hi , all
i want to wtite hangupcause to cdr, but both caller hangup and
callee hangup result in hangupcause code 16.
how would i know whether caller or callee or system error hangup the phone?
please help.
thanks!
2010/4/22 Alejandro Recarey alexreca...@gmail.com:
However, as I can see
Asterisk variable hangupcause
Page Contents
* Asterisk variable Hangupcause
o Recommended SIP - ISDN Cause codes (from RFC3398):
o PRI Hangup Codes
o Version notes
o Tip
o Examples
+ Example 1
+ Example
Thanks, I figured it out. I was using 1.4 but have had to move to 1.6.1
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 10 May 2010 17:27
To: Asterisk Users Mailing List -
thank you for reply.
but hangupcause cant different whether caller hangup or callee hangup?
above two situation both return 16.
2010/5/11 Vardan hvarda...@gmail.com:
Asterisk variable hangupcause
Page Contents
* Asterisk variable Hangupcause
o Recommended SIP - ISDN Cause
Can you show your dialplan part for that call and log also please
Thanks
Zhang Shukun wrote:
thank you for reply.
but hangupcause cant different whether caller hangup or callee hangup?
above two situation both return 16.
2010/5/11 Vardanhvarda...@gmail.com:
Asterisk variable hangupcause
Thanks Vardan,
I will like to know if this scenario can work when peer is not having fixed
ip and we use
host = nasir.server.com
?
also I have set insecure=invite,port
what if i use
insecure=no
thanks again.
Message: 24
Date: Tue, 11 May 2010 10:52:14 +0500
From: Vardan hvarda...@gmail.com
Hello Mailinglist,
we have a Xorcom Astribank which is installed at our client and working since
10/2008. Suddenly last week there where odd behaviors, no calls going through
(it is used to route calls to GSM Gateways).
Upon testing (reinitialization of the complete gear) it turned out that
Hello there,
I have successfully installed and configured asterisk for use as an
office PBX using SIP trucks and Voip handsets (using g.729 codec)
which works great.
Now I wish to try and configure asterisk to do a HTTP request and
submit callerID to an external website when a call is missed. eg
Hello
Yes, you can just remove insecure line, if with out this line is worked
by default insecury=no, so if you not write this line, it will be NO
Also you can use hostname in host field:
===
host =
Hi!
I think Asterisk will detect the dtmf for you and the speach recognition
will detect speach.
That's what I was hoping could be done. How do you set up the dialplan to
have both of those functions run simultaneously?
Look at SpeechBackground() that comes with Asterisk.
Look here:
this is dialplan:
exten = 123,1,Dial(SIP/1000,10,L(1))
exten = 123,2,NoOp(HANGUPCAUSE is ${HANGUPCAUSE})
this is the log which hangup by caller:
== Using SIP RTP CoS mark 5
-- Executing [...@95040:1] Dial(SIP/1001-0031,
SIP/1000,10,L(1)) in new stack
-- Setting call duration
On Tue, May 11, 2010 at 11:09:05AM +0200, Raimund Sacherer wrote:
Hello Mailinglist,
we have a Xorcom Astribank which is installed at our client and
working since 10/2008. Suddenly last week there where odd behaviors,
no calls going through (it is used to route calls to GSM Gateways).
That is normal.
In any case, if the one of call party is hangup, then you take the
hangupcase 16 - normal call clearing.
But if you want to see who is was terminatet the call, you must look in
h in boot leg of the call, incoming and outgoing. and so you can see who
is hangup the call.
I think
Issuing HTTP request from dialplan is simple: Use System call when you
have all the statuses:
exten = _X.,n,System(curl -d number=${EXTEN},status=${STATUS}
http://mywebsite/)
Check your dialplan when you have to issue the command
and
man 1 curl
Good luck
On Tue, May 11, 2010 at 12:21 PM, Zhang
Hello all,
i have asterisk installed in our call centre and we work 24h in day with
this server ,the problem is each day in the night the server hangs and the
calls stopped
And i must to restart asterisk with this command “service asterisk restart”
When i make service asterisk start i got
I upgraded to 1.6 and tried F and it didn't do the same as the g option. I
will have to use the h extension to finish the logging.
Thanks
Lee
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Sent: 10 May 2010 14:45
Execute such commands with cronjob every night:
/etc/init.d/asterisk stop
sleep 3
killall -9 safe_asterisk
killall -9 asterisk
/etc/init.d/asterisk start
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
From:
Am Dienstag, 11. Mai 2010, um 12:36:41 schrieb Motiejus Jakštys:
Issuing HTTP request from dialplan is simple: Use System call when you
have all the statuses:
exten = _X.,n,System(curl -d number=${EXTEN},status=${STATUS}
http://mywebsite/)
Check your dialplan when you have to issue the
Hello
I think this is not right way :)
Look the log's files, find the problem and resolv
The cronjob is the way to stay fat always online, until you find the
problem :)
Vardan
Mindaugas Kezys wrote:
Execute such commands with cronjob every night:
/etc/init.d/asterisk stop
sleep 3
killall
Danny Nicholas wrote:
The file that should get the change is /etc/asterisk/chan_dahdi.conf. you
can apply to one channel by doing this
Hanguponpolarityswitch=yes
Channel=1
Hanguponpolarityswitch=no
Channel=2-x
All assume the source PBX does reflect the hangup. Not all do.
If the
Check Asterisk changelog
(http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.31) with
ctrl+f deadlock.
Guess how many deadlock related bugs wonderful Digium programmers will solve
in future releases?
My proposition is not solution to the problem, its the survival guide in
Hello everyone,
I haven't said much here about the project for awhile now but I think
this is worth a mention. After about 18 months of development, version
2.0 of the software is available!
- http://www.askozia.com/news/2010/5/4/new-release-200.html
For those unfamiliar with the project,
Look at SpeechBackground() that comes with Asterisk.
Look here:
http://www.lumenvox.com/help/speechEngineAsterisk/development/dtmf-and-
speech.htm
When you call SpeechBackground() to perform speech recognition, Asterisk
listens for both speech and DTMF entry. As soon as it detects a
Here is a snippet from my lumenvox dialplan (works pretty much the same for
Vestec)
exten =
s,n,Set(GRAM=/etc/asterisk/grammars/yesno_+${CHANNEL(language)}+.gram)
exten = s,n,SpeechLoadGrammar(yesno|${GRAM})
exten = s,n,SpeechActivateGrammar(yesno)
exten = s,n,Set(SPEECH_DTMF_MAXLEN=1)
exten =
On Mon, May 10, 2010 at 3:43 PM, Jaap Winius jwin...@umrk.to wrote:
No, I'm afraid you misunderstand. This has nothing to do with DNS and
not being able to reach my second PBX -- that's all fine. The
hostname, sip.provider.com, is fictitious anyway.
Try using this line in the [general] section
I use www.melissadata.com, weather.yahooapis.com and Cepstral to give a
current forecast for any 3,6,7 or 10 digit number you enter. The built-in
asterisk sounds offer most of the information you would say; I generate
whatever is missing (city names mostly) with Cepstral.
For your application,
Here is a snippet from my lumenvox dialplan (works pretty much the same for
Vestec)
Thanks for the confirmation and sample.
Sorry to be dense, but you're saying that the DTMF comes back in
SPEECH_TEXT(0)? What about SPEECH_SCORE in that case? And what's the
exact difference with Vestec since
Lumenvox and Vestec will both handle this with this caveat - the number has
to be spoken as single digits, IE 5000 is 5 0 0 0 not 5 thousand.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
All input (DTMF and voice) processed by SpeechBackground are returned to the
dialplan in SPEECH_TEXT(0). The grammar controls how speech is returned.
When you press DTMF instead of speaking you typically get a speech score of
999 or 1000 (a quick and dirty way of knowing that you pressed one
Hi,
My queue members use Local channels and their queue state is In use while
their hint value is Idle.
Since I have Ringinuse=no, I'm experiencing issues such as incoming calls
waiting too much because the agent's phone isn't ringing even though it's
idle/free.
I read somewhere that this is
Hi all,
Could someone please tell me what is the relative cost in using conf files
oppose to the astdb? Basically I need to match a name to a phone number in
order to have all users registered by name and not by number (which I
understood is not a good practice). I have 2000 users and a complex
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Date: Tue, 11 M
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-- Bandwidth
Hi all,
In order to use the open url function of zoiper (it opens an url
based on the asterisk $callerid(dnid)), I need rewriting of the dnid.
In my dialplan I have:
exten = 1000,3,Set(CALLERID(dnid)=newdnid)
exten = 1000,4,Noop(${CALLERID(dnid)})
exten = 1000,5,Queue(test-queue)
but the
On Tue, 11 May 2010, Danny Nicholas wrote:
For your application, the best (IMO) strategy would be to have an AGI
that is cronned to run every X minutes and launch a call when needed
using AMI.
If the script or executable is started by cron, it is not an AGI.
--
Thanks in advance,
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Message: 7
Date: Tue, 11 M
Jerry Geis wrote:
I am using asterisk connected to shoretel pbx using SIP trunk that
then is connected to PRI to the world.
connection is there and I can make calls out.
