Displaying 20 results from an estimated 1000 matches similar to: "Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE."
2012 Jul 13
8
How to set SIP to auto answer in the dial plan .
Hi,
I am trying to write dial plan for sip to auto answer (auto attend) the
incoming call to the sip phone.
- If i call from sip1 to sip2 then sip2 should automatically answer the
call and play some sound file.
I am trying to do this but as new to the asterisk dial plan configuration ,
so not able Todo this.
help me if anyone already done this setup.
Regards
Upendra.
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2012 Jan 12
1
how to set callerid in php AGI file.
Hi,
I am using phpagi for agi scripting. and want to update callerid number but
didn't get any success. please help me how to update PHPAGI is new for me.
Below is the code which I write.
#!/usr/bin/php -q
<?php
set_time_limit(30);
//require(.phpagi.php.);
include("phpagi.php");
$agi = new AGI();
//answer the call
$agi-> answer();
2013 Mar 06
1
Asterisk crashed
Hi,
I am running asterisk 1.8.14.0, It was running fine for last few days and
suddenly crashed today
In logs I can see that abrt tried to save the core dump but it couldn't
Mar 6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac
ip 0000000000533c19 sp 00007f7db9ce3af0 error 4 in asterisk[400000+1d1000]
Mar 6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528
2012 Jun 15
1
Does Asterisk support AMR and AMR-WB
Hi all, I have a project for the 3G related, AMR and AMR-WB support.
I'm using the client develop suite from the PortSIP(http://www.portsip.com),
as their said
support the AMR, AMR-WB with RFC4867.
Now I have to setup a SIP server/SIP PBX in our Lab for test, does the
Asterisk
support these codecs and RFC4867 ? If no, there has any plugin to support
this ?
Also, any other Server/PBX which
2013 Feb 15
6
Cisco 7942 Connected line ID
Hi,
Is it working for anyone?
I have tried with
trustrpid=yes
sendrpid=yes/pai
but can not get it working, Asterisk cli shows prevented message like this.
Connected line update to SIP/1231-00000200 prevented
Regards,
Zohair Raza
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2011 Dec 16
1
CDR END TIME in correct in 1.8+
Hi,
I've tested 1.8.6.0, 1.8.4.0 and 1.8.0
I can get proper start and answer time but not the end time of call
<SIP/11-00000000>AGI Rx << GET VARIABLE CDR(start)
<SIP/11-00000000>AGI Tx >> 200 result=1 (2011-12-16 18:34:48)
<SIP/11-00000000>AGI Rx << GET VARIABLE CDR(end)
<SIP/11-00000000>AGI Tx >> 200 result=1 (2011 12-16 18:34:48)
2008 Jan 22
2
Difference between Asterisk and FreeSwitch
what is the difference between FreeSwitch and Asterisk , whitch one is more scalable and reliable?
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2006 Feb 02
3
Slightly OT: OpenPBX.org and Freeswitch
This is slightly OT in that it isn't specifically *-related, but I was
wondering what the members of the * user community felt about these two
subjects. I've been perusing the OpenPBX.org mail list and the current hot
topic is the fact that their project has come to a grinding halt. They are
concerned that they don't have enough people working on their project. They
feel that * has
2011 Dec 28
1
cdr call time
Hi team,
On event of no answer in CDR the starttime and endtime of call remains the same.
Is there any way how can actually track call originate time and call end time.
Thanks
Vinod dharashive.
Sent from BlackBerry? on Airtel
2013 Aug 06
2
Using freeswitch and Icecast
Hi I am trying to use icecast to broadcast a realtime conference from
freeswitch. But I am having a delay like 20 seconds then I reduced it to
12s. But I don't know if somebody can help me how to reduce it as lower as
possible.
Thanks
Jorge
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2012 Nov 19
3
Allowing peers from specific subnet only
Hi;
How I can make my configuration to allow the sip phones only from specific IP addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed to connect for asterisk?
In other words, in addition to be authenticated based on the username and password, it is required that the IP address of the Phone to be from this range. How?
Regards
Bilal
2013 Aug 08
2
Freeswitch with Digium T316 timed out, T316 timed out
Hi
I am trying to deploy freeswitch with Digium TE121 card for my office
setup, but it is continuously showing Signaling is up and channels are
down except D channel.
Our Architecture is like
We have freeswitch installed with libpri1.4 and Dahdi.
I am from India and here we are having E1 trunk.
Dahdi Configuration is
cat system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 7
2011 Mar 23
4
What is the most stable version of asterisk?
1.2? 1.4? 1.6? 1.8?
Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCSA, Security+, A+
Linux+, Network+, Server+
.
www.impalanetworks.com
P: (505) 327-7300
F: (505) 327-7545
2008 Nov 04
5
VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing
This Friday's edition of the weekly VoIP Users Conference call is all
about wideband audio (aka HD Voice) and conferencing. The guest for
this call is David Frankel, CEO of ZipDX a commercial service that
specializes in wideband conferencing. We expect an interesting call
touching on many aspects of VoIP going beyond the traditional phone
service, conference bridges, technical standards,
2013 Aug 07
1
Using freeswitch and Icecast
what-he-said
On 08/07/2013 06:48 AM, Basil Mohamed Gohar wrote:
> On 08/06/2013 07:40 PM, Jorge N??ez wrote:
>> Hi I am trying to use icecast to broadcast a realtime conference from
>> freeswitch. But I am having a delay like 20 seconds then I reduced it to
>> 12s. But I don't know if somebody can help me how to reduce it as lower
>> as possible.
>>
>>
2008 Nov 18
2
Asterisk with or without OpenSER
Hello,
I am running a small installation of asterisk and looking for future
expansion of it to handle thousands of users. From what I read I see that
usually large installation place OpenSER (or similar solution) in front of
Asterisk in order to provide high call rate because "OpenSER does only
signalling while Asterisk does all". My question is: If Asterisk also does
only signalling
2008 Jun 12
2
Reg. setting Domain name on Cento 5 pc
Hi all,
I am running centos 5.1 and I wish to change the domain name and
dnsdomainname of my PC. currently the settings are--
$ hostname
sipx.com
$ hostname --fqdn
sipx.com
$ domainname
(none)
$ dnsdomainname
com
I have searched in the net for tips but everywhere only the hostname change
is provided. I need to change/set the domain name and the dnsdomain name on
my pc to sipx.com and this
2009 Apr 25
1
Callweaver/Asterisk 'outgoing' spool
c/c'd to the Asterisk list as this is probably relevant to Asterisk as well.
My detailed study of the operation of the 'outgoing' directory reveals that
TXFax() does not delete an expired fax batch file (In the 'outgoing'
directory) until after the end of the dial plan execution.
Is there a way to get it to delete any expired fax batch file before the end
of the dial plan
2009 Apr 18
0
Callweaver TXfax queuing
If TXfax is presented with a whole lot of jobs at once, what (if any) is it's
queuing capability?
Michael
2009 Mar 16
3
Asterisk is not designed for University with large user base?
Hello,
I just had a meeting about a pilot project going on in our University, The
project manager has done some research in the past year and concluded that
Asterisk can not scale well to large user base like 10,000 users, thus
Asterisk is not fit for large University environment.
The project manager instead choosed sipX and said it scales well for large user base.
I had an Asterisk running