[asterisk-users] SayDecimal Number
Hi there, I am wondering is there a preset command to saydecimal number? Currently if you put comand in dialplan as SayNumber(1234) it will repeat to you. But how about if the number is decimal like 12.34. Is there any command? Thanks -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Is asterisk able to integrate with MS SQL
Yes, I have done it. I am able to connect using odbc. Now able to write to
ms sql and also retrieve in db. Now my next steps is I need to write an app
which takes a phone call, asks for the user to input a number and then
queries a MS SQL db and reads the results a row at a time back to the
caller.
anyway got example or how to go about this? I am really refresh in
programming. thanks in advance!
On 11/15/06, Wes Baehr [EMAIL PROTECTED] wrote:
Func_odbc (which is new in 1.4) was backported to 1.2. See
http://www.asterisk.org/func_odbc
While it only will return one row (there are patches to make it return
multiple rows), it's very useful for our purposes. You set up the function
in func_odbc.conf, call it with ${ODBC_FunctionName(arg1,arg2,…)} and it
executes and returns the specified data.
--
Wes Baehr
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Bruce Reeves
*Sent:* Wednesday, November 15, 2006 7:56 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Re: Is asterisk able to integrate with MS
SQL
I have an IVR for employees to enter certain information, like employee
number and such and then I pass that to a simple agi/php script that build
the query string and uses freetds. It took me a while to get it working and
reproduce it on several systems, but I am rather new to Linux in general.
On 11/15/06, *Tony Mountifield* [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Sharon Lim [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
-=-=-=-=-=-
Thanks, will do more research on that part. By the way, Im trying to do
IVR
where caller enter the pin the retrieve some information out of the MS
SQL.
I am wondering, what is the constraints or how to go about it. As per
said
MS SQL is about CDR. Now like i want to match and retrieve data out of
the
DB through IVR. Any guidance?
I don't think there is any direct access to MS SQL via FreeTDS from the
dialplan, but there are ODBC functions you could use. See this page:
http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc
Alternatively, implement your IVR using AGI or the ExternalIVR application
and then you can do what you like with the database.
See http://www.voip-info.org/wiki-Asterisk+AGI
and http://www.voip-info.org/wiki-Asterisk+cmd+ExternalIVR
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Nortex Networks
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*Good memories are to be folded neatly and tucked away into the back pocket
*
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[asterisk-users] Is asterisk able to integrate with MS SQL
Hi there, I am looking around, is there anyone did any integration asterisk talk to / connect to MS SQL? Thanks-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Is asterisk able to integrate with MS SQL
Thanks, will do more research on that part. By the way, Im trying to do IVR where caller enter the pin the retrieve some information out of the MS SQL. I am wondering, what is the constraints or how to go about it. As per said MS SQL is about CDR. Now like i want to match and retrieve data out of the DB through IVR. Any guidance? Thanks. On 11/14/06, Vicky [EMAIL PROTECTED] wrote: oops sorry i thought its my sql didnt notice it's MS SQL :D On 14/11/06, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED] ,Sharon Lim [EMAIL PROTECTED] wrote: Hi there, I am looking around, is there anyone did any integration asterisk talk to / connect to MS SQL?Look for the package FreeTDS and install it. Then build Asterisk and it will include the TDS driver that can log CDRs to MS SQL.Alternatively, install ODBC drivers for MS SQL and then use Asterisk's ODBCfunctions.CheersTony--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED] - http://tony.mountifield.org ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Questions . . .
Maybe you should try this http://www.digium.com/en/products/hardware/aadk.php . Is very heavy loaded if 9PCI cards at a server. But is possible but not encourge. Maybe you can consider to have digital extension with IP phone. THis is my opinion. :-) good luck On 11/14/06, Jason Flatt [EMAIL PROTECTED] wrote: Hello all.My company currently has an older Executone PBX system that we are outgrowing.Rather than wait until the last minute to make a hasty decision, I thought itwould be a good idea to do some research and compare options first.My expertise is in computers and networking, and telephony systems are mostlyforeign to me.What we currently have are 5 incoming POTS lines and 25 stations and arewanting to add 1 or 2 more stations.I think we might have added at least one more incoming line, except that the phones we have only support 5 lines(so I'm told).Our PBX system has room for 5 more stations, then it's timeto buy a new one.I'm assuming I need to add some hardware in order to make Asterisk work with our existing setup, but I'm not entirely sure what.Based on the readingI've done so far and my limited understanding, if we wanted to use it inplace of our existing PBX system, I would need to get an analog interface card (several, actually), like Digium's TDM400P, like so:2 - Wildcard TDM04B cards for FXO and7 - Wildcard TDM40B cards for FXS-or-1 - Wildcard TDM04B card for FXO and1 - Wildcard TDM22B card for FXO FXS and 7 - Wildcard TDM40B cards for FXSI might as well use the top configuration for future expansion.If I am correct, that is 9 PCI cards in a PC.I don't know of any motherboardthat supports that many cards, so either I'm wrong, or I'll need different cards, or I'll need to utilize 2 or more PCs in conjunction with each other.I haven't yet found any mention on the last two options, so I'm assuming I'mwrong and I need a little enlightenment.Thank you for any information that will help me better understand this. --Jason FlattFather of Six:http://www.flattfamily.com/ (Joseph, 13; Cramer, 11; Travis,9; Angela; Harry, 5; and William, 12:04 am, 12-29-2005)Linux User: http://www.sourcemage.org/Drupal Fanatic: http://drupal.org/___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [EMAIL PROTECTED] problems
On http://70.89.124.237/ click on the Asterisk Management Portal with the username : maint and password : password. Then you will be login to Asterisk Management Portal which it new name is Freepbx. Click on the Setup button on top then you will be in freepbx applications. Maybe you can try the latest version of [EMAIL PROTECTED] which is now change to www.trixbox.orgGood luck!On 10/10/06, Alex Robar [EMAIL PROTECTED] wrote: FreePBX is the successor to the Asterisk Management Portal, a part of [EMAIL PROTECTED] I don't know what version that is, so you might not even have the module I'm talking about. From the command shell, type asterisk -r, which will drop you to another prompt. From there, type sip show peers and paste the output. AlexOn 10/9/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Alex...I do not have FreePBX. What I have is this: http://70.89.124.237/ Ed ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple asterisk same GUI
Hi there, Im wondering, is it possible to have single GUI on same DB but write to different asterisk server? Means assuming you have 3 asterisk server with same configurations. Therefore with the same DB but it write to different asterisk server conf files. where is the connection that we should focus? Thanks-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple asterisk same GUI
Hi there stephen, I have tried to login and the application seem very complete. Does all the configuration share the same DB for all the extensions and only write to different asterisk server? Cause as I understand, asterisk can be write as realtime means pointing for a DB without writing to conf files. Another one is like freepbx where you store data in db and also write to conf. Which one is this application refer to?
Can you please explain more on the backend? Do it comes with billing application as well? thanks in advance.On 9/28/06, Stephen Wingfield
[EMAIL PROTECTED] wrote:
Sharon,
pbxware.bicomsystems.com
U: [EMAIL PROTECTED]
P: pbxware
All standard.
Steve
steve {at] bicomsystems [dot} com
- Original Message -
From:
Sharon Lim
To:
Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, September 28, 2006 9:52
AM
Subject: [asterisk-users] Multiple
asterisk same GUI
Hi there, Im wondering, is it possible to have single
GUI on same DB but write to different asterisk server? Means assuming you have
3 asterisk server with same configurations. Therefore with the same DB but it
write to different asterisk server conf files. where is the connection that we
should focus? Thanks-- Regards, Sharon Lim
*Good memories are to be folded neatly and tucked away into the back
pocket *
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Re: [asterisk-users] Help in Reloading of Asterisk...
if you are using asterisk realtime no need to reload. If you want to reload type command asterisk -rvvv then you type reloadOn 9/21/06, raviprakash sunkara [EMAIL PROTECTED] wrote: Hi Users,I need help or clues from U, please help me...I'm new Asterisk, I want to do the Asterisk in RealTime ConfiguringMy problem is below one . After every change to the database, the asterisk will need to be reloaded.How to Reload the Asterisk server.Now its simple , stupid doubt to put in mail-list...I don't know it. -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Uninstalling Trixbox
>From my understanding, tribox is to control asterisk via web interface. so, if u want to uninstall the tribox, i guess just delete the web folder will do then do can edit direction frm your asterisk files. On 9/20/06, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, trixbox has taken control of my asterisk system, i dont like that. i just installed trixbox for rersearch purpose now i want to uninstall it and do some research on asterisk. So plz tell me how to uninstall trixbox. will it uninstall asterisk also? -- RegardsRizwan HishamSoftware Engineer ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Uninstalling Trixbox
hehehe, then you have to edit the startup manual. I dont think so there is a way to uninstall the tribox. for example to disable start up for asterisk, i think you can try type the command chkconfig asterisk stop ..good luck On 9/20/06, Rizwan Hisham [EMAIL PROTECTED] wrote: no, im not using FreePBX, actually freepbx is a part of trixbox as is sugarCRM, FOP etc. and i also dont know about CentOS, im using RHL EE. By saying that 'It has taken control of my system', i meant asterisk. Now i dont want any web based interface to asterisk. i only want asterisk on my system. So plz help me uninstall trixbox. And Sharon, thanx for the tip, but what about the rest of the scripts trixbox has installed on my system. for example i dont want to start asterisk on system startup, but trixbox does that. so anymore help will be helpfull :)On 9/20/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Sep 20, 2006 at 01:07:11PM +0500, Rizwan Hisham wrote: Hi all, trixbox has taken control of my asterisk system, i dont like that. i just installed trixbox for rersearch purpose now i want to uninstall it and do some research on asterisk. So plz tell me how to uninstall trixbox. will it uninstall asterisk also?Trixbox has not taken over your system. It is your system. It is notjust the web interface. Trixbox is a customized CentOS distribution. It uses a number of its own packages in its own yum source (in addition tostandard CentOS packages), and has a number of non-default settings.e.g: selinux disabled.Or you may be confusing it with FreePBX? --Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- RegardsRizwan HishamSoftware Engineer ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SkypeOut with Asterisk?
I have successful link skype with asterisk with http://www.nch.com.au/skypetosip/index.html but not sure whether you need this. here is another link http://www.voip-info.org/wiki/index.php?page=Skype%20Gateways. Good luck!On 9/20/06, Devraj Mukherjee [EMAIL PROTECTED] wrote:Has anyone managed to use SkypeOut as your VoIP provider? --I never look back, it distracts from the now, Edna Mode (The Incredibles)___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to integrate freepbx with a2billing?
I have trixbox installed but dont see a2billing installed together with it...anyone have integrate this before or is there any billing system that can integrate with freepbx. thanks On 9/12/06, William Piper [EMAIL PROTECTED] wrote: Both trixbox and asterisk2billing have their own lists... you may have better luck searching there. bp On 9/11/06, Steve Totaro [EMAIL PROTECTED] wrote: Sharon Lim wrote: Hi all, I have tried to install freepbx and a2billing application. Now see both application is not integrated special on cdr part. Any idea how to integrated it?Confuse! -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *You could just install TrixBox and disable a couple things, then youwould have a working system or one you could at least look at how it isconfigured to get ideas on how to configure your machine. Thanks,Steve___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] beginners question....
http://www.voip-info.org/wiki/ here got alots of example but you need to find it. You can start with http://www.trixbox.org/ that install everything. Good luck! On 9/11/06, Panagiotis Zikos [EMAIL PROTECTED] wrote: Hi all,I am new in the asterisk company. I need to set up a small voip system for about 60 phones ( a small enterprise organization). The system must support voip calls (calls inside the enterprise) but must be able to send calls over isdn (24 channels). Thus the asterisk server must operate as SIP server and gateway at the same time.Since i am new can somebody please provide me some sample configurations for sip server and gateway, or at least send me a url for finding these samples? Thanks a lot, Panos Get your email and more, right on the new Yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to integrate freepbx with a2billing?
Hi all, I have tried to install freepbx and a2billing application. Now see both application is not integrated special on cdr part. Any idea how to integrated it?Confuse!-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TrixBox install
http://www.trixbox.org/modules/smartsection/item.php?itemid=4 On 8/28/06, Rizwan Hisham [EMAIL PROTECTED] wrote: hi guys, i need to install the .tar.gz version of trixbox. i cant find any help files for installation in it and also there is no help for it on the website. can anybody please help? Thanx in advance-- RegardsRizwan HishamSoftware Engineer ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution
I am not sure whether username can be xyz.abc cause normally is single words. try to change it. On 8/14/06, Crazy Boy
[EMAIL PROTECTED] wrote:Hi,Thank you for your response. As you said, I executed the command sip show registry. But, its not showing anything. Teliax people are also telling that my Asterisk server doesn't register with Teliax. So, the final conclusion is
My Asterisk server doesn't register with Teliax. Here I am giving my configuration files. Now, What I have to do to register my Asterisk server with Teliax?
Please tell me.SIP.CONF contents:[general] register =
xyz.abc:[EMAIL PROTECTED] [authentication] auth = xyz.abc:[EMAIL PROTECTED]
[teliax] context=teliax-incoming type=friend username=xyz.abc user=xyz.abc host=
voip-co1.teliax.com secret=xxx
insecure=very canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm[105] type=friend username=105 secret=rani callerid=Ranikumar host=dynamic
context=leader canreinvite=no nat=yes dtmfmode=rfc2833 allow=allEXTENSIONS.CONF contents:[leader] exten = 105,1,Dial(SIP/105,15)
exten = 105,2,Voicemail(u105) exten = 105,3,Voicemail(b105) exten = 105,4,Hangup exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)[teliax-incoming]
exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15)Please tell me the solution. Looking forward to your response.Thank you.Regards, Chandra.
Rich Adamson [EMAIL PROTECTED]
wrote: Thank you for your response. As you said, I changed the context default to general. Now,
1) When I am making call to our DID, its ringing. But, call is not transferring to 105 extension.No one can guess at the above without you providing something from the CLI to indicate what is going on.
2) Teliax people told me that my Asterisk server doesn't register with Teliax. But, I am making calls to US using Teliax. Without registering with Teliax, How is it possible?The register statement is only used to inform teliax that your system
is on line, can be reached at the IP address determined via the register effort, and if you have something at the end of the register statement (like /1234) teliax will send that 1234 extension in their effort to
complete a call to your asterisk system.The register statement has nothing to do with you initiating calls to them. 3) How can I know that whether my Asterisk server is registered with
Teliax or not? From the CLI, do a 'sip show registry' and it will tell you. If there is an entry shown, its registered. 4) Registering with Teliax is different for outgoing and incoming or the
same?As mentioned above, the registering is only used to inform the teliax boxes how to reach you.Your outgoing calls to teliax use the definitions you provided in the [teliax] context, just exactly like you copied them from the
teliax.com web site.
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Re: [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution
is there something wrong with ur syntax at exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr) exten = _1XX,1,DIAL(SIP/teliax,${EXTEN},30,tr)
On 8/14/06, Crazy Boy
[EMAIL PROTECTED] wrote:
Hi, My user name is : rudy.pandya Thank you.Sharon Lim
[EMAIL PROTECTED] wrote: I am not sure whether username can be xyz.abc cause normally is single words. try to change it.
On 8/14/06, Crazy Boy [EMAIL PROTECTED]
wrote:Hi,Thank you for your response. As you said, I executed the command sip show registry. But, its not showing anything. Teliax people are also telling that my Asterisk server doesn't register with Teliax. So, the final conclusion is
My Asterisk server doesn't register with
Teliax. Here I am giving my configuration files. Now, What I have to do to register my Asterisk server with Teliax? Please tell me.
SIP.CONF contents:[general] register = xyz.abc:[EMAIL PROTECTED]
[authentication] auth = xyz.abc:[EMAIL PROTECTED]
[teliax] context=teliax-incoming type=friend username=xyz.abc user=xyz.abc host=
voip-co1.teliax.com secret=xxxinsecure=very canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm[105] type=friend username=105 secret=rani
callerid=Ranikumar host=dynamic context=leader canreinvite=no nat=yes dtmfmode=rfc2833 allow=allEXTENSIONS.CONF contents:
[leader] exten = 105,1,Dial(SIP/105,15) exten = 105,2,Voicemail(u105) exten = 105,3,Voicemail(b105) exten = 105,4,Hangup exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
[teliax-incoming]exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15)Please tell me the solution. Looking forward to your response.
Thank you.Regards, Chandra.
Rich Adamson [EMAIL PROTECTED] wrote:
Thank you for your response. As you said, I changed the context default to general. Now, 1) When I am making call to our DID, its ringing. But, call is not transferring to 105 extension.
No one can guess at the above without you providing something from the CLI to indicate what is going on. 2) Teliax people told me that my Asterisk server doesn't register with Teliax. But, I am making calls to US using Teliax. Without registering
with Teliax, How is it possible?The register statement is only used to inform teliax that your system is on line, can be reached at the IP address determined via the
register effort, and if you have something at the end of the register statement (like /1234) teliax will send that 1234 extension in their effort to complete a call to your asterisk system.
The register statement has nothing to do with you initiating calls to them. 3) How can I know that whether my Asterisk server is registered with Teliax or not? From the CLI, do a 'sip show registry' and it will tell you. If there
is an entry shown, its registered. 4) Registering with Teliax is different for outgoing and incoming or the same?As mentioned above, the registering is only used to inform the teliax
boxes how to reach you.Your outgoing calls to teliax use the definitions you provided in the [teliax] context, just exactly like you copied them from the
teliax.com web
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Re: [asterisk-users] Asterisk Billing
http://www.itelbilling.com/ try this! On 8/12/06, Wasif [EMAIL PROTECTED] wrote: Hello,Does anyone knowabout open source wholesale billing for Asterisk?Thanks___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime+mysql
If i am not mistaken, you need to have another IAX user tables to store all the iax users. HEre is some example http://www.voip-info.org/wiki/view/Asterisk+RealTime+IAX
good luck!On 8/9/06, Shaun [EMAIL PROTECTED] wrote:
I'm attempting to setup asterisk running real-time with mysql.Right now Ican get asterisk to start and run but a show dialplan shows basicallynothing other than parking extensions.I'm watching the full log also for
debug messages and I can see that asterisk is connecting to mysql with out aissue but for some reason it does not seam to read the extensions table.Below is a few of my configs..#My slimmed down modules.conf
[EMAIL PROTECTED] asterisk]# cat /etc/asterisk/modules.conf[modules]autoload=noload = res_config_mysql.soload = res_crypto.soload = res_features.soload = chan_features.soload = chan_iax2.so
load = pbx_realtime.soload = app_realtime.so#My extconfig.conf[EMAIL PROTECTED] asterisk]# cat /etc/asterisk/extconfig.conf[settings]extensions = mysql,asterisk,extensions#mysqldump of my asterisk database
DROP TABLE IF EXISTS `extensions`;CREATE TABLE `extensions` (`context` varchar(20) NOT NULL default 'default',`extension` varchar(20) NOT NULL default '',`priority` int(2) NOT NULL default '1',
`application` varchar(20) NOT NULL default '',`args` varchar(50) default NULL,`descr` text,`flags` int(1) NOT NULL default '0',PRIMARY KEY(`context`,`extension`,`priority`)) ENGINE=MyISAM DEFAULT CHARSET=latin1;
Dumping data for table `extensions`--INSERT INTO `extensions` VALUES('sortcalls','1949265snip',1,'Wait','10','Wait(10)',0)INSERT INTO `extensions` VALUES('sortcalls','_1949265snip',1,'Wait','10','Wait(10)',0);
My iax.conf is setup to forward calls to sortcalls context... this is what iget when a call comes in..Aug8 22:47:58 NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejectedconnect attempt from
64.61.93.87, request '1949265snip@sortcalls' does notexistAug8 22:47:58 NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejectedconnect attempt from 64.61.93.90, request '1949265snip@sortcalls' does not
existAnybody know whats going wrong here?--~Shaun___--Bandwidth and Colocation provided by
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Re: [asterisk-users] Re: realtime+mysql
Sorry if i am wrong. Did you add something in extensions.conf to identify your context ? Something like this http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
BTW, how come your extensions got snip? I taught extension is a number that you dial?On 8/9/06, Shaun
[EMAIL PROTECTED] wrote:
IAX is being read from the flat config like it
normally is. I can verify this because asterisk registers with my
provider.
-- ~Shaun
Sharon Lim [EMAIL PROTECTED] wrote
in message news:[EMAIL PROTECTED]
...If
i am not mistaken, you need to have another IAX user tables to store all the
iax users. HEre is some example http://www.voip-info.org/wiki/view/Asterisk+RealTime+IAX
good luck!
On 8/9/06, Shaun
[EMAIL PROTECTED]
wrote:
I'm
attempting to setup asterisk running real-time with mysql.Right
now Ican get asterisk to start and run but a show dialplan shows
basicallynothing other than parking extensions.I'm watching
the full log also for debug messages and I can see that asterisk is
connecting to mysql with out aissue but for some reason it does not seam
to read the extensions table.Below is a few of my configs..#My
slimmed down modules.conf [EMAIL PROTECTED] asterisk]# cat
/etc/asterisk/modules.conf[modules]autoload=noload =
res_config_mysql.soload = res_crypto.soload =
res_features.soload = chan_features.soload = chan_iax2.so
load = pbx_realtime.soload = app_realtime.so#My
extconfig.conf[EMAIL PROTECTED] asterisk]# cat
/etc/asterisk/extconfig.conf[settings]extensions =
mysql,asterisk,extensions#mysqldump of my asterisk database
DROP TABLE IF EXISTS `extensions`;CREATE TABLE `extensions`
(`context` varchar(20) NOT NULL default
'default',`extension` varchar(20) NOT NULL default
'',`priority` int(2) NOT NULL default '1',
`application` varchar(20) NOT NULL default
'',`args` varchar(50) default NULL,`descr`
text,`flags` int(1) NOT NULL default
'0',PRIMARY
KEY(`context`,`extension`,`priority`)) ENGINE=MyISAM DEFAULT
CHARSET=latin1; Dumping data for table
`extensions`--INSERT INTO `extensions`
VALUES('sortcalls','1949265snip',1,'Wait','10','Wait(10)',0)INSERT
INTO `extensions`
VALUES('sortcalls','_1949265snip',1,'Wait','10','Wait(10)',0);
My iax.conf is setup to forward calls to sortcalls context...
this is what iget when a call comes in..Aug8 22:47:58
NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejectedconnect attempt
from 64.61.93.87, request
'1949265snip@sortcalls' does notexistAug8
22:47:58 NOTICE[10074]: chan_iax2.c:7303 socket_read: Rejectedconnect
attempt from 64.61.93.90, request
'1949265snip@sortcalls' does not existAnybody
know whats going wrong
here?--~Shaun___--Bandwidth
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Re: [asterisk-users] problem- 0:10 long message
Did you leave any message in your voicemail? Cause by default if 10 seconds silence then it will end the recording. On 8/8/06, ismir saljic [EMAIL PROTECTED] wrote:Hi , I have the problem with voicemail message duration.Every message is only 10 seconds long. just wanted to let you know you were just left a 0:10 long message... Any idea? Thanks! Yahoo! Music Unlimited - Access over 1 million songs. Try it free. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ANI agi
I am trying to do a simple agi connection to db with the guidance from http://www.voip-info.org/wiki/view/Asterisk+AGI+php Item 13 with ani.agi file, db and extensions.conf13. another sample, ANI Scenario - did callers call the Asterisks box and land on the context did, Asterisks answers the call and then execute the script ani.agi. This script will connect to mysql and see whether originators phone number(caller ID) is already in the database. If it exists, then agi.script will pass the control back to dial plan. Dial plan now will progress the call. If the caller ID does not exists, ani.agi will pass the control to the dial plan as un authenticated call. ani.agi #!/usr/local/bin/php-q ?php ob_implicit_flush(true); set_time_limit(6); $in=fopen(php://stdin,r); $stdlog=fopen(/var/log/asterisk/my_agi.log,w); i have put the value of the callerid i called frm assume then when i do debug i got the callerid in agi but seem that ani.agi is not working cause is asking for password. From my understanding, if the callerid is in the db then it will not ask to authenticate right? anything missing here? THanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail dial pattern from old pbx
I have an old pbx and I want to pass callerid frm the old pbx to asterisk as a voicemail server. My old pbx have sent the callerid but i am not sure how to make it into dialpattern cause if I have 1000 callerid then i have to enter 1000 enter into
extensions.conf. I m using tdm400p where i pull an extension(728) into asterisk box. Therefore other extension in oldpbx will divert to this extension(728)My current dialplan which is working as below :
[defaults]exten =s, 1, Answer()exten =740,1,Voicemail(${EXTEN})After answer then it will get the callerid (I do a READ cmd then it
read as s extension but after that it stated User entered '721')How to write the dial patterns. Went I put exten = XXX,1, Voicemail(${EXTEN}), it wont do to this lines but will just hangup cause cant scan the next steps to do. Thanks in advance.
