[asterisk-users] sipp application/dtmf-relay not work properly in Asterisk!

[Zhang Shukun]

hi, everyone
 i want to use sipp to auto answer the ivr， to simulate the keypad send
digital sequence， so i try to send DTMF by application/dtmf-relay, but i
have got this error message in the asterisk CLI, Could you help me? Thanks!

[Jun 16 17:11:34] WARNING[26321]: rtp.c:3207 ast_rtp_senddigit_begin: Don't
know how to represent '

the whole CLI message as follows:

<--- SIP read from UDP:211.150.88.154:5067 --->
INFO sip:01025475845@211.150.88.155:5060 SIP/2.0
Via: SIP/2.0/UDP 211.150.88.154:5067;branch=z9hG4bK-3-1-7;rport
From: 1000 ;tag=1
To: 01025475845 
Call-Id: 1-3@211.150.88.154
CSeq: 2 INFO
Contact: sip:1000@211.150.88.154:5067
Event: dtmf
Content-Type: application/dtmf-relay
Content-Length:31

Signal= 11037845
Duration= 100
<->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received:
<--- Transmitting (NAT) to 211.150.88.154:5067 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.150.88.154:5067
;branch=z9hG4bK-3-1-7;received=211.150.88.154;rport=5067
From: 1000 ;tag=1
To: 01025475845 ;tag=as7af6d579
Call-ID: 1-3@211.150.88.154
CSeq: 2 INFO
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<>
[Jun 16 17:11:34] WARNING[26321]: rtp.c:3207 ast_rtp_senddigit_begin: Don't
know how to represent '
-- 
Appreciate your kindly advise and help.
Thanks & Regards
Sucan
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[asterisk-users] How to execute Asterisk Functions in PHPAGI

[Zhang Shukun]

Hi, all
I can use $agi->exec() to excute applications in Asterisk.
such as $agi->exec("Set",abc=1)

But how could i excute Asterisk Functions use agi functions?  for example:

I want to excute SHARED function in PHPAGI. i use
$agi->exec("SHARED",callernum) to do it.

but failed with info:

AGI Rx << EXEC SHARED callernum
-- AGI Script Executing Application: (SHARED) Options: (callernum)
[Oct 19 00:06:13] WARNING[2638]: res_agi.c:1757 handle_exec: Could not
find application (SHARED)

Could you help me?
-- 
Appreciate your kindly advise and help.
Thanks & Regards
Sucan

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[asterisk-users] How to Set Callerid Of Originate a call?

[Zhang Shukun]

Dear all,

as you know, we can use Originate Command to auto-dial a out-bound
call to a extention or app since 1.6.2.

but when i Originate a call, and hangup. the cdr of this call has no
CDR(clid) and CDR(src).

Could you tell me how to set the Callerid to cdr from an Originate
call? I use Originate directly in the dialplan

not AMI, so i can't set the callerid property like AMI use OriginateAction.

my dialplan is :

[test]
exten => 123,1,NoCDR()
exten => 123,n(me),MeetMe(123,AMX,123)
exten => 0,1,Read(DEST,dial,,i)
exten => 0,n,ResetCDR()
exten => 0,n,Originate(SIP/${DEST},exten,test,s,1)
exten => 0,n,Goto(123,me)

exten => s,1,MeetMe(123,M,123)

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Sucan

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Re: [asterisk-users] Could MeetMe invite someone to the conference?

[Zhang Shukun]

Thank you !  Do you konw how to realtime billing for MeetMe conference?

2010/8/30 Paul Belanger :
> On Sun, Aug 29, 2010 at 10:52 PM, Zhang Shukun  wrote:
>> but i want to know if i can invite some one to the conference when i
>> already in the conference?
>>
> http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
>
> --
> Paul Belanger | dCAP
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>
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[asterisk-users] How to Billing for MeetMe Conference?

[Zhang Shukun]

hi,Dear all
as you know, MeetMe has cdr for each attendant. but the fee is
always paid by the moderator. not by each one.
and the the members in the conference are dynamic changed.

in this scenario, how to billing for MeetMe conference? and i want to
hungup all the calls when the account fee of the moderator is not
enough.

thanks for your help!

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[asterisk-users] Could MeetMe invite someone to the conference?

[Zhang Shukun]

hi,all
 i know i use MeetMe, any one has the correct password can dial in.

but i want to know if i can invite some one to the conference when i
already in the conference?

Thanks!

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[asterisk-users] codec_g729.so not work!

[Zhang Shukun]

hi, all
  i want to use g729 codec for set up a call. so i donwloaded the
so file from web site: http://asterisk.hosting.lv/#bin
and install it properly.

*CLI>
*CLI> core show translation
 Translation times between formats (in microseconds) for one
second of data
  Source Format (Rows) Destination Format (Columns)

   g723   gsm  ulaw  alaw g726aal2 adpcm  slin lpc10  g729
speex  ilbc  g726  g722 siren7 siren14 slin16
 g723 - - - -- - - - -
- - - -  -   -  -
  gsm - - 2 2 2000 2 1  3001  3000
- -  2001  1001  -   -   2001
 ulaw -  3000 - 1 2000 2 1  3001  3000
- -  2001  1001  -   -   2001
 alaw -  3000 1 - 2000 2 1  3001  3000
- -  2001  1001  -   -   2001
 g726aal2 -  3999  1001  1001-  1001  1000  4000  3999
- -  3000  2000  -   -   3000
adpcm -  3999  1001  1001 2999 -  1000  4000  3999
- -  3000  2000  -   -   3000
 slin -  2999 1 1 1999 1 -  3000  2999
- -  2000  1000  -   -   2000
lpc10 -  4998  2000  2000 3998  2000  1999 -  4998
- -  3999  2999  -   -   3999
 g729 -  3999  1001  1001 2999  1001  1000  4000 -
- -  3000  2000  -   -   3000
speex - - - -- - - - -
- - - -  -   -  -
 ilbc - - - -- - - - -
- - - -  -   -  -
 g726 -  3998  1000  1000 2998  1000   999  3999  3998
- - -  1999  -   -   2999
 g722 -  3998  1000  1000 2998  1000   999  3999  3998
- -  2999 -  -   -   1000
   siren7 - - - -- - - - -
- - - -  -   -  -
  siren14 - - - -- - - - -
- - - -  -   -  -
   slin16 -  4998  2000  2000 3998  2000  1999  4999  4998
- -  3999  1000  -   -  -

my sip.conf add two account:

[123]
type=friend
username=123
host=dynamic
context=95040
dtmfmode=rfc2833
disallow=all
allow=g729
insecure=port,invite
canreinvite=no
nat=yes

[321]
type=friend
username=321
host=dynamic
context=95040
dtmfmode=rfc2833
disallow=all
allow=g729
insecure=port,invite
canreinvite=no

my extension is :

exten => 321,1,Dial(SIP/321)


when i want to set up a call (123 dial 321). but failed. it says:

 == Using SIP RTP CoS mark 5
[Aug 20 18:23:21] NOTICE[14543]: chan_sip.c:8454 process_sdp: No
compatible codecs, not accepting this offer!

Could you tell me what 's wrong?


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[asterisk-users] How to Record with Konference when it has no record option?

[Zhang Shukun]

hi,list
  i installed App_Konference in my Asterisk 1.6.2.11.
and i write in dialplan like this:

exten => 95040,n,konference(1234,RVxTH)

it works fine. but I want to record the conference, if use MeetMe , i
can use 'r' option to do this.

but there is no 'r' option in konference , Could you tell me how to do this?

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[asterisk-users] MeetMe will record automaticlly even without 'r' option??

[Zhang Shukun]

hi,all
i install MeetMe module on Asterisk 1.6.2.10.
when i use MeetMe to open a conference. even without 'r' option .it
will record too.
is this the bug of this module?

my dialplan is :

[95040]
exten => 95040263007,1,MeetMe(95040,sM,123)

the CLI output is :

*CLI>   == Using SIP RTP CoS mark 5
-- Executing [95040263...@95040:1] MeetMe("SIP/999-0021",
"95040,sM,123") in new stack
   > Starting recording of MeetMe Conference 95040 into file (null).(null).

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[asterisk-users] MeetMe VS. Conference

[Zhang Shukun]

hi, group
 there are two module can used for meeting. MeetMe and
Conference(which is a plugin)

My question is :

which is better for large conference(maybe above 100 people in a meeting)?


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Re: [asterisk-users] Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ?

[Zhang Shukun]

Thanks. it is depends on mysqlclient.so. after i installed this module. it's ok.

2010/7/22 Gareth Blades :
> Zhang Shukun wrote:
>> hi,list
>>       Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ?  after
>> i make and make install. i cant find the .so file.
>>
>> is this mean it can't install on 64bit Cent-OS. ps: it works fine on
>> the 32 bit Cent-OS
>>
>> Thanks very much!
>>
>
> I have a live system running centos 5.5 64bit and asterisk 1.2.6 with
> the addons installed for mysql support etc...
>
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[asterisk-users] Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ?

[Zhang Shukun]

hi,list
  Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ?  after
i make and make install. i cant find the .so file.

is this mean it can't install on 64bit Cent-OS. ps: it works fine on
the 32 bit Cent-OS

Thanks very much!

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Sucan

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Re: [asterisk-users] GotoIfTime problem

[Zhang Shukun]

Thank you very much! Lesher

2010/7/2 Tilghman Lesher :
> On Thursday 01 July 2010 21:59:21 Zhang Shukun wrote:
>> hi, all
>>
>>     recently, i face a GotoIfTime problem
>>
>> GotoIfTime("08:00:00-07:00:00,mon-sun,*,*?95040263008,start")
>>
>> as you can see the section is 08:00:00-07:00:00  , which is the begin
>> time is later than the end time
>>
>> what's this refers then?
>>
>> in my test , my system time is 10:57:00, but this check will pass,
>> although i guess i will not.
>>
>> is begin time later than the end time  means * (all the day 24 hours)?
>
> No, it means 23 hours a day, excluding 7:00 a.m. to 8:00 a.m.
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
>
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[asterisk-users] GotoIfTime problem

[Zhang Shukun]

hi, all

recently, i face a GotoIfTime problem

GotoIfTime("08:00:00-07:00:00,mon-sun,*,*?95040263008,start")

as you can see the section is 08:00:00-07:00:00  , which is the begin
time is later than the end time

what's this refers then?

in my test , my system time is 10:57:00, but this check will pass,
although i guess i will not.

is begin time later than the end time  means * (all the day 24 hours)?

Could you help me ? Thanks
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[asterisk-users] Is Centos 64 bit or 32 bit better?

[Zhang Shukun]

hi, all
after a long time development, i need to deploy a production system.

i want to install latest Asterisk 1.6.2.9 on Centos 5.4 . one thing confused me.

my computer hardware support 64 bit OS.

my question is : should i use Centos 5.4 64bit or  Centos 5.4 32bit?

which is better for my asterisk ? consider compatibilityand stability.

this is a new machine , only used for asterisk, no other apps.

Thank you in advance!
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[asterisk-users] What‘s the best operating sys tem suggest for Asterisk 1.6.2.9

[Zhang Shukun]

hi, list
 i want to know what is the best OS for install Asterisk 1.6.2.9,
which should work properly on working system.

i want to use CentOS5.2 or CentOS 5.4.  Which is better and stable?
Thanks for your help.


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Re: [asterisk-users] NO ANSWER before playback or background function?

[Zhang Shukun]

2010/6/22 Philipp von Klitzing :
> Hi!
>
>> but i want to answer the channel when dial someone and pick up the
>> phone.not play a file.
>
> Search this list for "early media" and maybe also for "progress".

Thanks , i have search for "early media", and have get some valuable infomation.

i can play files with noanswer .

exten => s,1,Progress
exten => s,n,Playback(hello,noanswer)  ;this works good.
exten => s,n,Dial(SIP/1...@bd-test,30)
exten => s,n,Playback(hello,noanswer) ; this works no sound

the first Playback works good. i can hear the sound and it won't
answer the channel first.

my problem is after Dial command, if not answer the channel(connected).

next will execute: exten => s,n,Playback(hello,noanswer) ; this works no sound

but this Playback have no sound.

Do you know what's wrong?



>
> Philipp
>


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[asterisk-users] NO ANSWER before playback or background function?

