thanks for your replay,
but i am not able to set this fecility in agent phone.
any other solution ?
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hi, all
Is ther any way to set 3-way conference using queue app other other way
using queue app.
scenario:
custmore call to queue , agent answered than agent transfer to third
persion, so the three
call communicate with each other.
Regards,
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Bhrugu Mehta
Sr. S/W Engineer (DD)
VOIP
hi, all
Is ther any way to set up call-waiting feature in asterisk using dialplan or
any other ways. I want to use only
asterisk for that not any other gui.
I am using asterisk 1.4.28.
Regards,
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Bhrugu Mehta
Sr. S/W Engineer (DD)
VOIP,Telephony Team
India
Hi, all
how to get hold event in asterisk.
is it possible, when user1 put on hold in queue moh1 file played.
when call transfer to agent and answered agent put hold at that time
moh2 file played ?
I have used asterisk 1.4 version.
Regards,
--
Bhrugu Mehta
Sr. S/W Engineer (DD)
VOIP,Telephony
Thnks for ur reply,
SendImage() doesn't work with asterisk sip channel.
any other solution?
Regards,
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Sr. S/W Engineer (DD)
VOIP,Telephony Team
India
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hi, all
is there any way to send image on sip channel ?
Regards,
--
Bhrugu Mehta
Sr. S/W Engineer (DD)
VOIP,Telephony Team
India
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Hi, all
What is the use of astdb?
Is it used to store realtime values like sip etc.
Regards,
Bhrugu Mehta
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PM, bhrugu mehta mehtabhr...@gmail.com wrote:
Hi, all
What is the use of astdb?
Is it used to store realtime values like sip etc.
Regards,
Bhrugu Mehta
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Bhrugu Mehta
Sr. S/W Engineer (DD)
VOIP,Telephony Team
India
Hi, all
Is ther any way to pass channel queue such a way
Queue(SIP/1001SIP/1002SIP/1003)
thanks,
Bhrugu Mehta
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. Generaly , g729 is greater.
regards,
Bhrugu Mehta
On 5/16/08, gincantalupo [EMAIL PROTECTED] wrote:
Hi,
hope not to be OT :)
after more than 3 years of PBX installations we can adfirm Asterisk is
stable enough to be considered a good product but still we encounter a
lot of problems when
hi,
I have not tested that but I have seen 100 agents configure with asterisk.
thnks
Bhrugu mehta
On 5/15/08, gmail [EMAIL PROTECTED] wrote:
Is Asterisk practically stable and reliable for a larg Enterprise has say a
1 phones , is there any case study like
, Bhrugu Mehta
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hi, all
i am using zma800p card( from zapmicro).
i create small ivrs.
when i call on fxs channel calls lended and ivrs start on that channel.
but when i use callerid app. from asterisk , doesn't displayed any
callerid on asterisk.
any suggestion.
thanks in advance.
Bhrugu mehta
context=mycontext
signalling=fxl_ls
group=1
channel=1-4
thanks' in advance.
Bhrugu mehta
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, in advance
Bhrugu Mehta
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=ipofserver2
context=any
in server2, sip.conf
[user]
type=friend
username=user
secret=user
host=dynamic
context=anyyouwant
Bhrugu Mehta (SAI INFO SYSTEM LTD.)
On 2/15/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I want that an sjphone registered using serverA can call to an sjphone
registered
hi, all
I want to use two zaptel card(TE210p) in pc for asterisk.
Is there any special requirement for this configuratin.
any suggestion.
thanks ,
Bhrugu Mehta
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hi,
I have used asterisk 1.2.12.1 and using linux 4 enterprise edition.
Bhrugu Mehta
On Jan 22, 2008 11:33 AM, ram [EMAIL PROTECTED] wrote:
On Jan 22, 2008 9:36 AM, Bhrugu Mehta [EMAIL PROTECTED] wrote:
hi, all
I set up asterisk with 5 to 6 agent . in these all are going well. but
when
hi, all
I set up asterisk with 5 to 6 agent . in these all are going well. but
when i increase agent 12 to 13 asterisk crashed. Any suggetion.
thnks
Bhrugu mehta
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hi, all,
can anybody tell me how to be a part of asterisk developer team.
I am so much intersted.
thnks in advance.
Bhrugu Mehta
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hi, all
I am using asterisk 1.2.12.1 and zaptel 1.2.7 and libpri 1.2.1 version.
I have created Ivrs(very big) .It works fine in sip phone , but when i call
through zaptel digit sens problem occured. Asterisk doesn't sens any digit
pressed.Our pstn is CORAL pbx.
any suggesion..
thnks,
Bhrugu Mehta
hi, all
I am new to zaptel programming.
can anybody help me how to start this. or any ref. site or matirial availabel.
i want to use c lang. for this.
thnks in advance.
