similar to: Simple question about SIP community

Hi to all, to manage properly a call center for multiple companies is possible to let the X-lite/X-Pro softphone to display the number or context called from PSTN to let operator answer with the correct name of the company?? I explain better. If a call come from PSTN to Number A for company A i want the operator recognize it and answer "Good Morning, I'm Operator of company A"

Hi, i read the * support ringing multiple devices at the same time, i inserted this line on my configuration on default context: exten => s,1,Dial(SIP/260&SIP/261&SIP/262&SIP/263|30) exten => s,2,Voicemail,u260 exten => s,3,Hangup And i have both 4 clients in sip.conf . The problem is that if i call it fall immediately in the Voicemail if the client 260 is not registered .

Hi to all, i saw that in chan_sip there is the possibility to let the * to take the number from the Remote-Party-ID header field on incoming calls from gateway. What about to let the * to generate the Remote-Party-ID on outgoing calls? this is is useful for us to let the users to have their outgoing number hidden but let our switch to get the correct record for accounting. I think that If i hide

Hi to all, i already searched the archive without finding a solution to my problem. I have asterisk installation 1.2.18 to support multiple virtiual PBXs. I use SIP peer in the format <ID>-<EXT> to let every virtual PBX to share the same numbers of EXT. Ex. (PBX ID 10 Extensions) 10-101 10-102 10-103 (PBX ID 20 Extensions) 20-101 20-102 20-103 I use some rules in the dialplan to

Hello, I'm glad to announce that DeStar 0.2.2 version has been released. This release contains a large number of bugfixes and new features, see CHANGELOG.txt for the full list. You can find it in the usual place: http://developer.berlios.de/project/showfiles.php?group_id=2112 Thanks for using DeStar, Santiago Ruano Rinc?n http://destar.berlios.de -------------- next part -------------- A

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

Hi, notice that i have Grandstream phones and i have the problem if i activate the Send Anonymous function on them. If i do not activate that option the ACCOUNTCODE is correctly populated. SO i think it may be a bug of asterisk. I'm using Asterisk CVS-HEAD-10/07/04-18:07:25 . Thanks, Bye, Marcello

I'm pleased to announce that Flash Operator Panel 0.26 has been released! FOP is a GPL'd switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in FreePBX, Asterisk@Home, DeStar, startShop, and several other projects both free and commercial. You can grab the

Hi to all, can you suggest to me the best way to avoid problems in the CDRs for anonymous sip calls? I have some peoples that set Send Anonymous : Yes in their Grandstream phones and i don't receive the username as phone number that i use to make billing. It is empty. The only place where there is the phone number is in the peer name where it write the name of the peer that in this case is

Is there a version of the pv_ops kernel that will do HVM and work with fc11? I have been failing miserably trying to get it to work. I have followed Boris''s guide at: http://bderzhavets.wordpress.com/2009/06/10/setup-fedora-11-pv-domu-at-xen-3-4-1-dom0-kernel-2-6-30-rc6-tip-on-top-of-fedora-11/ But have had no success so far. I am wanting to know what is the version of the pv_ops

On Fri, 2019-12-06 at 12:22 +0000, Rowland penny via samba wrote: > On 06/12/2019 11:47, Marco Gaiarin via samba wrote: > > Mandi! Rowland penny via samba > > In chel di` si favelave... > > > > > You cannot create an ldap filter using the above, you would have > > > to filter > > > the result of the ldap search. > > > > I can

People, I have a Debian box with Asterisk and I've installed the Destar package in order to get web managing of my voip system. After I installed Destar, it runs on "localhost:8080", but my server does not have X-Window to access to it so I can engter the web interface.. So how can I change localhost:8080 to IP_ASTERISK:8080 in order to access Destar via web from another PC ???

I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI

Good day Im look at http://www.voip-info.org/wiki-Asterisk+GUI And I see there are a few GUI for asterisk What do you guys prefer? What is the best and simplest? Id like something that give me access to backend for a little bit of customization Thanks for you help and time -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all I would like to know your opinions on free GUI used to manage Asterisk. Which is better? My setup is quite small, about 15-20 phones. I've seen the liste on voip-info. Thanks all. fred -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url :

Is there anyway in asterisk to figure out how much time an agent has spent on the phone? I know I can see total time for a call (inbound or outbound) but where/how do I view queue stats?

Ok, we're trying to use Asterisk as a PBX in our office. Our original plan was to use a Cisco 7960 with a 7914 attached. Short story is, no one updated chan_sccp in a long time and the 7914 is questionable at best anyway from what I've heard. We couldn't ever get chan_sccp to compile, I went to an older version of Asterisk and that broke some of our SIP devices. We tried using a couple

Has anyone gotten this configured to show all extensions vertically instead of filling up the window. If so would you mind sharing your configuration Yes I have tried searching terms like +asternic +op_panel +vertical and a slew of others. Unsucessful though. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g'

I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks

Hi, I finally got Asterisk installed using the German installation CD from http://www.asterisk.de.ms. I got two SIP phones working (SIPPS) asterisk*CLI> sip show inuse Username incoming Limit outgoing Limit 5678 0 N/A 0 N/A 1234 0 N/A 0 N/A