Displaying 20 results from an estimated 1000 matches similar to: "asterisk security....again"
2011 May 06
7
Background music during a call
Hi All,
I am in desperate need of this feature. I want to play background music
during a call while the 2 parties are having some lovely conversation (or
maybe give them a sort of cursing background if they are cursing each
other). I found this post which talks about creating a ghost call with the
help of queues and putting that queue in a meetme room where queue will play
the song/curse and the
2011 Apr 28
1
odbc error - server is gone
Hi list,
yesterday I converted my voicemail.conf to realtime voicemail and also
configured to store the voicemessages in a database using odbc as described
here <http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail> and
here <http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage>.
I am using asterisk 1.4.2 with mysql. I also installed the proper odbc
driver for
2011 Apr 04
2
call forwarding
Hello list,
i have one question related to call forwarding.
i have 2 number for the inbound and i want to configure asterisk like that.
When the customer call the first number 0522XXXXXX the call will be
forwarding automatically to anther number 0520xxxxxx
Does anybody have a solution to this problem.
Thanks and Regards.
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2007 Aug 17
4
Call Limits
Hi all,
Some of my asterisk users have used their maximum call limit for incoming
calls (peers). There incoming call limit should automatically reset to zero
after hangup but its not happening and they no longer can recieve any calls
as their allowed limit is already full. So is there any way to reset the
call limit on peers by commands or do i have to restart my asterisk server?
--
Best Regards
2007 Oct 24
2
Remote provisioning for ATA's
Hi all,
I need a fully developed web based remote provisioning system. I cant find
anything reliable on the internet. Have already checked ataconfig.com and
voxilla-ays.com. have tried to contact them but got no response. So if
anybody knows a good provisioning system then plz tell me about it.
--
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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2008 Jan 25
1
Need sample configuration files for sipura/linksys ata
Hi all,
i need sample xml configuration files for linksys pap2, linksys pap-2t,
sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are
linksys/sipura products. So if anyone has these sample files then plz share.
--
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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2007 Sep 11
3
Prevent multiple sip registrations
Hi all,
Is there anyway i can prevent multiple sip registrations from different IPs
using single username in asterisk. Does asterisk provide any aid in this
respect? As far as my knowledge is concerned i dont think there is any
support for this in asterisk, so i think i'll have to makeup a script which
sniffs sip packets coming for asterisk and detect for multiple register
requests coming from
2007 May 30
12
False ring problem
Hi all,
when a user dials any number, asterisk automatically generates ringing which
caller can hear, and after 2 - 3 rings asterisk detects that the called user
is busy, then caller hears busy tone. for example user hears---
tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing
at the start so that user hears only beep beep beep if the called user is
busy. I have used the R
2007 Aug 09
1
strange warning
Hi all,
I am using an asterisk as a client to connect to another asterisk server by
registering with the register string. Registration is done without any
hassel, but after sometime my asterisk loses the registration with the
server and the server starts displaying the following msgs repeatedly:
[Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth,
but based on stale nonce
2011 Feb 28
5
Failover Routing
Hi,
I am doing failover routing based on 2 dial commands. First route sends back
4xx response and I don't want it to try 2nd route when it is 4xx response.
Can we do failover routing based on SIP 5xx response only ?
Thanks
Deepika
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2007 Oct 29
2
XML file for spa devices
Hi all,
i need an XML file format which is used in remote provisioning of different
spa devices. Please somebody tell me the format or tell me where can i find
it on the internet. I also need a list of parameters which are configured
using auto-provisioning.
--
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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2011 Mar 15
1
call being rejected
I am using asterisk 1.8.3.
I am getting this error:
[Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite:
Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected
because extension not found in context 'smvoice-mediaport'.
"dialplan show" gives me that the context is present:
[ Context 'smvoice-mediaport' created by
2011 Mar 05
1
2 ip phones and 1 normal, can't neither send nor receive calls at all...
I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco
spa8800, all them are internal lines.
1.- spa921, 401 ext
2.- spa921, 402 ext
3.- normal phone connected to spa8800 404 ext.
It had a very strange behavior when I was configuring call transfer and call
pickup.
These are steps to repeat it:
1.- from 401 call to 404
2.- from 404 don't answer it.
3.- from 402 press *8
2011 Feb 24
2
Carrying context from one server to another?
The relevant part of my setup is something like:
SIP phones -> local server -> remote server -> SIP-to-PSTN provider
I want _some_ of the SIP phones on the local server to be able to get
access to SIP-to-PSTN, but not all of them. The local-to-remote
connection is IAX2 over VPN.
Do I need to set up two separate IAX2 connections, one "privileged" and
the other not, or can I
2007 Aug 01
3
How to use stun server?
Hi all,
This is the first time i am using stun with asterisk for nat problems. I
have read the rfc which describes how stun works. i didnt have any problems
understanding it. I have also intalled the stun server called stund which i
downloaded from sourceforge. I have seen on the list that most people use
stund here. I have started the stun server and its running silently. Now i
dont know what to
2011 Mar 24
1
Fwd: Asterisk 1.6.2.10 & CDR custom added field
Hello,
is there anyone who can point me to correct information ?
Following http://pbxinaflash.com/forum/showthread.php?t=9042 and
http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql > Extending CDR
does not result in a working environment for me.
Any feedback appreciated.
Kind regards,
Jonas.
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Subject: [asterisk-users] Asterisk 1.6.2.10 & CDR
2007 May 31
2
How to read SIP debug?
Hi all,
i need to study the SIP protocol. can anybody tell me about any ebook which
could halp me understand the sip protocol, architecture, and how to read and
understand the sip signalling when i use "sip debug" in asterisk?
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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2007 Nov 02
2
asterisk as a gateway
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
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2011 Feb 24
1
Unknown calls
Hi there everyone,
I am a bit confused these days due to some problem I am having. Its not a
technical problem. Asterisk is working fine. Most of the users are happy,
but some handful of users are getting calls in the middle of the night even
though they have enabled "Anonymous Call Rejection (blocks calls with no
caller id on asterisk server)" and TIMED DO NOT DISTURB which also blocks
2007 Aug 23
1
channel not hungup (zombie?) so call limit not reset to zero
im having a strange problem related to call-limit for peers. well im not
sure if its related to call-limmit or not. Bottom line is:
I call a user A, from user B. user B hears silence, untill it goes to
voicemail. when user B hangsup. user B's call limit is reset to 0 but user
A's call limit is not reset.strange thing is user A's status on cli is shown
as NOANSWER, while user B did not