Bhrugu Mehta
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hi, all
i want to create cd-rom with asterisk. how it possible.
when i put disk in cdrom it boot automatifcally and auto-start
installation like TRIXBOX.
any idea.
thnks,
Bhrugu Mehta
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created.
enjoy
Bhrugu Mehta
On Jan 2, 2008 3:59 PM, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi,
Is it possible to let asterisk auto dial out and play the IVR? How?
i.e.
-asterisk auto dial out (use outgoing folder?)
-user pick the call
-play IVR (1-for English, 2-for Chinese
suggestion??
Bhrugu mehta
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hi,
first of chek you have permission on appropriate folder.
/etc/asterisk: directory not created automatically.
type command in asterisk source /usr/src/asterisk dir prompt
# make samples
this creates .conf files in /etc/asterisk dir.
enjoy!!
Bhrugu mehta
On Dec 29, 2007 2:49 PM, broadband
hi,
thnks 4 reply,
actully i am using asterisk 1.4.15 and that is defined in menuselect
file.(xml file)
so no need to add entry in module.conf
Bhrugu mehta
On Dec 27, 2007 7:37 PM, dave cantera [EMAIL PROTECTED] wrote:
bhrugu,
did you try and load it manually?
Modules are compiled
Bhrugu Mehta
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hi, all
I want to transfer my zap incoming call to another hard phone.
is there any way to transfer call.
our company is using CORAL EPBX.
thnks for any suggestion
Bhrugu Mehta
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hi, all
actually i can't understand what is the use of autoservice.c file.
can anybody help me.
thnks in advance.
Bhrugu mehta
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no , not at all, there is no need to install sound card in asteirsk system.
I am using asterisk server without soundcard.
so there may be antoher problem may in configurtion of zapata or other.
cheers!!!
Bhrugu mehta
On Dec 3, 2007 11:31 PM, Stefan Guenther [EMAIL PROTECTED] wrote:
Hi,
I
hi, all
proble:
I have add CALL-LIMIT field in my sip table in mysql.
but when i call using sip same error occurred when use simple sip.conf file.
is this possible to add CALL-LIMIT field in sip realtime table in mysql.
if yes than how
Bhrugu Mehta
hi,
ya, there is one s/w whiche is freely available for linux os as *
events.tar * .
it is in php. you can use this.
regards,
Bhrugu Mehta
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hi, all
I am new to use DeadAgi,
can anybody help me how to use DeadAgi,
actually i want this,
when caller hangup his/her phone, i want to send packet to my other app that
check caller hung up done.
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hi, all
I want to connect asterisk with oracle database.
how to start this , that's i dont know .
any pls help me
thnks in advance
Bhrugu mehta
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thnsk for giving me reply,
Bhrugu mehta
On Dec 3, 2007 12:41 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Monday 03 December 2007 00:48:55 Bhrugu Mehta wrote:
I want to connect asterisk with oracle database.
You'll need to install the Oracle ODBC driver for Linux. One word of warning
hi,
thnks for reply
I have already upgrade my odbc connector , but same error come.
Bhrugu Mehta
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hi,
I want to create connection using odbc for mysql
i have used cdr_odbc module for that.
but when asterisk insert record to my mysql database arise segfault error.
any suggetion, pls give me
tnks
Bhrugu Mehta
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Hi all
I want to create Ivrs using dialplan and aslo want to transfer call to
agent using Queue app in asterisk.
Is there any way to get IP ADDRESS of free agent which is found by asterisk
thnks ,
Bhrugu mehta
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Hi,
Various site available for asterisk,listed below,
www.asterisk.org
www.voip-info.com
www.digium.com
and best is
search in www.google.com
On Nov 5, 2007 5:22 AM, Michael Davidson [EMAIL PROTECTED] wrote:
Hi,
I'am comparative newbie to the world of Asterisk. I'd like to
introduce an
hi,all
I want make Autodialer in c++ using Asterisk Mangager Interfase;
how to syncronize originate action i.e. at a time one call made and
this time asterisk wait for some second to generate new call.
thnks in advance.
Bhrugu Mehta (SAI INFO SYSTEM
analog phone to launch ivrs welcome-ivrs.wav file plays and
when i press digit 1 play wecome and 2 play goodby.
but some time asterisk server doesn't sense which digit i pressed so
welcome-ivrs file continue with playing. Doesn't stop playing
thnks, regard
Bhrugu Mehta(SIS
Hi,
1. If you are connecting to remotly with asterisk server you have to use
asterisk -vvvrc
2. if your asterisk server is your pc then you have to use asterisk
-c
ok
enjoy
Bhrugu mehta
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HI,
I have read your mail. I get ready for that but pls tell me what i do
at remote support.
thnks for sending mail.
Bhrugu mehta
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To make call to X-lite or any sip phone ,
1. create extension in sip.conf for soft phone.
2. register sip phone with that exentension which is in sip.conf
-give your asterisk server ip in softphone and also username and password.
3. wait for some time
4. if all are good then your phone has been
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