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

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_senddig

[asterisk-users] How to execute Asterisk Functions in PHPAGI

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:

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

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 Origina

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

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

[asterisk-users] How to Billing for MeetMe Conference?

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 f

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

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! -- Thanks & Regards Sucan -- _ -- Bandwidt

[asterisk-users] codec_g729.so not work!

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 Forma

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

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 t

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

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 mar

[asterisk-users] MeetMe VS. Conference

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)? -- Thanks & Regards Sucan -- _ --

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

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. &

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

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! -- Thanks for your supporting, have a nice day. Sucan -- _

Re: [asterisk-users] GotoIfTime problem

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") >> &

[asterisk-users] GotoIfTime problem

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

[asterisk-users] Is Centos 64 bit or 32 bit better?

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

[asterisk-users] What‘s the best operating sys tem suggest for Asterisk 1.6.2.9

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. -- Thanks for your supporting, have a nice day. Sucan -- _

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

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 pla

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

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 c

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

ATION, 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,a

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

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! --

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

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 'bitzs

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

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 successful

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

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! -- Thanks for your supporting, have a nice d

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

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

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

hi, all i want to use PHP agi to do as a soap client. does php agi support this function? Thanks! -- Thanks for your supporting, have a nice day. Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -

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

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=

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

ck("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 no

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

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

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

   480 Temporarily unavailable >   21 call rejected                        403 Forbidden (+) >   22 number changed (w/o diagnostic)      410 Gone >   22 number changed (w/ diagnostic)       301 Moved Permanently >   23 redirection to new destination       410 Gone >   26 non-selected use

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

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 comman

Re: [asterisk-users] Time difference in CSV CDR's and MySQL CDR's

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 >> th

Re: [asterisk-users] Time difference in CSV CDR's and MySQL CDR's

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 > --

Re: [asterisk-users] G729 exhaustion conditions

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 licens

Re: [asterisk-users] Detect if a Number is up or not

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 ma

[asterisk-users] Set CDR amaflags not work

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! -- Best regards, Sucan -- _ -- Bandwidth and C

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

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 Discus

[asterisk-users] How to get Sip response codes in Dialplan?

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 th

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

- 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 ca

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

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 o

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

ope 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. i

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

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, ho

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

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(wel

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

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! -- Best regards, Sucan -- __

[asterisk-users] realtime modules not load ?

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 l

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

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

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

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 .

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

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. -- Best

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

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

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

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

[asterisk-users] Does Playback will answer the call?

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 ? -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provide

[asterisk-users] Realtime queue strategy issue

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

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

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,

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

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 wor

Re: [asterisk-users] MYSQL problem

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 ?

Re: [asterisk-users] GoToIfTime issue

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) >> >> >> >>

Re: [asterisk-users] MySQL RealTime Error

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 >> > >>

Re: [asterisk-users] MYSQL problem

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. Than

Re: [asterisk-users] MYSQL grammar diff in 1.6.2.1?

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 >

[asterisk-users] Realtime Queue not work in 1.6.2.1

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

[asterisk-users] MYSQL grammar diff in 1.6.2.1?

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:

[asterisk-users] MySQL RealTime Error

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://

Re: [asterisk-users] GoToIfTime issue

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) >> >> >> >>

Re: [asterisk-users] GoToIfTime issue

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

[asterisk-users] GoToIfTime issue

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 '

[asterisk-users] MYSQL problem

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! -- Best regards, Sucan -- __

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

 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 Mailin

[asterisk-users] Which to choose? Realtime extension OR Static extension with MYSQL command

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 dat

[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/net2

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

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 re

[asterisk-users] What's customer_id mean?

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

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

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/

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

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

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

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-bou

Re: [asterisk-users] Realtime queue not work

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 &g

Re: [asterisk-users] Realtime queue not work

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@lis

Re: [asterisk-users] Realtime queue not work

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 confitur

[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

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

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. -- Best regards, Sucan -- _ -- Bandw

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

Hi,all What about the performance visit MYSQL in DialPlan code? if use MySQL RealTime connection -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To

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

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

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

terisk 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 >> >> age

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

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 C

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

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 inte

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

vironments. 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!

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

nts? 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 round

[asterisk-users] Why agent log out automaticly？

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

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

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 chan

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

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 h

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

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 >> > informa

[asterisk-users] How to test if a telephone is busy now?

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! -- Best regards, Su

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

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! -- Best r

[asterisk-users] Zhang Shukun 想跟您聊天

--- 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

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

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! -- Best regards, Sucan __

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

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 Shuku

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

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! -- Best regards, Sucan ___ -- Bandwidth and Colocation Provided by http:/

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

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 >>

[asterisk-users] How to see STDERR message?

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 -- Best regards, Sucan ___ -- Bandwidth and Colocation Provided

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

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 bindtextdomai

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

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? -- Thanks, Sucan __

[asterisk-users] Does A2Billing has mial list?

hi, Does A2Billing has mial list? -- Thanks, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user

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