[Zhang Shukun]

hi,all

i find in asterisk 1.6.2.1, before play a sound file use playback or
background, it will answer the channel first.

but i want to answer the channel when dial someone and pick up the
phone.not play a file.

i know there are some params such as 'noanswer' for playback or 'n'
for background can not answer before play a file.

but it is not always take effect on my tests.as it said:

noanswer: Play the sound file, but don't answer the channel first (if
hasn't been answered already). Not all channels support playing
messages while still on hook.

Could you help me ?

Thanks!

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Re: [asterisk-users] What's diff between ${CDR(start)} , ${CDR(answer)} , ${CDR(calldate)}

[Zhang Shukun]

Thank you.

another quesion is i want to get ${CDR(answer)} and ${CDR(end)} in the
hangup section. i can get ${CDR(answer)} sucessfully but get
${CDR(end)} is null.

i know i can set any variable i want into CDR table if i want .

but i want to know without any setting . which variabes will set automatic?

is that all of the following(i use asterisk 1.6.2.1)?


${CDR(clid)} Caller ID
${CDR(src)} Source
${CDR(dst)} Destination
${CDR(dcontext)} Destination context
${CDR(channel)} Channel name
${CDR(dstchannel)} Destination channel
${CDR(lastapp)} Last app executed
${CDR(lastdata)} Last app's arguments
${CDR(start)} Time the call started.
${CDR(answer)} Time the call was answered.
${CDR(end)} Time the call ended.
${CDR(duration)} Duration of the call.
${CDR(billsec)} Duration of the call once it was answered.
${CDR(disposition)} ANSWERED, NO ANSWER, BUSY
${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc
${CDR(accountcode)} The channel's account code (read-write).
${CDR(uniqueid)} The channel's unique id.
${CDR(userfield)} The channels uses specified field (read-write).



2010/6/18 Tilghman Lesher :
> On Friday 18 June 2010 03:21:32 Zhang Shukun wrote:
>> hi,all
>>    for a long time, i cant understand the difference between
>> ${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)}
>>
>> i know ${CDR(start)} mean when a call is start. and  ${CDR(answer)}
>> means when a call was pick up.
>>
>> but what's  ${CDR(calldate)} mean?
>
> It could mean whatever you want.  CDRs (at least the internal representation)
> have support for arbitrary additional variables.  Whether a particular backend
> has support to carry those over into permanent storage is another question
> (in 1.6.2, most CDR backends have it, as long as the underlying table has a
> column to receive the data).
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
>
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[asterisk-users] What's diff between ${CDR(start)} , ${CDR(answer)} , ${CDR(calldate)}

[Zhang Shukun]

hi,all
   for a long time, i cant understand the difference between
${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)}

i know ${CDR(start)} mean when a call is start. and  ${CDR(answer)}
means when a call was pick up.

but what's  ${CDR(calldate)} mean?


Could you help me ?

Thansk a lot!


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[asterisk-users] Why Manager account log on and log off alternatively all the time?

[Zhang Shukun]

  hi, guys,

 when i create a manager account used for freepbx, the follow info
produce all the time?

do you know that's the reason?

== Manager 'bitzsk' logged off from 127.0.0.1
  == Manager 'bitzsk' logged on from 127.0.0.1
  == Manager 'bitzsk' logged off from 127.0.0.1
  == Manager 'bitzsk' logged on from 127.0.0.1
  == Manager 'bitzsk' logged off from 127.0.0.1
  == Manager 'bitzsk' logged on from 127.0.0.1
  == Manager 'bitzsk' logged off from 127.0.0.1
  == Manager 'bitzsk' logged on from 127.0.0.1
  == Manager 'bitzsk' logged off from 127.0.0.1
  == Manager 'bitzsk' logged on from 127.0.0.1
  == Manager 'bitzsk' logged off from 127.0.0.1
  == Manager 'bitzsk' logged on from 127.0.0.1
  == Manager 'bitzsk' logged off from 127.0.0.1
  == Manager 'bitzsk' logged on from 127.0.0.1
  == Manager 'bitzsk' logged off from 127.0.0.1
  == Manager 'bitzsk' logged on from 127.0.0.1
  == Manager 'bitzsk' logged off from 127.0.0.1
  == Manager 'bitzsk' logged on from 127.0.0.1
  == Manager 'bitzsk' logged off from 127.0.0.1
  == Manager 'bitzsk' logged on from 127.0.0.1
  == Manager 'bitzsk' logged off from 127.0.0.1
  == Manager 'bitzsk' logged on from 127.0.0.1


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Re: [asterisk-users] is my PHPAGI Soap code right?

[Zhang Shukun]

2010/5/14 --[ UxBoD ]-- :
>
> - Original Message -
>> Hello,
>>
>> i try to use soap in the phpagi.
>> i want to call a function from a web service
>> when a user call a telephne failed.
>>
>> this is my phpagi script, Could you show me what's wrong ? becasue i
>> can't excute it successfully.
>>
>> please open the following url to see my code:
>>
>> http://pastebin.com/uzvWSxPy
>>
>> Thanks!
>>
>
> Perhaps if you explained what errors you were seeing would help ? Have you 
> tried running it from the CLI to see if the syntax is correct ?

Thanks!  the systax is right in my php code. but when excute the php
script. there is errer happend in the server side.

as follows, i don't know what's wrong with it, please help me. thank you:

2010-05-17 14:08:19,359 INFO
[org.codehaus.xfire.handler.DefaultFaultHandler] - 
org.codehaus.xfire.fault.XFireFault: Not enough message parts were
received for the operation.
at 
org.codehaus.xfire.service.binding.ServiceInvocationHandler.fillInHolders(ServiceInvocationHandler.java:238)
at 
org.codehaus.xfire.service.binding.ServiceInvocationHandler.invoke(ServiceInvocationHandler.java:73)
at 
org.codehaus.xfire.handler.HandlerPipeline.invoke(HandlerPipeline.java:131)
at 
org.codehaus.xfire.transport.DefaultEndpoint.onReceive(DefaultEndpoint.java:64)
at 
org.codehaus.xfire.transport.AbstractChannel.receive(AbstractChannel.java:38)
at 
org.codehaus.xfire.transport.http.XFireServletController.invoke(XFireServletController.java:304)
at 
org.codehaus.xfire.transport.http.XFireServletController.doService(XFireServletController.java:129)
at 
org.codehaus.xfire.transport.http.XFireServlet.doPost(XFireServlet.java:116)
at javax.servlet.http.HttpServlet.service(HttpServlet.java:637)
at javax.servlet.http.HttpServlet.service(HttpServlet.java:717)
at 
org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290)
at 
org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206)
at 
org.extremecomponents.table.filter.AbstractExportFilter.doFilter(AbstractExportFilter.java:49)
at 
org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235)
at 
org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206)
at 
org.springframework.web.filter.CharacterEncodingFilter.doFilterInternal(CharacterEncodingFilter.java:96)
at 
org.springframework.web.filter.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:75)
at 
org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235)
at 
org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206)
at org.apache.

>
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>
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[asterisk-users] identify caller hangup or callee hangup?

[Zhang Shukun]

Hello,

you know , when a call setup, either caller hangup first or callee
hangup first , the hangupcause will set to 16(means Call Clearing
Causes)

My question is how could i identify whether the caller or callee
hangup the phone first?

Best Regards!
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[asterisk-users] is my PHPAGI Soap code right?

[Zhang Shukun]

Hello,

i try to use soap in the phpagi.
i want  to call a function from a web service
when a user call a telephne failed.

this is my phpagi script, Could you show me what's wrong ? becasue i
can't excute it successfully.

please open the following url to see my code:

http://pastebin.com/uzvWSxPy

Thanks!

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[asterisk-users] Could Asterisk PHP agi be a SOAP Client?

[Zhang Shukun]

hi, all

i want to use PHP agi to do as a soap client. does php agi support
this function?

Thanks!

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Re: [asterisk-users] Creating a HTTP Request on missed call?

[Zhang Shukun]

Thank you!  Motiejus Jakštys  and Sebastian Denz

it's helpful!

2010/5/11 Sebastian Denz :
> 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 command
>> and
>> man 1 curl
>>
>> Good luck
>
> Depending on his asterisk version, the dialplan function CURL() could be an
> option too..
>
> --
> Sebastian Denz  (System Engineer)
> * GONICUS GmbH * Zentrale * Moehnestrasse 11-17 * D-59755 Arnsberg
> * Tel.: +49 (0) 29 32 / 9 16 - 0 * Fax: +49 (0) 29 32 / 9 16  - 270
> * http://www.GONICUS.de
>
> *Sitz der Gesellschaft: Moehnestrasse 11-17 * D-59755 Arnsberg
> *Geschaeftsfuehrer: Rainer Luelsdorf, Alfred Schroeder
> *Vorsitzender des Beirats: Juergen Michels
> *Amtsgericht Arnsberg * HRB 1968
>
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Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension

[Zhang Shukun]

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 limit to 10.000 seconds.
  == Using SIP RTP CoS mark 5
-- Called 1000
-- SIP/1000-0032 is ringing
-- SIP/1000-0032 answered SIP/1001-0031
-- Executing [...@95040:1] Playback("SIP/1001-0031",
"vm-goodbye") in new stack
[May 11 17:23:16] WARNING[4258]: file.c:750 ast_readaudio_callback:
Failed to write frame
--  Playing 'vm-goodbye.gsm' (language 'en')
[May 11 17:23:16] WARNING[4258]: app_playback.c:471 playback_exec:
ast_streamfile failed on SIP/1001-0031 for vm-goodbye
-- Executing [...@95040:2] NoOp("SIP/1001-0031", "HANGUPCAUSE is
16") in new stack
  == Spawn extension (95040, 123, 1) exited non-zero on 'SIP/1001-0031'



this is the log which hangup by callee:

   == Using SIP RTP CoS mark 5
-- Executing [...@95040:1] Dial("SIP/1001-0033",
"SIP/1000,10,L(1)") in new stack
-- Setting call duration limit to 10.000 seconds.
  == Using SIP RTP CoS mark 5
-- Called 1000
-- SIP/1000-0034 is ringing
-- SIP/1000-0034 answered SIP/1001-0033
-- Executing [...@95040:1] Playback("SIP/1001-0033",
"vm-goodbye") in new stack
--  Playing 'vm-goodbye.gsm' (language 'en')
-- Executing [...@95040:2] NoOp("SIP/1001-0033", "HANGUPCAUSE is
16") in new stack
  == Spawn extension (95040, 123, 1) exited non-zero on 'SIP/1001-0033'


2010/5/11 Vardan :
> 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 Vardan:
>>> 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 2
>>>                  + Example 3: Macro for handling hangupcause
>>>                  + Example 4: Set the hangup cause text to a variable
>>>            o See also
>>>
>>>
>>> Asterisk variable Hangupcause
>>> Hangupcause is the latest PRI hangup return code on a zap channel
>>> connected to a PRI interface. Note that this also works on SIP channels,
>>> maybe other channels as well.
>>> Tip: The packet isdnutils contains a utility called isdncause that
>>> provides a textual explanation of the error number that you feed it with
>>> (watch the entry format).
>>>
>>> Previous to CVS 2004-08-12:
>>>
>>>   From causes.h:
>>>   #define AST_CAUSE_NOTDEFINED    0
>>>   #define AST_CAUSE_NORMAL        1
>>>   #define AST_CAUSE_BUSY          2
>>>   #define AST_CAUSE_FAILURE       3
>>>   #define AST_CAUSE_CONGESTION    4
>>>   #define AST_CAUSE_UNALLOCATED   5
>>>
>>>
>>> For CVS head releases after 2004-08-12:
>>>
>>>   /* Causes for disconnection (from Q.931) */
>>>   #define AST_CAUSE_UNALLOCATED 1
>>>   #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2
>>>   #define AST_CAUSE_NO_ROUTE_DESTINATION 3
>>>   #define AST_CAUSE_CHANNEL_UNACCEPTABLE 6
>>>   #define AST_CAUSE_CALL_AWARDED_DELIVERED 7
>>>   #define AST_CAUSE_NORMAL_CLEARING 16
>>>   #define AST_CAUSE_USER_BUSY 17
>>>   #define AST_CAUSE_NO_USER_RESPONSE 18
>>>   #define AST_CAUSE_NO_ANSWER 19
>>>   #define AST_CAUSE_CALL_REJECTED 21
>>>   #define AST_CAUSE_NUMBER_CHANGED 22
>>>   #define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27
>>>   #define AST_CAUSE_INVALID_NUMBER_FORMAT 28
>>>   #define AST_CAUSE_FACILITY_REJECTED 29
>>>   #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY 30
>>>   #define AST_CAUSE_NORMAL_UNSPECIFIED 31
>>>   #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34
>>>   #define AST_CAUSE_NETWORK_OUT_OF_ORDER 38
>>>   #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41
>>>   #define AST_CAUSE_SWITCH_CONGESTION 42
>>>   #define AST_CAUSE_ACCESS_INFO_DI

[asterisk-users] Creating a HTTP Request on missed call?

[Zhang Shukun]

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
Someone calls PBX and rings extension 100 -> Call is not answered ->
HTTP request is initiated to the following URL
"http://www.mywebsite.com/index.php?=NumberHere";
or something similar.

There are a couple of threads I have read on similar topics but none
seem to fit the bill. One involved installing a DialPlan Injection
module, however I was unable to get the associated SQL database
installed so I fell at the first hurdle.

I hope this is not too taxing to solve and understand that this may
not be a native function of asterisk, but any help would be greatly
appreciated.
Thanks in advance!


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Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension

[Zhang Shukun]

thank you for reply.

but hangupcause cant different whether caller hangup or callee hangup?

above two situation both return 16.

2010/5/11 Vardan :
> 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 2
>                 + Example 3: Macro for handling hangupcause
>                 + Example 4: Set the hangup cause text to a variable
>           o See also
>
>
> Asterisk variable Hangupcause
> Hangupcause is the latest PRI hangup return code on a zap channel
> connected to a PRI interface. Note that this also works on SIP channels,
> maybe other channels as well.
> Tip: The packet isdnutils contains a utility called isdncause that
> provides a textual explanation of the error number that you feed it with
> (watch the entry format).
>
> Previous to CVS 2004-08-12:
>
>  From causes.h:
>  #define AST_CAUSE_NOTDEFINED    0
>  #define AST_CAUSE_NORMAL        1
>  #define AST_CAUSE_BUSY          2
>  #define AST_CAUSE_FAILURE       3
>  #define AST_CAUSE_CONGESTION    4
>  #define AST_CAUSE_UNALLOCATED   5
>
>
> For CVS head releases after 2004-08-12:
>
>  /* Causes for disconnection (from Q.931) */
>  #define AST_CAUSE_UNALLOCATED 1
>  #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2
>  #define AST_CAUSE_NO_ROUTE_DESTINATION 3
>  #define AST_CAUSE_CHANNEL_UNACCEPTABLE 6
>  #define AST_CAUSE_CALL_AWARDED_DELIVERED 7
>  #define AST_CAUSE_NORMAL_CLEARING 16
>  #define AST_CAUSE_USER_BUSY 17
>  #define AST_CAUSE_NO_USER_RESPONSE 18
>  #define AST_CAUSE_NO_ANSWER 19
>  #define AST_CAUSE_CALL_REJECTED 21
>  #define AST_CAUSE_NUMBER_CHANGED 22
>  #define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27
>  #define AST_CAUSE_INVALID_NUMBER_FORMAT 28
>  #define AST_CAUSE_FACILITY_REJECTED 29
>  #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY 30
>  #define AST_CAUSE_NORMAL_UNSPECIFIED 31
>  #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34
>  #define AST_CAUSE_NETWORK_OUT_OF_ORDER 38
>  #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41
>  #define AST_CAUSE_SWITCH_CONGESTION 42
>  #define AST_CAUSE_ACCESS_INFO_DISCARDED 43
>  #define AST_CAUSE_REQUESTED_CHAN_UNAVAIL 44
>  #define AST_CAUSE_PRE_EMPTED 45
>  #define AST_CAUSE_FACILITY_NOT_SUBSCRIBED   50
>  #define AST_CAUSE_OUTGOING_CALL_BARRED      52
>  #define AST_CAUSE_INCOMING_CALL_BARRED      54
>  #define AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57
>  #define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL     58
>  #define AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65
>  #define AST_CAUSE_CHAN_NOT_IMPLEMENTED      66
>  #define AST_CAUSE_FACILITY_NOT_IMPLEMENTED      69
>  #define AST_CAUSE_INVALID_CALL_REFERENCE 81
>  #define AST_CAUSE_INCOMPATIBLE_DESTINATION 88
>  #define AST_CAUSE_INVALID_MSG_UNSPECIFIED   95
>  #define AST_CAUSE_MANDATORY_IE_MISSING 96
>  #define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97
>  #define AST_CAUSE_WRONG_MESSAGE 98
>  #define AST_CAUSE_IE_NONEXIST 99
>  #define AST_CAUSE_INVALID_IE_CONTENTS 100
>  #define AST_CAUSE_WRONG_CALL_STATE 101
>  #define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE 102
>  #define AST_CAUSE_MANDATORY_IE_LENGTH_ERROR 103
>  #define AST_CAUSE_PROTOCOL_ERROR 111
>  #define AST_CAUSE_INTERWORKING 127
>  /* Special Asterisk aliases */
>  #define AST_CAUSE_BUSY  AST_CAUSE_USER_BUSY
>  #define AST_CAUSE_FAILURE  AST_CAUSE_NETWORK_OUT_OF_ORDER
>  #define AST_CAUSE_NORMAL  AST_CAUSE_NORMAL_CLEARING
>  #define AST_CAUSE_NOANSWER   AST_CAUSE_NO_ANSWER
>  #define AST_CAUSE_CONGESTION   AST_CAUSE_NORMAL_CIRCUIT_CONGESTION
>  #define AST_CAUSE_NOTDEFINED  0
>
>
>
> Note: This does not work in 0.7.1 (maybe other versions) See:
> http://bugs.digium.com/bug_view_page.php?bug_id=890
>
> Recommended SIP <-> ISDN Cause codes (from RFC3398):
>
>   ISUP Cause value                        SIP response
>                           
>   1  unallocated number                   404 Not Found
>   2  no route to network                  404 Not found
>   3  no route to destination              404 Not found
>   16 normal call clearing                 --- (*)
>   17 user busy                            486 Busy here
>   18 no user responding                   408 Request Timeout
>   19 no answer from the user              480 Temporarily unavailable
>   20 subscriber absent                    480 Temporarily unavailable
>   21 call rejected                        403 Forbidden (+)
>   22 number changed (w/o diagnostic)      410 Gone
>   22 number changed (w/ diagnostic)       301 Moved 

Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension

[Zhang Shukun]

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 :
>> However, as I can see by the verbose command, ${HANGUPCAUSE} is always
>> 0. I thought it was a channel variable that contained the hangupcause?
>
> Just an update, if the call is established, then there is a
> hangupcause received.
>
> The above problem only happens if the caller hangs up before pickup.
>
> This is usualy a cause 16, not 0.
>
> Alex
>
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Re: [asterisk-users] Time difference in CSV CDR's and MySQL CDR's

[Zhang Shukun]

2010/4/22 Steve Edwards :
> Un-top-posting...
>
>> 2010/4/22 Alejandro Recarey :
>>>
>>> I am having a curious problem. I use two cdr backends, csv and MySQL.
>>>
>>> I am finding that the calldate field varies between 3 seconds and 3
>>> minutes between the MySQL database and the CSV files! Is this expected
>>> behaviour? I thought they should both use the same timestamp.
>
> On Thu, 22 Apr 2010, Zhang Shukun wrote:
>
>> the time in the file cdr is right, as mysql. calldate is the time when
>> the record insert into mysql.
>
> I'm just a 1.2 Luddite, but...
>
> In cdr_addon_mysql.c:
>
>         localtime_r(&cdr->start.tv_sec,&tm);
>         strftime(timestr,128,DATE_FORMAT,&tm);
>
> and then timestr is used to populate the 'calldate' column when the insert
> statement is built.
>
> Which is consistent with my CDRs -- they show the time the call was
> started, not some time after the call is finished when the row is inserted
> into the database.
but in the cdr_mysql.conf, it said as following:


; Older versions of cdr_mysql set the calldate field to whenever the
; record was posted, rather than the start date of the call.  This flag
; reverts to the old (incorrect) behavior.  Note that you'll also need
; to comment out the "start=calldate" alias, below, to use this.
compat=no


i use asterisk 1.6.2.1


>
> --
> Thanks in advance,
> -
> Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                              Fax: +1-760-731-3000
>
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Re: [asterisk-users] Time difference in CSV CDR's and MySQL CDR's

[Zhang Shukun]

the time in the file cdr is right, as mysql. calldate is the time when
the record insert into mysql.

2010/4/22 Alejandro Recarey :
> Hi all,
>
> I am having a curious problem. I use two cdr backends, csv and MySQL.
> These are my settings:
>
> Call Detail Record (CDR) settings
> --
>  Logging:                    Enabled
>  Mode:                       Batch
>  Log unanswered calls:       Yes
>
> * Batch Mode Settings
>  ---
>  Safe shutdown:              Enabled
>  Threading model:            Scheduler plus separate threads
>  Current batch size:         0 records
>  Maximum batch size:         25 records
>  Maximum batch time:         10 seconds
>  Next batch processing time: 7 seconds
>
> * Registered Backends
>  ---
>    csv
>    mysql
>    cdr-custom
>
>
> I am finding that the calldate field varies between 3 seconds and 3
> minutes between the MySQL database and the CSV files! Is this expected
> behaviour? I thought they should both use the same timestamp. Is is
> very difficult to match CDR's this way, and I am finding it hard to
> trust the results, as I wanted to make sure that my database was
> behaving correctly and not "losing" any CDR's along the way.
>
> Which one of the two CDR's is correct?
>
> Should this be posted as a bug?
>
> Regards,
>
> Alex
>
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Re: [asterisk-users] G729 exhaustion conditions

[Zhang Shukun]

G729 is free for use is no transcoding is done.

2010/4/19 Harel Cohen :
> Hi all,
>
> Suppose I buy and install one G729 codec. Suppose there is one call going on
> where both end-points have G729 codecs and the Asterisk is not doing any
> transcoding. Does this conversation exhaust my G729 license (even though
> this call would have worked without license in the first place) or do I
> still have the ability to use this G729 codec for other call which requires
> transcoding?
>
> Thank you,
>
> Harel Cohen
>
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Re: [asterisk-users] Detect if a Number is up or not

[Zhang Shukun]

Dial() will return Dialstatus , if the number dialed is busy or off
now. use this application you can detect a number is busy or not

in several seconds. i use this method in my dialplan.

2010/4/19 ABBAS SHAKEEL :
> Hello Community,
>
> I Want to detect if a cell number is ON or OFF... for that matter i can
> generate call to it using PSTN lines (configured with asterisk).
>
> The problem is that  i only want to see if the cell number can receive a
> ring or not. If ring is recieved at called number end then mark it as ON in
> database...
>
> Dial application or Originate action might not be that helpful. Do you have
> any idea regarding this..
>
> --
> Best Regards
> Shakeel Abbas
>
>
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[asterisk-users] Set CDR amaflags not work

[Zhang Shukun]

Hi, when i use Set(CDR(amaflags)=1) in my dial plan , after a call. i
look at the cdr table in mysql database.

the value of amaflags is -1 not 1, do you know what's wrong?

Thanks!
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Re: [asterisk-users] How to get Sip response codes in Dialplan?

[Zhang Shukun]

2010/3/25 Juan E. Rodríguez :
> Try using DIALSTATUS.

Thank you!

but DIALSTATUS IS used for Dial. not for queue

>
> --Mensaje original--
> De: Zhang Shukun
> Remitente: asterisk-users-boun...@lists.digium.com
> Para: Asterisk Users Mailing List - Non-Commercial Discussion
> Responder a: Asterisk Users Mailing List - Non-Commercial Discussion
> Asunto: [asterisk-users] How to get Sip response codes in Dialplan?
> Enviado: 24 Mar, 2010 23:29
>
> hi ,all
>
> when a Dial or Queue excutes, a sip response code will return. like
>
> == Using SIP RTP CoS mark 5
>    -- Got SIP response 502 "Bad Gateway" back from 211.150.119.32
>    -- SIP/95040-004a is circuit-busy
>    -- Nobody picked up in 2000 ms
>
> My quesion is how to get the response code in the dial plan
> immediatelly in order to do different thing according the returned
> codes?
>
> for example: a queue response code is "busy now" i will queue another
> number immediately not let the user waiting for the timeout.
>
> Thanks!
>
> --
> Best regards,
> Sucan
>
> --
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>
> Saludos,
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[asterisk-users] How to get Sip response codes in Dialplan?

[Zhang Shukun]

hi ,all

when a Dial or Queue excutes, a sip response code will return. like

== Using SIP RTP CoS mark 5
-- Got SIP response 502 "Bad Gateway" back from 211.150.119.32
-- SIP/95040-004a is circuit-busy
-- Nobody picked up in 2000 ms

My quesion is how to get the response code in the dial plan
immediatelly in order to do different thing according the returned
codes?

for example: a queue response code is "busy now" i will queue another
number immediately not let the user waiting for the timeout.

Thanks!

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Re: [asterisk-users] Mobile phone shut down, but Queue() Ring as usual

[Zhang Shukun]

2010/3/24 Alyed :
> Try the same as in
>
> http://lists.digium.com/pipermail/asterisk-users/2010-March/246316.html
>
> just make sure to add this in the [channels] context ;)

Thanks for your answer!  but i am not use zaptel device card to
connect to PSTN but use Mediant 2000 - AudioCodes to connect
to PSTN. so i can't change Zapata.conf  file to add something like

busydetect=yes
busycount=3


In this situation how should i do?

>
> Hope it helps.
>
> Alyed
>
>
> 2010/3/23 Zhang Shukun 
>>
>> hi, all
>>
>> i use Queue() to call a Mobile phone, there is only one mobile phone
>> in the queue. even if the mobile phone shut down, Queue() is ring in
>> the cli verbose
>>
>> as mobile phone is normally working. what i want to see is if the
>> mobile phone is shut down.
>>
>> queue() will end immediately to tell on one in the queue.
>>
>> is there any method to do this ?
>>
>> --
>> Best regards,
>> Sucan
>>
>> --
>> _
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>
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[asterisk-users] Mobile phone shut down, but Queue() Ring as usual

[Zhang Shukun]

hi, all

i use Queue() to call a Mobile phone, there is only one mobile phone
in the queue. even if the mobile phone shut down, Queue() is ring in
the cli verbose

as mobile phone is normally working. what i want to see is if the
mobile phone is shut down.

queue() will end immediately to tell on one in the queue.

is there any method to do this ?

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Re: [asterisk-users] How to detect a PSTN telephone is busy or not?

[Zhang Shukun]

Thanks!  but if  i use Queue to call out not Dial.

how should i know the status like busy or free?

for now . i know asterisk have QUEUESTATUS variable，

QUEUESTATUS   The status of the call as a text string, one of TIMEOUT
| FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL

but the variable have no busy or free status?  how to  know the
numbers in the queue is busy or not at present?

Need your help. thanks!

2010/3/18 ABBAS SHAKEEL :
> Hello,
> Please have a look to DIALSTATUS variable.
> here http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
> I hope it helps
>
>
> On Thu, Mar 18, 2010 at 1:31 PM, Zhang Shukun  wrote:
>>
>> hi,all
>>
>> one problem confuse me these days. i want to sequence dial three PSTN
>> number(a,b,c)
>>
>> first, if i dial number a, if a is busy , i will dial number b. if b
>> is busy, i will dial number c.
>>
>> Dial(SIP/a...@ip,30)
>> Dial(SIP/b...@ip,30)
>> Dial(SIP/c...@ip,30)
>>
>> i want to know before i dial number a, how to know if a is busy now?
>>
>> if a is busy now. i will not dial a, instead, i will dial number b
>> directly.
>>
>> to summary is : in asterisk, how to detect a pstn telephone number is
>> busy or not before dialing it?
>>
>> Thanks!
>>
>> --
>> Best regards,
>> Sucan
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>
>
> --
> Best Regards
> Shakeel Abbas
>
>
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[asterisk-users] How to detect a PSTN telephone is busy or not?

[Zhang Shukun]

hi,all

one problem confuse me these days. i want to sequence dial three PSTN
number(a,b,c)

first, if i dial number a, if a is busy , i will dial number b. if b
is busy, i will dial number c.

Dial(SIP/a...@ip,30)
Dial(SIP/b...@ip,30)
Dial(SIP/c...@ip,30)

i want to know before i dial number a, how to know if a is busy now?

if a is busy now. i will not dial a, instead, i will dial number b directly.

to summary is : in asterisk, how to detect a pstn telephone number is
busy or not before dialing it?

Thanks!

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[asterisk-users] Extensions.conf changed but not take effect

[Zhang Shukun]

hi, All

one thing confused me a long time.
when i change the extensions.conf file. why not take effects after
restart the asterisk? details as follow:

my dailplan is :

[95040]

exten => _95040X,1,Set(CALLINNUM=${CALLERID(dnid)})
exten => _95040X,n(start),Answer
exten => _95040X,n(welcome),Background(${welcomefile},,123)
...

exten => i,1,Playback(invalid)
exten => i,2,Goto(${CALLINNUM},welcome)


first, i have changed  _95040XX  to _95040X ,  but when excute

exten => i,2,Goto(${CALLINNUM},welcome)

cli error message as follow:

 -- Executing [...@extinvalid:2] Goto("SIP/1000-0005",
"95040,_95040XX,welcome") in new stack
[Mar 10 17:17:27] NOTICE[5057]: pbx.c:3731 pbx_extension_helper:
Cannot find extension '_95040XX' in context '95040'
[Mar 10 17:17:27] WARNING[5057]: pbx.c:9602 pbx_parseable_goto:
Priority 'welcome' must be a number > 0, or valid label

do you know what's wrong? does asterisk have some buffer or cache
files? although i change but read the old file?

thank you very much!

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[asterisk-users] Does Asterisk 1.6.2.1 Support SIP TLS encryption

[Zhang Shukun]

hi, all

i want to realize more secure communication between asterisk sip end users.

so i want to know Does Asterisk 1.6.2.1 Support SIP TLS encryption?

if you can tell me same specific example to do encrypt, it's very appreciated.

Thanks!

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[asterisk-users] realtime modules not load ?

[Zhang Shukun]

hi, all

i want to try realtime function. but after i install the adds-on . i
cant see the realtime modules have been loaded.

modules exist here:

[r...@localhost modules]# ls *mysql*
app_addon_sql_mysql.so  cdr_addon_mysql.so  res_config_mysql.so

and i can't find the modules

*CLI> module show like sql
Module Description
 Use Count
0 modules loaded

what should i do in order to load the app_addon_sql_mysql.so  and
cdr_addon_mysql.so module.

i have add this lines to modules.conf :

load => cdr_addon_mysql.so

load => app_addon_sql_mysql.so

but seems not loaded either.

need your help.

thanks!



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Re: [asterisk-users] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available

[Zhang Shukun]

2010/2/26 Tilghman Lesher :
> On Friday 26 February 2010 00:09:55 Warren Selby wrote:
>> On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun  wrote:
>> > [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime
>> > mapping for 'sippeers' found to engine 'mysql', but the engine is not
>> > available--
>>
>> Is MySQL running and all the proper values set in the appropriate files?
>
> Does the config name in extconfig.conf right after the word "mysql" exist as a
> section in res_mysql.conf?

the section in extconfig.conf is :


sipusers => mysql,asterisk,sip_buddies
sippeers => mysql,asterisk,sip_buddies


and section in res_mysql.conf is :



[asterisk]
;dbhost = 127.0.0.1
dbname = asterisk
dbuser = root
dbpass = net263
dbport = 3306
;dbsock = /tmp/mysql.sock
;dbsock = /var/run/mysqld/mysqld.sock
requirements=createclose ; or createclose or createchar


>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available

[Zhang Shukun]

yes. mysql run ok

the configuration is ok too. i think

is this error shows asterisk can't find mysql database?

2010/2/26 Warren Selby :
> On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun  wrote:
>>
>> hi, all
>>
>> after my installation of asterisk and adds-on .
>>
>> when start astrisk, error accours as follow:
>>
>>
>> [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime
>> mapping for 'sippeers' found to engine 'mysql', but the engine is not
>> available--
>
> Is MySQL running and all the proper values set in the appropriate files?
>
> Thanks,
> --Warren Selby
> http://www.selbytech.com
>
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[asterisk-users] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available

[Zhang Shukun]

hi, all

after my installation of asterisk and adds-on .

when start astrisk, error accours as follow:


[Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime
mapping for 'sippeers' found to engine 'mysql', but the engine is not
available

what's wrong with me ?

Thanks.


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Re: [asterisk-users] Do i need install Dahdi or libpri ?

[Zhang Shukun]

Thank you! it's very helpful

2010/2/25 Steve Howes :
>
> On 25 Feb 2010, at 02:16, Zhang Shukun wrote:
>> there is a AudioCodes Mediant 2000 out there. i want to realise ip to
>> PSTN and PSTN to ip connection.
>
> Ok.
>
>> after some configuration on AudioCodes Mediant 2000, PSTN to ip
>> connecttion works.
>
> Thats good.
>
>> a, Do i need install DAHDI or libpri in my system?
> Depends what else you are doing. I'd always just install it anyway.
>
>> b, how to write in dialplan to realise connection to PSTN.
> The Mediant 2000 can be used like any other sip device. Dial(SIP/
> whatever/1234567890)
>
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[asterisk-users] Do i need install Dahdi or libpri ?

[Zhang Shukun]

hello,all

there is a AudioCodes Mediant 2000 out there. i want to realise ip to
PSTN and PSTN to ip connection.

after some configuration on AudioCodes Mediant 2000, PSTN to ip
connecttion works.

next ,i want to dial from asterisk to PSTN now. i have see the sample
in the extensions.conf relevent to PSTN as follow:

; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325

but above shows something about DAHDI card.

my question is:

a, Do i need install DAHDI or libpri in my system?
b, how to write in dialplan to realise connection to PSTN.

That'sPSTN->AudioCodes Mediant
2000--->IP(asterisk)->AudioCodes Mediant 2000
-->PSTN

?

Thanks very much!
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[asterisk-users] Does Playback will answer the call?

[Zhang Shukun]

hi, all

in my test,it shows Playback will answer the call automaticly, but i
don't want to so.

i will use answer function to answer the call. could you help me ?

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[asterisk-users] Realtime queue strategy issue

[Zhang Shukun]

hi,all

all realtime queue work fine except one thing:

in the queue_table ,when i change strategy from ringall to linear need
asterisk to restart!

[Feb  2 15:41:51] WARNING[4106]: app_queue.c:1532 queue_set_param:
Changing to the linear strategy currently requires asterisk to be
restarted.

i use asterisk 1.6.2.1, is this a bug of Asterisk 1.6.2.1?

does anyone have some idea for me ?

Thanks!
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Re: [asterisk-users] Realtime Queue not work in 1.6.2.1

[Zhang Shukun]

2010/1/28 Håkon Nessjøen :
> All your agents have paused=1. They will not receive calls while they are
> paused.

Solved Thanks very much!

>
> Håkon
>
> On Thu, Jan 28, 2010 at 3:23 AM, Zhang Shukun  wrote:
>>
>> 2010/1/28 Carlos Chavez :
>> > On Wed, 2010-01-27 at 10:27 +0800, Zhang Shukun wrote:
>> >> hi,all
>> >>
>> >> i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except
>> >> realtime queue.
>> >>
>> >> it seems queue_table works fine, but queue_member_queue not work, the
>> >> two tables works fine when in 1.4.28.
>> >>
>> >> is that something changed related to realtime queue configuration?
>> >>
>> >> more detail about two table definition and data stored in , please see:
>> >>
>> >> http://pastebin.com/m33f9539e
>> >>
>> >> the extconfig.conf file, please see:
>> >>
>> >> http://pastebin.com/m2008ced1
>> >>
>> >> and the res_mysql.conf file:
>> >>
>> >> http://pastebin.com/m27d3fdc5
>> >>
>> >> Could you tell me what's wrong with me ?
>> >>
>> >> Thanks!
>> >
>> >        How do your agents log into the system?
>>
>> Thanks! i don't want to use agents member to login to system. i just
>> want to set static SIP peers in the queue
>>
>> and they all can work according to the strategy when have call to the
>> queue.just like follows:
>>
>> mysql> select * from queue_table;
>> +--+---+-+
>> | name         | beginworktime | endworktime |
>> +--+---+-+
>> | 950401234561 | 09:30:00      | 17:30:00    |
>> +--+---+-+
>> 3 rows in set (0.00 sec)
>>
>> mysql> select * from queue_member_table;
>>
>> +--++--+---+-++
>> | uniqueid | membername     | queue_name   | interface | penalty | paused
>> |
>>
>> +--++--+---+-++
>> |       18 | Zhang Shukun   | 950401234561 | SIP/1001  |       0 |      1
>> |
>> |       19 | Li Aiwei       | 950401234561 | SIP/1002  |       0 |      1
>> |
>> |       20 | Zhang Jianming | 950401234561 | SIP/1003  |       0 |      1
>> |
>>
>> +--++--+---+-++
>> 3 rows in set (0.00 sec)
>>
>> in above two table. queue:950401234561  have three queue members:
>> SIP/1001 ,  SIP/1002 , SIP/1003
>>
>> when Queue(950401234561) app is invoked, all three queue members will
>> ring at the same time by default strategy(ringall).
>>
>> my problem now use asterisk 1.6.2.1 is :
>>
>> when Queue(950401234561) app is running, i can here music on hold, but
>> none of my sip phones(SIP/1001 ,  SIP/1002 , SIP/1003) will ring, is
>> that in asterisk 1.6.2.1, it's not support static realtime queue
>> member any more?
>>
>> > If you were using
>> > agentcallbacklogin that was deprecated and does not exist in version 1.6
>> > of Asterisk.  The queue_member_table was used by agentcallbacklogin or
>> > the agentlogin commands.  With Asterisk 1.6 you are supposed to be using
>> > dynamic agents so there is no purpose for that table.
>> >
>> >        That is what may be wrong with Asterisk.  What is wrong with you
>> > is a
>> > very different question ;)
>> >
>> > --
>> > Telecomunicaciones Abiertas de México S.A. de C.V.
>> > Carlos Chávez Prats
>> > Director de Tecnología
>> > +52-55-91169161 ext 2001
>> >
>> > --
>> > _
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>> >
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>> > To UNSUBSCRIBE or update options visit:
>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
>>
>>
>> --
>> Best regards,
>> Sucan
>>
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Re: [asterisk-users] Realtime Queue not work in 1.6.2.1

[Zhang Shukun]

2010/1/28 Carlos Chavez :
> On Wed, 2010-01-27 at 10:27 +0800, Zhang Shukun wrote:
>> hi,all
>>
>> i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except
>> realtime queue.
>>
>> it seems queue_table works fine, but queue_member_queue not work, the
>> two tables works fine when in 1.4.28.
>>
>> is that something changed related to realtime queue configuration?
>>
>> more detail about two table definition and data stored in , please see:
>>
>> http://pastebin.com/m33f9539e
>>
>> the extconfig.conf file, please see:
>>
>> http://pastebin.com/m2008ced1
>>
>> and the res_mysql.conf file:
>>
>> http://pastebin.com/m27d3fdc5
>>
>> Could you tell me what's wrong with me ?
>>
>> Thanks!
>
>        How do your agents log into the system?

Thanks! i don't want to use agents member to login to system. i just
want to set static SIP peers in the queue

and they all can work according to the strategy when have call to the
queue.just like follows:

mysql> select * from queue_table;
+--+---+-+
| name | beginworktime | endworktime |
+--+---+-+
| 950401234561 | 09:30:00  | 17:30:00|
+--+---+-+
3 rows in set (0.00 sec)

mysql> select * from queue_member_table;
+--++--+---+-+----+
| uniqueid | membername | queue_name   | interface | penalty | paused |
+--++--+---+-++
|   18 | Zhang Shukun   | 950401234561 | SIP/1001  |   0 |  1 |
|   19 | Li Aiwei   | 950401234561 | SIP/1002  |   0 |  1 |
|   20 | Zhang Jianming | 950401234561 | SIP/1003  |   0 |  1 |
+--++--+---+-++
3 rows in set (0.00 sec)

in above two table. queue:950401234561  have three queue members:
SIP/1001 ,  SIP/1002 , SIP/1003

when Queue(950401234561) app is invoked, all three queue members will
ring at the same time by default strategy(ringall).

my problem now use asterisk 1.6.2.1 is :

when Queue(950401234561) app is running, i can here music on hold, but
none of my sip phones(SIP/1001 ,  SIP/1002 , SIP/1003) will ring, is
that in asterisk 1.6.2.1, it's not support static realtime queue
member any more?

> If you were using
> agentcallbacklogin that was deprecated and does not exist in version 1.6
> of Asterisk.  The queue_member_table was used by agentcallbacklogin or
> the agentlogin commands.  With Asterisk 1.6 you are supposed to be using
> dynamic agents so there is no purpose for that table.
>
>        That is what may be wrong with Asterisk.  What is wrong with you is a
> very different question ;)
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
> _
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>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Re: [asterisk-users] MYSQL problem

[Zhang Shukun]

2010/1/27 Steve Edwards :
> Un-mid-posting...
>
>>> On Fri, 22 Jan 2010, Zhang Shukun wrote:
>>>
>>>> as you know, we can use MYSQL command to visit mysql database but if i
>>>> use other database like Oracke,sybase,etc, Could i use MYSQL command ?
>>>
>> 2010/1/23 Steve Edwards :
>
>>> ODBC will do what you want.
>>>
>>> Personally, I'd vote for an AGI using whatever C API your DB provides
>>> -- like Pro*C to access Oracle.
>>>
>>> You will have access to all of the features of your DB and your
>>> dialplan will be a lot cleaner and easier to maintain.
>
> On Wed, 27 Jan 2010, Zhang Shukun wrote:
>
>> Thanks, while i think because oracle has no offical ODBC for linux
>> system. is that better use mysql than oracle. considering the
>> perfoermace and speed. many people around think mysql is not a good
>> option for database, they think mysql is only suit for small business.
>> but i want to have a try. i need to convince them to use this.
>
> So don't use ODBC, use Pro*C...

you said" Personally, I'd vote for an AGI using whatever C API your DB provides"

what do you think about phpagi and cagi, if i choose the agi method.

while phpagi seems used more popular than cagi.

>
> Back when Yahoo was relevant, they ran on MySQL.
>
> Can you quantify your requirements (number of rows, queries per second,
> simultaneous connections) and test it on hardware similar to your
> production environment?

i cant quantify my requirement now. but the business has not be start.

and the user will increase as time going.

>
> While I'm sure Yahoo spent a lot of time and money designing and tuning
> their system, sometimes "explain plan" can point you to small changes that
> yield significant results.
>
> If your shop is committed to Oracle, can finance the licenses, and has the
> in-house talent -- use it. Nobody ever lost their job by buying IBM...

>
> --
> Thanks in advance,
> -
> Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                              Fax: +1-760-731-3000
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Re: [asterisk-users] GoToIfTime issue

[Zhang Shukun]

2010/1/22 Tilghman Lesher :
> On Friday 22 January 2010 04:06:29 Zhang Shukun wrote:
>> 2010/1/22 Randy R :
>> > On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun  wrote:
>> >> exten => 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1)
>> >>
>> >> but what should i do. if i want to set seperate weekdays,like mon,wed.
>> >> not continuous weekday like mon-fri.
>> >
>> > I couldn't find any reference to multiple, non-contiguous days on a
>> > quick Google, but this would work at the cost of an extra line:
>> >
>> > exten => 222,1,GoToIfTime(11:00-14:00|mon|*|*?1:3,1)
>> > exten => 222,2,GoToIfTime(11:00-14:00|wed|*|*?1:3,1)
>>
>> Thank you, but why don't it to be comma seperate to represent seperate
>> weekdays?  as | mon,wed,fri |
>>
>> it's also very understandable.
>
> Starting in 1.6.2, you can use the ampersand to join days.

when i update from 1.4.28 to 1.6.2.1, the join days problem solved.
but not the realtime queue not work.

should i change the queue_table and queue_member_table definition
along with the upgrading?

because i don't know how the two table match each other.

mysql> select * from queue_table;
+--+---+-+
| name | beginworktime | endworktime |
+--+---+-+
| 950401234561 | 09:30:00  | 17:30:00|
| 950401234562 | 11:30:00  | 17:30:00|
| 950401234563 | 16:30:00  | 17:30:00|
+--+---+-+
3 rows in set (0.00 sec)

mysql> select * from queue_member_table;
+--++--+---+-----++
| uniqueid | membername | queue_name   | interface | penalty | paused |
+--++--+---+-++
|   18 | Zhang Shukun   | 950401234561 | SIP/1001  |   0 |  1 |
|   19 | Li Aiwei   | 950401234561 | SIP/1002  |   0 |  1 |
|   20 | Zhang Jianming | 950401234561 | SIP/1003  |   0 |  1 |
+--++--+---+-++
3 rows in set (0.00 sec)


how colume name queue_name in queue_member_table match name in queue_table?

how does the system recognize them.   i mean queue_name is not an
configure option in agent.conf

>
> --
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> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] MySQL RealTime Error

[Zhang Shukun]

2010/1/26 Carlos Chavez :
>        You must read the upgrade instructions.  The database definitions in
> res_mysql.conf have changed.  The way you reference the database in
> extconfig.conf is also different.

solved...

it is my configuration error of res_mysql.conf and extconfig.conf file

the database name not matched.

>
> On Mon, 2010-01-25 at 09:33 +, Ishfaq Malik wrote:
>> What happens when you try the command
>>
>> mysql -uroot -proot asterisk
>>
>> Ish
>>
>> Zhang Shukun wrote:
>> > hi,all
>> >
>> > when i upgrade from 1.4.28 to 1.6.2.1, i can't connect to mysql
>> > database anymore, error as follow:
>> >
>> > [Jan 25 15:38:25] WARNING[3003]: res_config_mysql.c:325
>> > realtime_mysql: MySQL RealTime: Invalid database specified: asterisk
>> > (check res_mysql.conf)
>> >
>> > the content of res_mysql.conf is:
>> >
>> > http://www.pastebin.org/81966
>> >
>> > i've try command " mysql -uroot -proot" ,i can connect to mysql 
>> > successfully.
>> >
>> > Could you tell me what's wrong with me ?
>> >
>> >
>>
>> --
>> Ishfaq Malik
>> Software Developer
>> PackNet Ltd
>>
>> Office:   0161 660 3062
>>
>
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> Carlos Chávez Prats
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Re: [asterisk-users] MYSQL problem

[Zhang Shukun]

2010/1/23 Steve Edwards :
> On Fri, 22 Jan 2010, Zhang Shukun wrote:
>
>> as you know, we can use MYSQL command to visit mysql database
>>
>> but if i use other database like Oracke,sybase,etc, Could i use MYSQL
>> command ?
>
> ODBC will do what you want.

Thanks, while i think because oracle has no offical ODBC for linux system.

is that better use mysql than oracle. considering the perfoermace and speed.

many people around think mysql is not a good option for database, they
think mysql

is only suit for small business. but i want to have a try. i need to
convince them to use this.

>
> Personally, I'd vote for an AGI using whatever C API your DB provides
> -- like Pro*C to access Oracle.
>
> You will have access to all of the features of your DB and your dialplan
> will be a lot cleaner and easier to maintain.
>
> --
> Thanks in advance,
> -
> Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                              Fax: +1-760-731-3000
>
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Re: [asterisk-users] MYSQL grammar diff in 1.6.2.1?

[Zhang Shukun]

2010/1/26 Tilghman Lesher :
> On Monday 25 January 2010 03:12:08 Zhang Shukun wrote:
>> hi, dear all
>>
>> MYSQL commands work well in 1.4.28 edition, but not in 1.6.21
>>
>> is that the grammar is different between them?
>>
>> extensions.conf
>>
>> exten => s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
>> blacklist\ where\ companycode = ${ARG2} and callerid=${ARG1} and
>> blockenabled = 1)
>>
>> cli:
>>     -- Executing [...@macro-checkblacklist:2] MYSQL("SIP/1003-0006",
>> "Query resultid 1 SELECT\ callerid\ from\ blacklist\ where\
>> companycode = 95040654321 and callerid=1003 and blockenabled = 1") in
>> new stack
>> [Jan 25 17:05:34] WARNING[2583]: app_addon_sql_mysql.c:374
>> aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an
>> error in your SQL syntax; check the manual that corresponds to your
>> MySQL server version for the right syntax to use near '\ callerid\
>> from\ blacklist\ where\ companycode = 95040654321 and callerid=1003 '
>> at line 1
>
> I have no idea why you backslashed your spaces in 1.4, as that has never been
> a requirement (as is evident later in the line, where you neglected them).

wow, i learn this method from voip-info and do the same as the example showed.

so i add backslash.

> This is the problem, and if you remove the backslashes, you should be fine.

you are right! it really works now by remove backslash.

maybe in asterisk 1.4 add backslash or not is both working.

but in asterisk 1.6 , must not contain backslash. or it will not work any more.

>
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>
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[asterisk-users] Realtime Queue not work in 1.6.2.1

[Zhang Shukun]

hi,all

i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except
realtime queue.

it seems queue_table works fine, but queue_member_queue not work, the
two tables works fine when in 1.4.28.

is that something changed related to realtime queue configuration?

more detail about two table definition and data stored in , please see:

http://pastebin.com/m33f9539e

the extconfig.conf file, please see:

http://pastebin.com/m2008ced1

and the res_mysql.conf file:

http://pastebin.com/m27d3fdc5

Could you tell me what's wrong with me ?

Thanks!



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[asterisk-users] MYSQL grammar diff in 1.6.2.1?

[Zhang Shukun]

hi, dear all

MYSQL commands work well in 1.4.28 edition, but not in 1.6.21

is that the grammar is different between them?

extensions.conf

exten => s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
blacklist\ where\ companycode = ${ARG2} and callerid=${ARG1} and
blockenabled = 1)

cli:
-- Executing [...@macro-checkblacklist:2] MYSQL("SIP/1003-0006",
"Query resultid 1 SELECT\ callerid\ from\ blacklist\ where\
companycode = 95040654321 and callerid=1003 and blockenabled = 1") in
new stack
[Jan 25 17:05:34] WARNING[2583]: app_addon_sql_mysql.c:374
aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an
error in your SQL syntax; check the manual that corresponds to your
MySQL server version for the right syntax to use near '\ callerid\
from\ blacklist\ where\ companycode = 95040654321 and callerid=1003 '
at line 1


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[asterisk-users] MySQL RealTime Error

[Zhang Shukun]

hi,all

when i upgrade from 1.4.28 to 1.6.2.1, i can't connect to mysql
database anymore, error as follow:

[Jan 25 15:38:25] WARNING[3003]: res_config_mysql.c:325
realtime_mysql: MySQL RealTime: Invalid database specified: asterisk
(check res_mysql.conf)

the content of res_mysql.conf is:

http://www.pastebin.org/81966

i've try command " mysql -uroot -proot" ,i can connect to mysql successfully.

Could you tell me what's wrong with me ?

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Re: [asterisk-users] GoToIfTime issue

[Zhang Shukun]

2010/1/22 Tilghman Lesher :
> On Friday 22 January 2010 04:06:29 Zhang Shukun wrote:
>> 2010/1/22 Randy R :
>> > On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun  wrote:
>> >> exten => 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1)
>> >>
>> >> but what should i do. if i want to set seperate weekdays,like mon,wed.
>> >> not continuous weekday like mon-fri.
>> >
>> > I couldn't find any reference to multiple, non-contiguous days on a
>> > quick Google, but this would work at the cost of an extra line:
>> >
>> > exten => 222,1,GoToIfTime(11:00-14:00|mon|*|*?1:3,1)
>> > exten => 222,2,GoToIfTime(11:00-14:00|wed|*|*?1:3,1)
>>
>> Thank you, but why don't it to be comma seperate to represent seperate
>> weekdays?  as | mon,wed,fri |
>>
>> it's also very understandable.
>
> Starting in 1.6.2, you can use the ampersand to join days.

thanks! but i wonder while in asterisk 1.6.2, the web sites says many
funtions of them is experimental,

is this edition good for production environment use？


>
> --
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> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] GoToIfTime issue

[Zhang Shukun]

2010/1/22 Randy R :
> On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun  wrote:
>> exten => 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1)
>
>> but what should i do. if i want to set seperate weekdays,like mon,wed.
>> not continuous weekday like mon-fri.
>
> I couldn't find any reference to multiple, non-contiguous days on a
> quick Google, but this would work at the cost of an extra line:
>
> exten => 222,1,GoToIfTime(11:00-14:00|mon|*|*?1:3,1)
> exten => 222,2,GoToIfTime(11:00-14:00|wed|*|*?1:3,1)

Thank you, but why don't it to be comma seperate to represent seperate
weekdays?  as | mon,wed,fri |

it's also very understandable.

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[asterisk-users] GoToIfTime issue

[Zhang Shukun]

hi , all

what's wrong with this command?

exten => 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1)

as i got the error:
-- Executing [...@95040:1] GotoIfTime("SIP/1001-0099",
"11:00-14:00|mon|wed|*|*?1:3|1") in new stack
[Jan 20 11:21:11] WARNING[16804]: pbx.c:4118 get_range: Invalid day
'wed', assuming none


but what should i do. if i want to set seperate weekdays,like mon,wed.
not continuous weekday like mon-fri.
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[asterisk-users] MYSQL problem

[Zhang Shukun]

hi,all

as you know, we can use MYSQL command to visit mysql database

but if i use other database like Oracke,sybase,etc, Could i use MYSQL command ?

if not, is there any other alternative could do the same
function(visit database in dailplan)?

Thanks!
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Re: [asterisk-users] wav to gsm can't play

[Zhang Shukun]

yes. asterisk can playback wav file . but need transfer to 8000hz.

Using WAV files
Asterisk has codecs for wav (pcm), gsm, g729, g726, and wav49, all of
which can be used for Playback and Background. However, Asterisk does
not understand ADPCM WAV files. To convert your WAV files to a format
which Asterisk can understand, use the following command:

   sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql

2010/1/21 Kyle Kienapfel :
> The playback command is designed to work with multiple formats
> If the channel in question is gsm it'll use a .gsm file before a .wav file
>
> if the .wav file is in the directory, is it playable by asterisk?
> (8000hz sample rate, etc etc)
>
> On Tue, Jan 19, 2010 at 8:20 AM, Danny Nicholas  wrote:
>> Just a WAG - Playback is freaking out because you have the wav and gsm file
>> there concurrently.  Remove the wav file and try it again.
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zhang Shukun
>> Sent: Tuesday, January 19, 2010 12:47 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] wav to gsm can't play
>>
>> hi,
>>
>> i try to convert wav file to gsm format.use following commands;
>>
>> sox net263-welcome.wav -r 8000 -g -c 1 net263-welcome.gsm resample -ql
>>
>> the file is located in /var/lib/asterisk/sounds/net263
>>  but cant' play.do you know what's wrong?
>>
>>    -- Executing Playback("SIP/1001-0091", "net263/net263-welcome")
>>    --  Playing 'net263/net263-welcome' (language 'en')
>> [Jan 19 14:36:06] WARNING[21397]: app_playback.c:440 playback_exec:
>> ast_streamfile failed on SIP/1001-0091 for net263/net263-welcome
>>    -- Executing WaitExten("SIP/1001-0091", "60")
>>
>>
>> --
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>> Sucan
>>
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[asterisk-users] Which to choose? Realtime extension OR Static extension with MYSQL command

[Zhang Shukun]

hi,all

one thing confused me these days. i don't know which method to choose,
and don't know which one is better perfoermance than another when in
production system.

i can save dialplan in the extension table , i also can write dialplan
in extension.conf with MYSQL commmand to fetch data from database.

but which one is better?

Could you shed some light to me?

Any suggestion will be appreciated.

Thank you!

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[asterisk-users] wav to gsm can't play

[Zhang Shukun]

hi,

i try to convert wav file to gsm format.use following commands;

sox net263-welcome.wav -r 8000 -g -c 1 net263-welcome.gsm resample -ql

the file is located in /var/lib/asterisk/sounds/net263
 but cant' play.do you know what's wrong?

-- Executing Playback("SIP/1001-0091", "net263/net263-welcome")
--  Playing 'net263/net263-welcome' (language 'en')
[Jan 19 14:36:06] WARNING[21397]: app_playback.c:440 playback_exec:
ast_streamfile failed on SIP/1001-0091 for net263/net263-welcome
-- Executing WaitExten("SIP/1001-0091", "60")


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Re: [asterisk-users] How to play the voicemail recorded?

[Zhang Shukun]

Sorry. I can hear now. last time i have not record successfully.

2010/1/18 Zhang Shukun :
> Hi,all
>
> i want to hear the voicemail recorded, but when hear "if you want to
> play message , press 3", after i press 3
>
> i only hear that that's the time the file recorded. not the content.
> do you know how to hear content of voicemail fle?
>
> debug message:
>
>  == Parsing '/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001.txt':
> Found
>    --  Playing 'vm-received' (language 'en')
>    --  Playing 'digits/at' (language 'en')
>    --  Playing 'digits/2' (language 'en')
>    --  Playing 'digits/30' (language 'en')
>    --  Playing 'digits/9' (language 'en')
>    --  Playing 'digits/p-m' (language 'en')
>    --  Playing
> '/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001' (language
> 'en')
>
>
> --
> Best regards,
> Sucan
>



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[asterisk-users] What's customer_id mean?

[Zhang Shukun]

hi ,all

I do'nt know exactly what  customer_id mean?  while  if i have
password i could visit the voicemail box.

CREATE TABLE voicemail_users (
 uniqueid int(11) NOT NULL auto_increment,
 customer_id int(11) NOT NULL default '0',
 context varchar(50) NOT NULL default '',
 mailbox int(5) NOT NULL default '0',
 password varchar(4) NOT NULL default '0',
 fullname varchar(50) NOT NULL default '',
 email varchar(50) NOT NULL default '',
 pager varchar(50) NOT NULL default '',
 stamp timestamp(14) NOT NULL,
 PRIMARY KEY  (uniqueid),
 KEY mailbox_context (mailbox,context)
) TYPE=MyISAM;

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[asterisk-users] How to play the voicemail recorded?

[Zhang Shukun]

Hi,all

i want to hear the voicemail recorded, but when hear "if you want to
play message , press 3", after i press 3

i only hear that that's the time the file recorded. not the content.
do you know how to hear content of voicemail fle?

debug message:

  == Parsing '/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001.txt':
Found
--  Playing 'vm-received' (language 'en')
--  Playing 'digits/at' (language 'en')
--  Playing 'digits/2' (language 'en')
--  Playing 'digits/30' (language 'en')
--  Playing 'digits/9' (language 'en')
--  Playing 'digits/p-m' (language 'en')
--  Playing
'/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001' (language
'en')


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[asterisk-users] Will SIP connection stop automaticlly when detect no voice between the channel after a period of time?

[Zhang Shukun]

hi,

in my test, i noticed that sip connection will hangup automaticlly
when no speaks between the channel. about half a minute.

is this the asterisk inner mechanism or is my configuration error?

Thanks!

messages on the cli as follow:

-- SIP/1003-001d is ringing
-- SIP/1003-001d answered SIP/1004-001c
-- Stopped music on hold on SIP/1004-001c
[Jan 18 10:08:42] WARNING[17022]: app_queue.c:3268 try_calling: The
device state of this queue member, Zhang Jianming, is still 'Not in
Use' when it probably should not be! Please check UPGRADE.txt for
correct configuration settings.
-- Packet2Packet bridging SIP/1004-001c and SIP/1003-001d
-- Executing Playback("SIP/1004-001c", "vm-goodbye")
[Jan 18 10:09:13] WARNING[17022]: file.c:764 ast_readaudio_callback:
Failed to write frame
--  Playing 'vm-goodbye' (language 'en')
[Jan 18 10:09:13] WARNING[17022]: app_playback.c:440 playback_exec:
ast_streamfile failed on SIP/1004-001c for vm-goodbye
  == Spawn extension (95040654321, 1, 2) exited non-zero on 'SIP/1004-001c'
[Jan 18 10:09:18] NOTICE[16700]: chan_sip.c:16209
handle_request_subscribe: Received SIP subscribe for peer without
mailbox: 1003

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Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4

[Zhang Shukun]

I suggest you install it from source, that way you can learn
more about asterisk.

