similar to: Remote SIP Extension Best Practices

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, Is there a "speed dial" type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar

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

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

I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get

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 list, I'm trying to create an asterisk 1.8 rpm with SRTP. I found mention of a libsrtp rpm, <http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm > in these instructions, <http://www.voip-info.org/wiki/view/Asterisk+SRTP> but it is unreachable (by me, anyway). The libSRTP source is here, <http://srtp.sourceforge.net/download.html>. Has this already been packaged for

Hello, When trying to use logger into any RSpec I get: logger anyboundefined local variable or method `logger'' for #<#<Class:0x278bee0>:0x253cdec> Not sure if this is the spected behaviour (haven''t checked under the hood) It was just annoying not having a logger instance available as in Rails out of the tests. -- An?bal Rojas http://www.rubycorner.com

Hello, Not sure if I am doing something really wrong (let''s not say stupid for now), but I haven''t been able to stub a controller method like: controller.stub!(:logged_in?).and_return(true) Please help, this is driving me nuts ;-) -- An?bal Rojas http://www.rubycorner.com http://www.hasmanydevelopers.com

On 2/22/19 7:56 PM, Joshua C. Colp wrote: > On Fri, Feb 22, 2019, at 2:48 PM, hw wrote: >> >> Hi, >> >> when trying to use SRTP, I can see UDP traffic from phones to the >> asterisk server being dropped be the firewall on arbitrary ports. > > There is no separate port range used for SRTP, and Asterisk does not control the port that the phone uses for sending

Hi, when trying to use SRTP, I can see UDP traffic from phones to the asterisk server being dropped be the firewall on arbitrary ports. Where do I configure the SRTP port range (like the rtp port range)? Why aren't the clients talking to each other directly but apparenty try to send the SRTP traffic to the server? That the traffic is being blocked by the firewall is probably the reason

We're compiling a detailed solution for this problem... We'll be posting it ASAP... Julio Rojas jrojasuft.edu.ve ----- Original Message ----- From: "Mats olsson" <mace2442@hotmail.com> To: <jrojas@uft.edu.ve>; <samba-technical@lists.samba.org> Sent: Wednesday, February 20, 2002 4:05 AM Subject: Re: [Samba] IT WORKS!!! > Howabout publishing the working

Hi all, We're testing TLS and SRTP on Asterisk 1.8.10.0 and have it working with a commerical (not self-sign) AlphaSSL wildcard (GlobalSign) using Blink Lite 1.6.2 as per https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial We've tested with Bria on an iPhone and that doesn't recognised the commercial CA (GlobalSign Root CA). On a Yealink 28P with V60/V61 is registers

On 2/23/19 1:15 PM, Joshua C. Colp wrote: > On Sat, Feb 23, 2019, at 8:06 AM, hw wrote: >> On 2/22/19 7:56 PM, Joshua C. Colp wrote: >>> On Fri, Feb 22, 2019, at 2:48 PM, hw wrote: >>>> >>>> Hi, >>>> >>>> when trying to use SRTP, I can see UDP traffic from phones to the >>>> asterisk server being dropped be the firewall