HOwever, as soon as I place the call with a call file it is telling me
the call was answered and my cell phone
has not
Hi Oliver,
I haven't tried exactly your setup, but:
1. dss1 patton settings is ok for me for european ISDN.
2. have you installed libpri? what about dahdi show status? and dahdi
show channels?
Try module reload chan_dahdi.so. It should say
-- Reloading module 'chan_dahdi.so' (DAHDI Telephony
Good Catch Steve - us Perl Weinies use module and AGI interchangeably.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, May 11, 2010 10:41 AM
To: Asterisk Users Mailing List -
On Tue, May 04, 2010 at 08:46:39AM +0100, Steve Howes wrote:
On 4 May 2010, at 03:44, Jack Bates wrote:
We recently got VoIP, so when we make a call, Asterisk should first try
to make the call with VoIP, but in case either our VoIP or our internet
service are down, Asterisk should then try
- Jack Bates ms...@freezone.co.uk wrote:
On Tue, May 04, 2010 at 08:46:39AM +0100, Steve Howes wrote:
On 4 May 2010, at 03:44, Jack Bates wrote:
We recently got VoIP, so when we make a call, Asterisk should
first try
to make the call with VoIP, but in case either our VoIP or our
On Wed, 2010-04-28 at 11:07 -0500, Danny Nicholas wrote:
FWIW, I would take your STDERR references and give them another handle,
since you're not really trying to produce a CLI/Console output.
The symptoms you have described in this thread are 100% compliant with AGI
protocol violation
On Wed, 2010-04-28 at 11:47 -0400, Fred Posner wrote:
For a AGI that is called repeatedly, maybe you should consider
implementing it in a compiled language.
You can execute XXX AGIs written in C in the time it takes to load the
Perl interpreter and parse your script.
Yes agreed but I
Hi folks,
I have a E1 interface and i need to get some informations about it,
like:
Informations about the Layer 2 (state - active, inactive)
HDLC messages (time, channel, events ...)
Messages from Layer 2 and 3 (Rec. Q.921 and Q.931)
Number of sent messages and bytes, received
Jerry Geis wrote:
Jerry Geis wrote:
I am using asterisk connected to shoretel pbx using SIP trunk that
then is connected to PRI to the world.
connection is there and I can make calls out.
HOwever, as soon as I place the call with a call file it is telling
me the call was answered and my
Greetings all-
I have a handful of Asterisk 1.4.x installations where users dial 'outbound
calls' to the PSTN even though the destination is on the same Asterisk box or
on another Asterisk box on the same network. Instead of paying twice for the
call to go out to the PSTN on one channel and
Hi all,
In order to use the open url function of zoiper (it opens an url
based on the asterisk $callerid(dnid)), I need rewriting of the dnid.
In my dialplan I have:
exten = 1000,3,Set(CALLERID(dnid)=newdnid)
exten = 1000,4,Noop(${CALLERID(dnid)})
exten = 1000,5,Queue(test-queue)
but the
Anybody know a reliable fax solution for 1.4.30 branch?
I am using PikaFax on another server and works very well (about 3000 faxes
a week), but it appears they no longer offer their product to open source
asterisk, only for there WARP appliance.
NOT really looking to migrate from 1.4.x
William Stillwell (Lists) wrote:
Anybody know a reliable fax solution for 1.4.30 branch?
That would be HylaFAX+ along with iaxmodem
http://hylafax.sourceforge.net
http://iaxmodem.sourceforge.net
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
I installed the Vestec system and am testing out using it to get strings of
digits (e.g. conference numbers). The sample grammer just allows saying
zero, but almost everybody will read it it oh. But when I try to
add that as an alternative in the grammer (either the word oh or
phonetically as
Free Fax for Asterisk works pretty well, but not knowing your trunk (DAHDI,
E1, ??) and considering this volume, I would consider something like the
MyFax service.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
This one works on my box (Vestec on 1.4.30 on OpenSuse)
#ABNF 1.0;
mode voice;
language en-US;
// The lumenvox tag format tracks the current working draft of
// the W3Cs semantic interpretation proposal.
// 1.0 corresponds to the working draft released on 01 April 2003
tag-format semantics/1.0;
This one works on my box (Vestec on 1.4.30 on OpenSuse)
Hmm... Not for me.
$Digit = (ONE:1 |
TWO:2 |
THREE:3 |
FOUR:4 |
FIVE:5 |
SIX:6 |
SEVEN:7 |
EIGHT:8 |
NINE:9 |
(OH|ZERO):0);
This is basically the first thing I tried. At least for my voice, this
gets whole lot of spurious 0's.
Tim Nelson wrote:
Greetings all-
box on the same network. Instead of paying twice for the call to go out to
the PSTN on one channel and back in on another channel, I'd like the ability
to lookup the destination number in a MySQL database
I use the mysql add-on, I'd create a subroutine
Doug Lytle wrote:
Tim Nelson wrote:
Greetings all-
box on the same network. Instead of paying twice for the call to go out to
the PSTN on one channel and back in on another channel, I'd like the ability
to lookup the destination number in a MySQL database
I use the mysql add-on,
Is anybody using checkbox.cc to make iax2 calls?
They have recently did some changes and my calls no no longer go through.
They don't have a best service either, not replying to emails..
--
Joseph
--
_
-- Bandwidth and
Joseph wrote:
Is anybody using checkbox.cc to make iax2 calls?
They have recently did some changes and my calls no no longer go through.
They don't have a best service either, not replying to emails..
I don't know about that company, but since you are sounding unhappy with them,
have
lists-asterisk-us...@yoinks.net wrote:
Joseph wrote:
Is anybody using checkbox.cc to make iax2 calls?
They have recently did some changes and my calls no no longer go through.
They don't have a best service either, not replying to emails..
I don't know about that company,
On 05/11/10 18:31, John Novack wrote:
lists-asterisk-us...@yoinks.net wrote:
Joseph wrote:
Is anybody using checkbox.cc to make iax2 calls?
They have recently did some changes and my calls no no longer go through.
They don't have a best service either, not replying to emails..
I don't
On 05/11/10 17:22, lists-asterisk-us...@yoinks.net wrote:
Joseph wrote:
Is anybody using checkbox.cc to make iax2 calls?
They have recently did some changes and my calls no no longer go through.
They don't have a best service either, not replying to emails..
I don't know about that company,
Hello,
I want to modify asterisknow distribution by adding, removing or editing
software.
How can I do that and recompile a new distribution and put it in a new iso.
Thank you.
--
_
-- Bandwidth and Colocation Provided
Make it say 'zed'.
It will make the British happy, and cause a different kind of
confusion for the Americans.
On Tue, May 11, 2010 at 4:09 PM, Richard Kenner ken...@gnat.com wrote:
This one works on my box (Vestec on 1.4.30 on OpenSuse)
Hmm... Not for me.
$Digit = (ONE:1 |
TWO:2 |
THREE:3
Ummm, zed is z.
I was thinking of nought.
On Tue, May 11, 2010 at 8:39 PM, David Backeberg dbackeb...@gmail.com wrote:
Make it say 'zed'.
It will make the British happy, and cause a different kind of
confusion for the Americans.
On Tue, May 11, 2010 at 4:09 PM, Richard Kenner ken...@gnat.com
On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
Anybody know a reliable fax solution for 1.4.30 branch?
I am using PikaFax on another server and works very well (about 3000 faxes
a week), but it appears they no longer offer their product
Joseph wrote:
On 05/11/10 18:31, John Novack wrote:
lists-asterisk-us...@yoinks.net wrote:
Joseph wrote:
Is anybody using checkbox.cc to make iax2 calls?
They have recently did some changes and my calls no no longer go through.
They don't have a best service either, not replying to
On Tue, May 11, 2010 at 11:32 AM, Carlo Dimaggio jaasmail...@gmail.com wrote:
Hi all,
In order to use the open url function of zoiper (it opens an url
based on the asterisk $callerid(dnid)), I need rewriting of the dnid.
In my dialplan I have:
exten = 1000,3,Set(CALLERID(dnid)=newdnid)
On Tue, May 11, 2010 at 04:42:30PM -0600, Joseph wrote:
On 05/11/10 17:22, lists-asterisk-us...@yoinks.net wrote:
Joseph wrote:
Is anybody using checkbox.cc to make iax2 calls?
They have recently did some changes and my calls no no longer go through.
They don't have a best service
I'm pretty sure you want it to say naught to make the british happy, for
zero anyway...
j
On Tue, 11 May 2010, David Backeberg wrote:
Make it say 'zed'.
It will make the British happy, and cause a different kind of
confusion for the Americans.
On Tue, May 11, 2010 at 4:09 PM, Richard
On 05/11/10 21:05, Barry Miller wrote:
[snip]
checkbox.cc is not customer friendly.
CallWithUs dropped IAX when they converted to FreeSWITCH.
But I still use them at home, and their email-only support and call
quality remain good.
--
Barry
I just got a reply form callwithus:
copy
Thank you! Motiejus Jakštys and Sebastian Denz
it's helpful!
2010/5/11 Sebastian Denz asterisk-us...@gonicus.de:
Am Dienstag, 11. Mai 2010, um 12:36:41 schrieb Motiejus Jakštys:
Issuing HTTP request from dialplan is simple: Use System call when you
have all the statuses:
exten =
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