2010/1/16 William Stillwell (Lists) :
> Here is the 1.4.x version on centos 5 walk through.
>
>
>
> http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
>
>
>
>
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Nik
> Sent: Friday, January 15, 2010 3:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script
> for CentOS 5.3 or 5.4
>
>
>
> Provided there is no comprehensive install guides (or is there?) yes I would
> like to see an easy install script which can install it all.
>
>
>
> On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik  wrote:
>
> Hi Guys,
>
>
>
> Other than than yum repository (which fails when installing freepbx with it)
> are there any automated install scripts out there that would install
> Asterisk 1.6 or 1.4 onto a CentOS LAMP system?
>
>
>
> If the script install FreePBX that would be a BONUS.
>
>
>
> Thanks,
>
> Bruce
>
>
>
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Re: [asterisk-users] Realtime queue not work

[Zhang Shukun]

2010/1/15 Robert Broyles :
> Zhang Shukun wrote:
>> 2010/1/15 Leif Neland :
>>
>>> - Original Message -
>>> From: Zhang Shukun
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Sent: Friday, January 15, 2010 11:48 AM
>>> Subject: [asterisk-users] Realtime queue not work
>>> hi, all
>>>
>>> i try to confiture realtime queue, but not work, details as below:
>>>
>>> Insert into queue_table(name)value('95040654321');
>>>
>>> INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
>>> '95040654321', 'SIP/1001', 2, 1);
>>> INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
>>> 'SIP/1002', 2, 1);
>>> INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
>>> '95040654321', 'SIP/1003', 2, 1);
>>>
>>> but when i dial 95040654321 and press extension 1. error happens:
>>>
>>>  -- Executing Queue("SIP/1003-", "950406543211")
>>> [Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
>>> to join queue '950406543211'
>>>   == Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'
>>>
>>>
>>> No golden answers, but something to try.
>>>
>>> queue names can not be just numbers? I'd try calling the queue
>>> "q95040654321".
>>>
>>
>> Thank you for reply. i've try "a95040654321" as the queue name but not
>> work either.
>>
>> there was the same error in the cli.
>>
>>
>>> Does "show queues" show the queue? Don't know if that's supposed to work on
>>> realtime queues.
>>>
>>>
>>> Leif
>>>
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>>>
>>
>>
>>
>>
> Which version of Asterisk are you using?  Can you paste ( use
> pastebin.com please ) your extconfig.conf and  res_mysql.conf  (or
> res_odbc.conf)?

Thanks.

I use Asterisk version 1.4.28.

in extconfig.conf related to queue is queues and queue_members:
--
queues => mysql,asterisk,queue_table
queue_members => mysql,asterisk,queue_member_table
--

res_mysql.conf file is configured right i think, because i have
sipusers, sippeers, and
extensions realtime work properly.


> Something must not be configured right.
>
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Re: [asterisk-users] Realtime queue not work

[Zhang Shukun]

2010/1/15 Robert Broyles :
> Leif Neland wrote:
>>
>>
>>     - Original Message -
>>     *From:* Zhang Shukun <mailto:bit...@gmail.com>
>>     *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>     <mailto:asterisk-users@lists.digium.com>
>>     *Sent:* Friday, January 15, 2010 11:48 AM
>>     *Subject:* [asterisk-users] Realtime queue not work
>>
>>     hi, all
>>
>>     i try to confiture realtime queue, but not work, details as below:
>>
>>     Insert into queue_table(name)value('95040654321');
>>
>>     INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
>>     '95040654321', 'SIP/1001', 2, 1);
>>     INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
>>     'SIP/1002', 2, 1);
>>     INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
>>     '95040654321', 'SIP/1003', 2, 1);
>>
>>     but when i dial 95040654321 and press extension 1. error happens:
>>
>>      -- Executing Queue("SIP/1003-", "950406543211")
>>     [Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
>>     to join queue '950406543211'
>>       == Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'
>>
>>
>> No golden answers, but something to try.
>>
>> queue names can not be just numbers? I'd try calling the queue
>> "q95040654321".
>>
>> Does "show queues" show the queue? Don't know if that's supposed to
>> work on realtime queues.
>>
>>
>> Leif
>>
> Yes, from my experience -  'queue show' will show realtime queues.
> 'show queues' technically is deprecated in 1.4, but should give the same
> results.

Thank you! as i am at home now.
 i can't test if 'queue show' will list the queue 'a95040654321'

I will test it in the future.

but i guess from the warning message:

 Unable to join queue '950406543211'

is this mean the queue has not been set up ?



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Re: [asterisk-users] Realtime queue not work

[Zhang Shukun]

2010/1/15 Leif Neland :
>
>
> - Original Message -
> From: Zhang Shukun
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Sent: Friday, January 15, 2010 11:48 AM
> Subject: [asterisk-users] Realtime queue not work
> hi, all
>
> i try to confiture realtime queue, but not work, details as below:
>
> Insert into queue_table(name)value('95040654321');
>
> INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
> '95040654321', 'SIP/1001', 2, 1);
> INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
> 'SIP/1002', 2, 1);
> INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
> '95040654321', 'SIP/1003', 2, 1);
>
> but when i dial 95040654321 and press extension 1. error happens:
>
>  -- Executing Queue("SIP/1003-", "950406543211")
> [Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
> to join queue '950406543211'
>   == Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'
>
>
> No golden answers, but something to try.
>
> queue names can not be just numbers? I'd try calling the queue
> "q95040654321".

Thank you for reply. i've try "a95040654321" as the queue name but not
work either.

there was the same error in the cli.

>
> Does "show queues" show the queue? Don't know if that's supposed to work on
> realtime queues.
>
>
> Leif
>
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[asterisk-users] Realtime queue not work

[Zhang Shukun]

hi, all

i try to confiture realtime queue, but not work, details as below:

Insert into queue_table(name)value('95040654321');

INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
'95040654321', 'SIP/1001', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
'SIP/1002', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
'95040654321', 'SIP/1003', 2, 1);

but when i dial 95040654321 and press extension 1. error happens:

 -- Executing Queue("SIP/1003-", "950406543211")
[Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
to join queue '950406543211'
  == Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'

DO you know what's wrong?

Thank you!

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[asterisk-users] different between ring groups and queue?

[Zhang Shukun]

hi,all

while you can set ring groups in the queue.conf file. what's diff between them.

is that all members in a queue are a group? if that true.

it should not need to define group at all.

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[asterisk-users] What about the performance visit MYSQL in DialPlan code?

[Zhang Shukun]

Hi,all

What about the performance visit MYSQL in DialPlan code? if use MySQL
RealTime connection

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Re: [asterisk-users] is there some Chinese version of sounds available?

[Zhang Shukun]

Thank you!

2010/1/14 Lee, John (Sydney) :
>> when use the VoiceMail , all the directions all english. i want to
>> know is there some Chinese version of sounds available now?
>>
>> or should i record it myself?
>
> http://www.voip-info.org/wiki/view/Asterisk+sound+files+international
> Look under Chinese (Mandarin)
>
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Re: [asterisk-users] is roundrobin and rrmemory the same meaning?

[Zhang Shukun]

Thanks for your reply.

I have read the Asterisk Realtime Architecture feature of Asterisk.

it says that we can save queue and queue_members in the database. and
queue_member don't need to login( because not support). and when
queue_member changed in database. don't need reload cant asterisk use
soon.

Has anyone deployed the ARA feature of Asterisk? and How do you think
about this feature?

2010/1/13 Robert Lister :
> On Wed, 2010-01-13 at 10:42 +0800, Zhang Shukun wrote:
>
>> is there some function used to login a agent automaticlly like
>>
>> agentlogin("agentname","agentpassword","phonenumber")?
>
> Depends what version you are running.
>
> AgentCallBackLogin() is deprecated and you should not use it.
> But the feature can be reproduced with dialplan logic.
>
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%
> 20AgentCallbackLogin
>
> This is a whole world of pain, as is using Agents in some situations.
> It is better to use SIP channels. (Agents do not seem to work nicely
> with a bunch of other features.) It is less flexible.
>
> It may be better for you to do this using AddQueueMember and
> RemoveQueueMember on SIP channels, and program a key (or keys) on the
> handset to add and remove the member from the queue dynamically instead
> of adding them as static members in queues.conf.
>
>
> Rob
>
>
>
>
>
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[asterisk-users] is there some Chinese version of sounds available?

[Zhang Shukun]

hi ,all

when use the VoiceMail , all the directions all english. i want to
know is there some Chinese version of sounds available now?

or should i record it myself?

just like:

Here is what you can do with your mailbox using VoiceMailMain.


1 Old Messages


3 Advanced options


1 Send reply
2 Call back
3 Envelope
4 Outgoing call
5 Leave message
* Return to main menu


4 Play previous message
5 Repeat current message
6 Play next message
7 Delete current message
8 Forward message to another mailbox
9 Save message in a folder
* Help; during msg playback: Rewind
# Exit; during msg playback: Fastforward


2 Change folders
3 Advanced options
0 Mailbox options


1 Record your unavailable message
2 Record your busy message
3 Record your name
4 Change your password
* Return to the main menu


* Help
# Exit



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Re: [asterisk-users] is roundrobin and rrmemory the same meaning?

[Zhang Shukun]

2010/1/13 Robert Lister :
> On Tue, 2010-01-12 at 11:26 +0800, Zhang Shukun wrote:
>> Dear all,
>>
>> I can't understand the diff between roundrobin and rrmemory strategy.
>> Could you explain for me ?
>>
>> and is roundrobin means each available interface ring once or several
>> times and ring another?
>
> roundrobin is deprecated in 1.4 and you probably shouldn't use it, but
> rrmemory is probably what you want, trying each extension in order,
> but continuing the position in the queue where it left off for
> subsequent calls.
>
> roundrobin always starts at the top of the queue and works along
>
> rrmemory remembers which queue member was tried last, and continues for
> subsequent calls from where it left off, rather than starting again from
> the top of the queue.
>
> In 1.6, the old "roundrobin" behaviour (or equivalent) is renamed
> "linear" and "rrmemory" is renamed "roundrobin"
>
Thank you! you explained very clear above about the two concept!

> If you want to add some dialplan actions for queue members, have a look
> at PauseQueueMember and UnpauseQueueMember which allows for queue
> members to be 'in' and 'out' of the group (although if using Agents then
> you will probably want to implement agents logging in and out), but you
> could replace agents with dynamic queues and program buttons on the
> phones which dial codes to pause and unpause the queue member.

is there some function used to login a agent automaticlly like

agentlogin("agentname","agentpassword","phonenumber")?

>
>
> Rob
>
>
>
>
>
>
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Re: [asterisk-users] is roundrobin and rrmemory the same meaning?

[Zhang Shukun]

2010/1/12 Lenz Emilitri :
> You can list phones directly as static members of the queue.

i know i can configure the queue.conf and agents.conf to add queue
name and queue members by hand.

Could i use functions to create queue name and add queue members dynamiclly.

because i want to create a call center use asterisk, which users can
register their own call number on the web site.

also they can add several service phone numbers along with a fix
extension (like:1), the phone numbers are customer

service number, when it's customer dial the "call number" and press
extension 1, one member should answer the caller.

so, when configured on the web. like:

extension 1:12345, 12346,12347,12348,12349

when finished the data above should stored in the database, when user
call in and press 1.

i should create a queue and add 12345, 12346,12347,12348,12349 to the queue.

is this possible?

>  this is generally sub.optimal because if. e.g. an agent of yours is home 
> sick, her
> phone will be ringing and you'll be wasting caller time. Also by tracking
> logins and logoffs you can measure agent productivity, and this is pretty
> useful in most environments.
because all the phones are office phone, if one phone can't response
as on people there , with 15 secs timeout,

it will select another phone in the queue according the strategy. so
it doesn't matter.
> l.
>
>
> 2010/1/12 Zhang Shukun 
>>
>> Thank you! it's very helpful.
>>
>> now i have another question:
>>
>> in asterisk, each agent should login first and then can response to
>> the caller. but i don't want to the login action.
>>
>> i need agent shold response directly without login first. how should i do
>> ?
>>
>> can users in sip.conf to be agents? so it can login  persistently on a
>> phone.
>
> --
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>
>
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Re: [asterisk-users] is roundrobin and rrmemory the same meaning?

[Zhang Shukun]

Thank you! it's very helpful.

now i have another question:

in asterisk, each agent should login first and then can response to
the caller. but i don't want to the login action.

i need agent shold response directly without login first. how should i do ?

can users in sip.conf to be agents? so it can login  persistently on a phone.


