similar to: asterisk-users Digest, Vol 40, Issue 37

Folks, Eric is spot-on--both phones are happy with the modified ${CALLERIDNAME} value since I removed the quotes in the call to SetCIDName(). I've replace my calls to SetCallerID() since I think using SetCIDName() is cleaner. Cheers, Rob > -----Original Message----- > Date: Sat, 29 Jan 2005 16:44:16 -0600 > From: Eric Wieling aka ManxPower <eric@fnords.org> >

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On Friday 09 February 2007 11:50, asterisk-users-request@lists.digium.com wrote: > Anyone got any experiences of good quality VoIP conferencing phones? > > I've used Polycom analogue units in the past, and I see that they have a > SIP version (the IP4000) - but it is better/worse/as good as an analogue > version? > > (ie. would I be better off with an analogue version

I've found OpenVPN to be easy to configure and very robust. It has a zillion options, but they are just that - options. I haven't used it for VoIP, but I've put it to good use doing layer 2 bridging which has eliminated many problems with certain programs traversing NAT and load-balancing routers. I can't think of any reason why it would not work well with Asterisk. > On

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is quite conservative in what they want to use :) meaning handsets are important; As the handsets have

I am stumped In features.conf,I programmed this [applicationmap] Answer0 => 0,self/both,Macro,nway_start But do I pass an argument or parameter to my macro? I tried Answer0 => 0,self/both,Macro,nway_start^0 Answer0 => 0,self/both,Macro,nway_start,0 but the usuar variable ${ARG1} is empty in my dialplan. The issue is that my macro needs to know what key was pressed. -------------- next

From: Doug <Doug at NaTel.net> >"Net Neutrality" is great in principle. But ISP's need to >somehow control those few percentage of users who suck down >a huge majority of the bandwidth. It's dollars and cents. There is a rational solution for the traffic management issue. It just needs to be aligned with the pricing model. I would like to see a tiered

asterisk-users-request@lists.digium.com wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >

Message: 6 Date: Fri, 03 Mar 2006 17:32:47 +0000 From: Conrad Wood <asterisk-users@conradwood.netasterisk-users@conradwood.netasterisk-users@conradwood.netasterisk-users@conradwood.net> Subject: Re: [Asterisk-Users] Problem with NAT!!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

I haven't intentionally done anything to the CID, the odd thing is that another aastra phone (480e) connected to the same port DOES display CID, it's only these PT350s that can't seem to read it.. Been talking to Aastra support & they've suggested some avenues to explore, but none of them have worked... The PT350s definitely have a problem w/*'s cid signals.. Just not sure

On 1/3/07, asterisk-users-request@lists.digium.com < asterisk-users-request@lists.digium.com> wrote: > > ---------- Forwarded message ---------- > From: "Dan Austin" <Dan_Austin@Phoenix.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > Date: Wed, 3 Jan 2007 10:33:02 -0800 > Subject:

Hi, I am looking for the easiest and fastest way to send or pull callerID and extension# from asterisk to a web server for sales data lookup and display. User would be logged in with known extension. I have been looking at several options but was hoping someone here would have the best one. I was looking at using Master.csv but would prefer to avoid parsing it. Any suggestions much

On Thursday 08 February 2007 07:32, asterisk-users-request@lists.digium.com wrote: > Does anyone have any recommendations for a phone that has easy to > understand/implement central provisioning? I've used CISCO 79XX phones, > and they're great (but too expensive). I like Grandstream phones, but > their provisioning sucks. > > What is everybody else using in large

> 1. Create agent phone logins. > 2. Create real-time report to monitor agent login/logout activity. > Should > have the ability to view which agents are currently logged in/out of > the > system. > 3. Create historical report to pull agent activity. Should display > login/logout activity. Be able to pull information by rep and > timeframe. > 4. Create "hold

---------------------------------------------------------------------- Message: 1 Date: Tue, 6 Mar 2007 20:02:07 +0100 From: Olle E Johansson <oej@edvina.net> Subject: [asterisk-users] Building a new voicemail system... Testers needed! To: Asterisk Non-Commercial Discussion Users Mailing List - <asterisk-users@lists.digium.com> Message-ID:

I understand Asterisk is more like a B2BUA. But when this INFO request is sent to asterisk, asterisk is supposed to bridge the request to the other endpoint, right? In what situation, it decides to send a reply; in what situation, it decides to bridge the request? What is the role of gateway in SIP world, a proxy, a B2BUA or something else? Thank you, Wei Date: Fri, 18 Mar 2005 12:51:28 -0600

On 28/07/2010 15:45, shorewall-users-request@lists.sourceforge.net wrote: > On 7/28/10 1:50 AM, Andrea Perdicchia wrote: > >> > Hi all, >> > Is possible log mac address in shorewall? >> > I try all configuration "debug,info..." in /etc/shorewall/shorewall.conf >> > but in /var/log/messages the log show only few information and not mac

> > Thanks Josiah Bryan, > > I do not have any dns server running on my asterisk server, we do have an > external DNS server working in the data center; the IP of this dns server is > 10.4.1.5... > > Following you will see my main configuration: > > /etc/resolv.conf: > > domain localdomain > search localdomain > nameserver 10.4.1.5 > nameserver 10.4.1.2

Post your iptables output ("iptables -nvL") and I'll send you the commands you need OCG -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Wednesday, June 29, 2005 9:17 PM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 11, Issue

Need help...I need to install a card to terminate 7 lines...I have not order the phone lines yet...I can either do analog lines 1FBs or order a fractional T1...any suggestions on what hardware would be easier to install and configure...also if I went with a T1...do I need an external CSU/DSU or anything or does it just plug into the T1 card...thanks.. -----Original Message----- From: