search for: jb_soft

Hi all, I have tried to install Wanpipe 7.0.20 in a fresh-installed Debian 8.5 (Jessie) server, but when I reach to the point where I have to enter the Linux Headers directory *(installed in default with ?apt-get install linux-headers-$(uname -r)?)*, I get an error, the package refuses to compile and installation aborts. If I try the same process with Debian 7 (Wheezy), installation works just

Hi, Thanks for your info, What is the impact of the following line in dialplan, Dial(SIP/19201/19202,300) On Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes <vinicius at aittelecom.com.br> wrote: > You might want to use the Originate() application instead. Check its usage > by issuing the command 'core show application originate' on Asterisk CLI. > > 2015-09-03

Hi AJ, Am 06.06.2016 um 10:14 schrieb A J Stiles: > But why not call an AGI script, have this check the caller ID against a MySQL > database and return a status -- blocked or not -- in a variable? Then you can > manage individual number blocking in a much cleaner, more extensible fashion. > > Feel free to ignore me if it sounds like I'm suggesting you walk all the way > to

I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk). VoIPtalk calls are unauthenticated and reach me fine, but Andrews & Arnold calls are authenticated. The last call I successfully received was on Tuesday afternoon. Initially, A&A were for some odd reason not sending calls to my server, but that has been resolved. The problem now is that the calls fail to

...The hardest thing about this was extracting all the numbers of previous callers from the CDR and adding it to the Previous_Callers AstDB for the lookup. I didn't want to make existing callers go through the initial learning process. -- Best regards, Julian mailto:jb_soft at trink.co.uk

Hi, I created an account but when I go to issues.asterisk.org <http://issues.asterisk.org/> It still asks for a client certificate. See this screen shot, hopefully it showswhat I mean. http://firestar-hosting.com/clientcert <http://firestar-hosting.com/clientcert>.png -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying set up asterisk on an amazon instance in Sydney. It's to use for our kids in Sydney to connect with their friends in the States. We've found iax works better than sip with these distances. But we now have weird problem: everybody has a cell phone, and it's much cheaper/better to use the house internet connection over the phones wifi. Each cell phone has it's own

Hello, I would to allow users to place calls overseas such as India and Malaysia but only with a security code. if they don't have a security code I want to be able to drop the calls. can someone point me to a right direction to achieve this goal? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

This is exactly what I do - “press 1 for a human” Works great ________________________________ From: asterisk-users <asterisk-users-bounces at lists.digium.com> on behalf of Greg Woods <greg at gregandeva.net> Sent: Thursday, December 12, 2019 6:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Block Spam Calls Most spam calls are

>>> Nobody has any information or opinions on any of this? Personally, I don't think MACROS are going anywhere any time soon, so I have not bothered looking into a substitution. As for ael; I've never used it. Doug

Hi, I need dialplan to set INCOMING call forwarding during lunch break to my secretary. I want that I can set call forwarding by dialing an extension number to turn it ON or OFF. I am using asterisk 11. Thanks Abdullah Faheem -------------- next part -------------- An HTML attachment was scrubbed... URL:

I was successful in compiling asterisk in raspbien except for the following error If I enable the gsm codec. It appears there is something in the Makefile n this directory that needs to be changed. Probably involving optimization. Not sure why it does not recognize the processor since it is one that is mentioned in the Makefile.? Any help would be appreciated. make[2]: Entering directory

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes