Hi all, I decided the best way to get to know * well is to do it from scratch. Having read the majority of the "Asterisk: The Future of Telephony" I am now attempting to compile zaptel 1.4.0 and am receving the very same series of errors mentioned in this post on the forums: http://forums.digium.com/viewtopic.php?t=13619&highlight=zaptel1+++zttranscode++error However, there has

Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including

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

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

Hello I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection ....now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. this is a requirement .. Is it possible ?? what is the way forward ?? please give me a

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have 4 employees and we're running Cisco 7970 phones. Each phone has a unique SCCP line configured (in the autologin area of the sccp.conf file) for each employee. We have hints set up in the extension.conf file like the following: exten => 101,hint,SCCP/101 exten => 102,hint,SCCP/102 exten => 103,hint,SCCP/103 exten => 104,hint,SCCP/104 We have speeddial= lines set

I get the following message when I dial 74#. Does anyone have any ideas on what might be going on? If I don't require numbers to be terminated with # everything works as expected, but you have to wait for the digit timeout, of course. MESSEGE: DEBUG[1150520624]: File pbx.c, Line 1683 (ast_pbx_run): Oooh, got something to jump out with ('#')! -- Invalid extension '#' in

Hello @List, i'm using a Cisco 7970 / 7914 phone with Asterisk 1.4 & sccp part of my sccp.conf type = 7914 (Cisco 7970 with 7914 phone extension) autologin = 117 description = Test speeddial = 10,Test (10),10@mycompany my speeddial line is for internal call howto call a external number eg. +xx467584933 ?? i want to press a button and call directly out thank you greetings, Daniel

Quoth: OCG Technical Support <support at ocg.ca> > >Although we've programmed the softkeys per the manuals, they seem to have no >effect (just dead). For example, our 57i is setup like this: I had similar problems and ended up using the speeddial inband functionality. FWIW, my 57i's setup like so: softkey4 type: speeddial softkey4 label: "*Park" softkey4