similar to: Smallest possible Asterisk VM

Not able to compile asterisk,zaptel,libpri in /usr/src -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100208/fa5ff126/attachment.htm

We're using FreePBX 2.8 and there is a Reports tab but it doesn't seem to actually do anything. Is there some secret/trick to getting a report out of it that will tell us which extensions are placing calls? I've tried every query on the form that I can think of. Is the reporting disabled by default or ??? Any tips/pointers appreciated. Ben M. Schorr Chief Executive Officer

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can make call outside and exten 2006 to 2010 can not make call outside. heres my dial plan. ? sip.conf ? [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=outside secret=1234 host=dynamic [2001] type=friend context=outside secret=1234 host=dynamic [2002] type=friend context=outside

We deployed a single phone handset (Polycom 331) at a remote site. We have a IPSEC VPN running between the firewall at the remote site and the firewall at the site where our Asterisk/FreePBX box lives. We have used a similar configuration for this site before and it worked fine. We gave the phone a static IP address and pointed it to the configuration server on the remote end that has the

Is there a database of MAC address prefixes used the common VoIP devices. I see the Linksys Sipura devices state with 00:0E. Does the same apply to other Linksys VoIP equipment? Is there some way VoIP equipment allow themselves to be identified by requesting data from some ports?

I am looking for PHP functions that implement passwords that much the dovecotpw implementation. I downloaded one from PEAR, Crypt_HMAC, but the passwords it generates look nothing like the dovecotpw passwords, which could be my fault because I know nothing about the field. Is there a guide or some library that implements them to match dovecotpw?

I have been asked by my church to recommend a VoIP system which can do the following. They do internet radio shows which are sometimes broadcast on radio. They are looking for a system which does the following for about 5 agents, exactly as they have described it. 1. Take incoming calls 2. Put them on hold if there is no one to handle the call immediately, or transfer them to an available

When I try to move messages between folders I get 'Quota exceeded' messages. I installed dovecot without any major configuration and accounts are created with PostfixAdmin. Is there some common setting that I must be missing? Version in use is 1.0 rc26 - Frank

Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows "(ULAW)" (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody suggest what I might be

I have install dovecot and sieve and it doesn't appear to be working. Below is my dovecot.conf Are there some debugging settings I can use to test if dovecot is linking with sieve properly. Nothing appears to show up in the logs start dovecot.conf ======================= cat /usr/local/etc/dovecot.conf base_dir = /var/run/dovecot/ auth_verbose=yes auth_debug=yes mail_debug=yes

The behaviour of my Asterisk appears to have changed suddenly without any apparent cause. The version is use is 1.4.27.1 When a call is not answered billsec is set to duration, and calls are charged. I can't see any change I could have made to cause this problem. Is it something already known in Asterisk? /voipfc

O.K., so AsteriskNow 1.4.26.2, FreePBX 2.6.0RC2.1 We have a Digium TE205P connected to a single span if ISDN PRI. The Telco has assigned us two local numbers to test incoming calls. I created an inbound route for one of those DID's and assigned it to one of our extensions. Sounds simple enough. Too simple, apparently, when I dial the number the caller gets a recording that it's a

Hi guys, I am having some problems relating to this "Fixing lda user_query error introduces new fault in mail delivery" and I'd like to know if the above wiki comment holds even if a user_query is configured to return the correct mail folder. According to this LDA delivers only to the mail_location setting. Does that mean that it will stick to the value in dovecot.conf or will the

Hello, Could anyone please give some information on configuring asterisk as a gateway. What contents have to add in h.323 .conf and extensions.conf files ? Thanks & Regards Bincy K Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071102/53be51ce/attachment.htm

Hi, On Tue, Nov 12, 2013 at 01:41:29PM -0800, H. Peter Anvin wrote: > On 11/12/2013 01:22 PM, Andrew J. Schorr wrote: > > I am using 4.05 because that's what's packaged in the latest Fedora > > release (Fedora 19 in my case). I wonder why they are using such an > > old version. I will take a look at how difficult it would be to build > > the latest version on

I have installed an Asterisk 1.4 on Suse93 using a FritzCard. Some calls are logged to the ISDN log, but Asterisk is not detecting incoming calls. I wonder whether some other device or process is preventing Asterisk from gaining access to the isdn line? Is there some way to ensure that only Asterisk can listening to the line, or get it to share the line with some other device, such as the fax

The AstLinux Team has released 1.2.0. All current users are encouraged to upgrade as this release addresses the bash "ShellShock" bug. New in 1.2.0: * New Linux Kernel 3.2.x * "igb" ethernet driver for Intel Atom C2000 * Enable AES-NI support * New "sip-user-agent" firewall plugin * New versions of Asterisk 11 and 1.8 * Bash "ShellShock" security fixes A

Hello Asterisk, Back in 2009 I built a small Intel Atom based computer running Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs line and six or so SIP numbers. So basically no load. I'm feeling like it's time to build another machine. It's probably silly, but it's been six years and I can't upgrade the OS which is falling behind. I'd likely just put

I am aware of Astlinux and the other embedded Asterisk solutions out there? Astlinux is nice but the problem is that when I hit a snag and need to incorporate a patch and what not I cannot do that with Astlinux because I cannot compile my own version. How hard is it to create my own version of Linux/Asterisk to run on Compact Flash. I have seen 1GB Sandisk CF for as low as $50 recently so small

Hi all, Has anyone had experience with installing AMP on a soekris box running Astlinux? Is it possible ? Cheers, Callum