Displaying 20 results from an estimated 2000 matches similar to: "Dial(Local/${EXTEN}@longdistance)?"
2007 Apr 07
2
Different devices for asterisk!!!
Hi all,
Im trying dial a user according to the device s/he uses. i mean if the user
is using asterisk as a peer, then i have to pass the extension in the dial
application like this:
Dial(SIP/${EXTEN}@user) ;so that s/he can perform routing according to the
DNID.
and if the user is using sipura, linksys or grandstream i dial the user like
this,
Dial(SIP/user)
so is there a way to know what kind
2007 Apr 19
1
CDR(dst) != CALLERID(dnid)
Hi guys,
i just came to know that CDR(dst) field is set to current extension instead
of the dialed no. i need to set it to DNID because our every user has 5 dids
and i want to show the caller at the end of the month which numbers he
dialed for every call, along with other cdr info. Our rating depends on the
dialed number also. here is my extensions.conf
exten=> 1212,1,Dial(SIP/rizwan)
2008 Aug 06
2
shared mysql connection in dialplan
hi all,
i just finished developing some incoming call features in a macro. that
macro gets executed everytime an incoming call is received and a new mysql
connection is made using the MYSQL cmd in dialplan. i want to use a single
mysql connection for every incoming call.
my idea of doing it is like this, i want to get a mysql connection in a
global variable, just to share the connection with
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 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 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
2008 Nov 17
2
two sip listening ports for single asterisk
Hi all,
We are planning to shift our sip users from one platform to another.
(basically from one asterisk server to another). the problem we are facing
is that both asterisk servers are using different ports to listen for sip.
and both have live customers on them. provisioning their ata's is not a
good option for us coz of our settup. we cant just ask the customers to
change their ports for
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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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
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 Jan 20
1
Connecting 2 asterisk servers
hi all,
actually i have partially connected the 2 servers but there is a problem.
2 servers A and B
server A forwards call to server B without any problem
but when i try to forward call from server B to A, server shows the
following error on the cli
WARNING[7751]: app_dial.c:1081 dial_exec_full: Unable to create channel of
type 'SIP' (cause 3 - No route to destination)
== Everyone is
2007 May 09
3
The 'h' extension problem
Hi all,
There is a problem with my dialplan. here is the dialplan:
exten=> 123,1,Dial(SIP/U1,,Ttg)
exten=> 123,2,Hangup
exten=> h,1,AGI(onhangup.pl)
The problem is whenever U1 is called or calls someone, if U1 hangsup the
call then the h extension is NOT executed. but if the other person hangsup
the call, then the h extension is executed (assuming that the other person
is calling
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
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
2013 Nov 21
3
Call files without permission for asterisk to read
Hi all,
I am syncing call files on my secondary asterisk server but without
permission to read for asterisk. So they should be executed when I grant
the right permissions (thats when my primary asterisk server crashes or
shutsdown somehow). But asterisk only tries to read the file at the time of
placing the file. So when i grant right permissions nothing happens. Is
there any workaround to this
2007 Aug 15
1
why is nonce="584760da" used in sip packets?
Hi all,
There is a parameter called "nonce" included in every register request that
a UA sends to asterisk. I have read sip debug a lot and only found out that
the "nonce" parameter value which is used in register request was generated
by asterisk server in a previous sip response.
As you can see in the sip debug (labled in red).
<--- Transmitting (NAT) to
2007 Mar 14
4
what happened to asterisk wiki???
Hi
im trying access the www.voip-info.org website since yesterday but i cant
open it. google search diaplay correct search results but it doesnt open
when i click the link. it displays a message about tcp error which says
-->"There was a problem communicating with the server". I dont know what the
problem is. I just want to ask whether their server is down or not and is
everybody
2013 Oct 31
3
Realtime Call Files
Hi all,
Is there any way of originating calls in future without using call files?
We have 2 servers (1 active at a time). If we use call files with
modification date in future, on the 1st server and it dies and, we activate
the second server but we lose the call files.
I could have a cronjob on both servers and create callfiles reading
execution time from database, but this involves some other