Displaying 20 results from an estimated 900 matches similar to: "How to read SIP debug?"
2007 May 31
4
Context documentation for the newbie!
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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 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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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
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 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 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 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 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 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
2007 Oct 25
3
Getting SIP Response Code from HANGUPCAUSE
I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead?
Doug.
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2011 Feb 28
2
asterisk security....again
Hi all,
The problem I have been experiencing since last month is that some of my
customers are getting calls with "Asterisk <Unknown>" caller id. Most of
them in the middle of the night. And my asterisk server has no record of
these calls. The customers were getting irritated as you can imagine. I
guessed the only way to receive incoming calls by by-passing the
registration server
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
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
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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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 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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2007 Dec 19
1
IAX for asterisk to asterisk
Hi,
I am not new to asterisk but this is the first time im using iax protocol. I
have always used sip before, but i have heard its better to use iax if i
want communication between 2 asterisk servers. I just registered asterisk
server1 with asterisk server2 and tried to call server1 from 2 but the call
does not pass thru. i dont see any messages on the recieving asterisk cli
but the caller
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