2010/1/12 Tony Mountifield :
> In article ,
> Zhang Shukun  wrote:
>> Dear all,
>>
>> I can't understand the diff between roundrobin and rrmemory strategy.
>> Could you explain for me ?
>>
>> and is roundrobin means each available interface ring once or several
>> times and ring another?
>>
>> ; A strategy may be specified.  Valid strategies include:
>> ;
>> ; ringall - ring all available channels until one answers (default)
>> ; roundrobin - take turns ringing each available interface
>> ; leastrecent - ring interface which was least recently called by this queue
>> ; fewestcalls - ring the one with fewest completed calls from this queue
>> ; random - ring random interface
>> ; rrmemory - round robin with memory, remember where we left off last ring 
>> pass
>> ;
>> ;strategy = ringall
>
> Both roundrobin and rrmemory will ring phones one at a time, for the
> length of time given in "timeout", and then if not answered will move
> along to the next phone and ring it.
>
> Let's say you have three of more phones in the queue. Phone 1 gets rung
> but not answered, then Phone 2 gets rung and is answered. When another
> call comes in, roundrobin would start again with Phone 1, but rrmemory
> would start with Phone 3, as it was Phone 2 that picked up the last call.
>
> Hope this helps,
> Tony
> --
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> Work: t...@softins.co.uk - http://www.softins.co.uk
> Play: t...@mountifield.org - http://tony.mountifield.org
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[asterisk-users] Why agent log out automaticly？

[Zhang Shukun]

hi,all

when in talking status, agent log out automaticly, why? following are
output in CLI*

do you know the reason？

  == Agent '1002' logged in (format ulaw/ulaw)
-- Executing [...@tutorial:1] Queue("SIP/ivan-0013", "queue1")
in new stack
-- Started music on hold, class 'default', on SIP/ivan-0013
-- Stopped music on hold on SIP/3172841667-0012
-- agent_call, call to agent '1002' call on 'SIP/3172841667-0012'
--  Playing 'beep' (language 'en')
-- Stopped music on hold on SIP/zsk-0006
-- agent_call, call to agent '1001' call on 'SIP/zsk-0006'
--  Playing 'beep' (language 'en')
-- Agent/1002 answered SIP/ivan-0013
-- Started music on hold, class 'default', on SIP/zsk-0006
-- Stopped music on hold on SIP/ivan-0013
  == Spawn extension (tutorial, 29, 1) exited non-zero on 'SIP/ivan-0013'
  == Agent '1002' logged out
  == Spawn extension (tutorial, 27, 1) exited non-zero on
'SIP/3172841667-0012'


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[asterisk-users] is roundrobin and rrmemory the same meaning?

[Zhang Shukun]

Dear all,

I can't understand the diff between roundrobin and rrmemory strategy.
Could you explain for me ?

and is roundrobin means each available interface ring once or several
times and ring another?

; A strategy may be specified.  Valid strategies include:
;
; ringall - ring all available channels until one answers (default)
; roundrobin - take turns ringing each available interface
; leastrecent - ring interface which was least recently called by this queue
; fewestcalls - ring the one with fewest completed calls from this queue
; random - ring random interface
; rrmemory - round robin with memory, remember where we left off last ring pass
;
;strategy = ringall


Thanks!
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Re: [asterisk-users] You won't help me anymore?

[Zhang Shukun]

you'd better paste your dialplan snip here, in order to get specific help.

2010/1/11 Darrick Hartman :
> On 01/10/2010 11:38 PM, hadi motamedi wrote:
>>
>>     FWIW, he did post his question yesterday. I've just taken a look and
>>     one potential issue I've spotted is that the external server he
>>     mentions is 192.168.0.139, which is part of the 192.168.0.0/16
>>     
>>     netblock reserved for private networks. So while the server might be
>>     192.168.0.139 on it's own LAN, I suspect that won't be its public IP
>>     address.
>>
>>     Other than that, I suspect there might be an issue with the dialplan.
>>     The OP posted an excerpt from his sip.conf but I suspect we'd need
>>     his extensions.conf or extensions.ael (whichever or both he's using)
>>     before being able to help further.
>>
>>
>>
>> Thank you very much for your reply . My Asterisk CallerId issue is as
>> the followings :
>> "My Asterisk has sip connection with external sip server and sip inbound
>> and outbound calls are ok . But for the sip inbound calls when the
>> external sip server sends SIP INVITE with CallerId field in the range of
>> my Asterisk sip phones the call will be rejected . For example , please
>> imagine that my Asterisk sip phones are at 667  range so when the
>> external sip server places sip inbound call with SIP INVITE CallerId as
>> say 667 2020 the call will be rejected . But if he modifies his CallerId
>> to say 021 667 2020 (i.e. with area code included) the call will get
>> through . Can you please let me know what is the problem here ?
>>
>
> It sounds like a dialplan issue where you don't have a pattern which
> matches 6662020 while you do have something that matches 0216672020.
> Without seeing the dialplan, we can only guess.
>
> --
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> DJH Solutions, LLC
> http://www.djhsolutions.com
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Re: [asterisk-users] You won't help me anymore?

[Zhang Shukun]

in your dialplan ,did you add area code automaticly? when dial out.

2010/1/11 hadi motamedi :
>
>
> On Sun, Jan 10, 2010 at 2:43 PM, Geoff Lane  wrote:
>>
>> On Sunday, January 10, 2010, Francesco Peeters wrote:
>>
>> > Yes, post your question clear and consicely, include all relevant
>> > information and snip all unneccessary history.
>>
>> > Note that: no reply != not wanting to help...
>> > It *is* obviously possible people just do not KNOW the answer!... (Oh
>> > what shock and horror!!!)
>>
>> FWIW, he did post his question yesterday. I've just taken a look and
>> one potential issue I've spotted is that the external server he
>> mentions is 192.168.0.139, which is part of the 192.168.0.0/16
>> netblock reserved for private networks. So while the server might be
>> 192.168.0.139 on it's own LAN, I suspect that won't be its public IP
>> address.
>>
>> Other than that, I suspect there might be an issue with the dialplan.
>> The OP posted an excerpt from his sip.conf but I suspect we'd need
>> his extensions.conf or extensions.ael (whichever or both he's using)
>> before being able to help further.
>>
>> HTH,
>>
>> --
>> Geoff
>>
>>
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>
>
>
> Thank you very much for your reply . My Asterisk CallerId issue is as the
> followings :
> "My Asterisk has sip connection with external sip server and sip inbound and
> outbound calls are ok . But for the sip inbound calls when the external sip
> server sends SIP INVITE with CallerId field in the range of my Asterisk sip
> phones the call will be rejected . For example , please imagine that my
> Asterisk sip phones are at 667  range so when the external sip server
> places sip inbound call with SIP INVITE CallerId as say 667 2020 the call
> will be rejected . But if he modifies his CallerId to say 021 667 2020 (i.e.
> with area code included) the call will get through . Can you please let me
> know what is the problem here ?
>
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[asterisk-users] How to test if a telephone is busy now?

[Zhang Shukun]

hi, all

i want to test if a telephone is busy now in agi php script?

Could you tell me how to do that judgement？


example：

if( ivan is not busy)
{
 $agi -> exec_dial("SIP","ivan");
}
else if (test is not busy)
{
$agi -> exec_dial("SIP","test");
}

Thanks very much!

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[asterisk-users] How to use AGI php script function $agi -> exec_dial

[Zhang Shukun]

hi,

i want to use $agi -> exec_dial() to dial .

this is in extention.conf

[tutorial]
exten => 1234,1,Dial(SIP/ivan)

is that i use

$agi -> exec_dial("SIP","tutorial|1234|1")

can dial ?

BTW, i want to know some turorial on how to use PHPAGI funtions? can
you tell me some?

Thanks!


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[asterisk-users] Zhang Shukun 想跟您聊天

[Zhang Shukun]

---
Zhang Shukun希望通过 Google 的一些最炫的新产品与您保持更密切的联系。

如果您已经拥有 Gmail 或 Google Talk，请访问：
http://mail.google.com/mail/b-1f731adb8c-861c77d1cc-d68ecfc46b4cc7de
您需要点击此链接才能与Zhang Shukun聊天。

要获取 Gmail（Google 提供的免费电子邮件帐户，存储空间超过 2,800 MB）并与Zhang Shukun聊天，请访问：
http://mail.google.com/mail/a-1f731adb8c-861c77d1cc-d68ecfc46b4cc7de

Gmail 提供以下功能：
- 直接在 Gmail 中进行即时消息传递
- 强大的垃圾邮件防护功能
- 可用于查找邮件的内置搜索功能，以及实用的邮件整理方法（将邮件整理到“会话”中）
- 没有弹出式广告或不相干的横幅广告，只显示文字广告和与邮件内容相关的信息
以上所有功能均免费为您提供。此外，我们还提供了更多服务！打开 Gmail 帐户后，您还可以访问 Google Talk（即时消息传输服务）：

http://www.google.com/talk/intl/zh-CN/

Google Talk 提供以下功能：
- 无需下载即可供您在任何地方使用的基于网络的聊天功能
- 与 Gmail 帐户同步的联系人列表
- 免费、高品质的 PC 间语音呼叫服务（下载 Google Talk 客户端后）

我们一直致力于增加新的功能并不断改进服务，因此我们还可能会定期请您提供意见和建议。感谢您为我们的产品改进所提供的帮助！

此致
Google 小组敬上

要了解有关 Gmail 和 Google Talk 的详细信息，请访问：
http://mail.google.com/mail/help/intl/zh_CN/about.html
http://www.google.com/talk/intl/zh-CN/about.html

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[asterisk-users] How to recieve number returned by $AGI->wait_for_digit($timeout)

[Zhang Shukun]

hi,

i use $AGI->wait_for_digit($timeout)  to wait for the user press key 1
,and then to do something.

but how can i get the return number ?

is that use $key = $AGI->wait_for_digit($timeout)

and $key will be "200 result=49" if i pressed number 1?

Thanks!

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Re: [asterisk-users] How to dial a number make two phone Ring at the same time?

[Zhang Shukun]

Thank you!
but how can i determine whether ring at the same time or

alternative ring?

BTW, the uri

http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con

can't open.

Could you paste it again?

2010/1/7 Randy R :
> On Thu, Jan 7, 2010 at 2:38 PM, Zhang Shukun  wrote:
>> hi,
>>
>> i want to dial a number to let two phone ring at the same time or
>> alternative ring,
>>
>> how should i configure in asterisk? or how to right the Dialplan code?
>
> exten => 12345,1,Dial(${PHONE1}&${PHONE2})
>
> each phone variable is defined as stated in docs depending on the
> protocol, SIP, IAX2, etc
>
> as in
>
> exten => s,1,Dial(SIP/2000)
>
> So PHONE1 would be SIP/2000
>
> See
>
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con
>
>
> /r
>
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[asterisk-users] How to dial a number make two phone Ring at the same time?

[Zhang Shukun]

hi,

i want to dial a number to let two phone ring at the same time or
alternative ring,

how should i configure in asterisk? or how to right the Dialplan code?

Thanks very much!
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Sucan

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Re: [asterisk-users] How to see STDERR message?

[Zhang Shukun]

Thank you for you reply?

is that mean STDERR couldn't show under Asterisk CLI mode?

it's only saved to some file?

2010/1/7 Steve Edwards :
> On Thu, 7 Jan 2010, Zhang Shukun wrote:
>
>> i use agi to send message back to Asterisk by STDERR, but why i could't
>> see the message in asterisk CLI?
>
> Output to STDERR does nothing for me either.
>
> I prefer to use syslog() to log the messages via syslogd.
>
> --
> Thanks in advance,
> -
> Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                              Fax: +1-760-731-3000
>
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Sucan

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[asterisk-users] How to see STDERR message?

[Zhang Shukun]

hi,

i use agi to send message back to Asterisk by STDERR, but why i
could't see the message in asterisk CLI?

i start asterisk use " asterisk -vc" in order to see all message.

Thanks

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Re: [asterisk-users] error when open a2billing web page!

[Zhang Shukun]

OK. Thanks

2009/12/29 ram :
>
>
> On Mon, Dec 28, 2009 at 10:39 PM, Zhang Shukun  wrote:
>>
>> hi,
>>   i have installed a2billing , when i open /admin web pages. errors as
>> follow:
>>
>> Fatal error: Call to undefined function bindtextdomain() in
>> /usr/local/src/a2billing/common/lib/languageSettings.php on line 130
>>
>> do you know what's wrong?
>>
>
> you get quick responce if you post the same in a2bill forum
>
> look at their site forum
>
> Ram
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[asterisk-users] error when open a2billing web page!

[Zhang Shukun]

hi,
   i have installed a2billing , when i open /admin web pages. errors as follow:

Fatal error: Call to undefined function bindtextdomain() in
/usr/local/src/a2billing/common/lib/languageSettings.php on line 130

do you know what's wrong?

-- 
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[asterisk-users] Does A2Billing has mial list?

[Zhang Shukun]

hi,

Does A2Billing has mial list?

-- 
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Sucan

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