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Re: [asterisk-users] Voicemail dial pattern from old pbx
Hello, Don and steven, Something like this should drop you to the voicemail box of the caller ID.
exten = s,1,Voicemail(${CALLERIDNUM})Don PobanzThe previous settings cant get any calleridnum. But this can
exten = XXX,1, Voicemail(${EXTEN}) may work, but you have to have it as _XXXThe underscore tells asterisk that this is an _expression_ and the X is any digit.XXX is sitting there waiting for someone to dial three Xs.
--StevenMuy current dial plan is exten =s, 1, Answer()
exten =_XXX,1,Voicemail(${EXTEN})there is a slient moment between ANSWER and VOICEMAIL. How do it control it too be faster? Thanks very much!
On 8/1/06, Don Pobanz [EMAIL PROTECTED] wrote:
Sharon Liam wrote: exten =s, 1, Answer() exten =740,1,Voicemail(${EXTEN}) After answer then it will get the callerid (I do a READ cmd then it read as s extension but after that it stated User entered '721')
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Re: [asterisk-users] SIP configuration by group
hmm...the group functions is to dial all the sip account, right. assuming if the dial plan is like exten = blah,1,Dial(${GROUP_A})exten = moreblah,1,Dial(${GROUP_B})then it will dial sip100 sip200 at the same time right? But i want to group it as different company. Is it possible?
Assuming, if 1 have 2 company and want to have same sip account context, how do i differentiate with it? Thanks in advance. On 7/17/06, El Flynn
[EMAIL PROTECTED] wrote:
Sharon Lim wrote: Hi there, I would like to ask, is it possible to group sip user? Means group A with sip user 100,200 and group B with sip user 100,200? thanks in advance.
in your dialplan, define the following variables:GROUP_A=SIP/100SIP/200GROUP_B=SIP/150SIP/200and in your dial stringexten = blah,1,Dial(${GROUP_A})exten = moreblah,1,Dial(${GROUP_B})
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Re: [asterisk-users] No ringing on outgoing SIP calls.
Perhaps you can check on the dtmf code. On 7/14/06, Robert La Ferla [EMAIL PROTECTED] wrote: When I dial out, I can't hear any ringing.I am using the latest SVNcode (SVN-branch-1.2-r37458M ).Is this a problem with Asterisk? Orwith my VOIP provider?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP configuration by group
Hi there, I would like to ask, is it possible to group sip user? Means group A with sip user 100,200 and group B with sip user 100,200? thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP conf
My concern on sip.conf , is user able to create their own extension . Therefore how to control the sip user account. My understand are, 1. You need to create either zap/sip/iax channel ~user account as a user. 2. Then to activate the account you need to have a extension number to ring the user account. i having problem, if I dont do checking on context in sip.conf, therefore it might have duplicate context. Like if i accidently created 2 Jenny sip accounts but refering to different context in extensions.conf. for outgoing calls is ok but for incoming call which account it will trigger and will ring which user phone. Cause if we have 2 jenny then if in extensions.conf exten=200,1,Dial (SIP/jenny) which account will ring. That's my problem. Thanks so much for the feedback.On 7/6/06, Olle E Johansson [EMAIL PROTECTED] wrote: 6 jul 2006 kl. 04.09 skrev Sharon Lim: hi, Is it possible to have same sip context but refering to different username and context? Is it a must to have username(test) and the sip context [test]the same? what the different with username sip context? can username variable be alphanumeric, like using email address as username?Please read the documentation carefully. The username setting is *not* the user name,strangely enough. You usually do not need to set that, especially notfor a user.The context has nothing to do with the username, it's a reference tothe dialplanentrypoint for this device. /O default sip as below : [test] disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm type=friend username=test secret=1234 host=dynamic dtmfmode=rfc2833 ; defaultip= 10.10.10.17 qualify=yes mailbox=2000 nat=yes context=extensions accountcode=test canreinvite=no thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users---* Olle E Johansson - [EMAIL PROTECTED]* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP conf
hi, Is it possible to have same sip context but refering to different username and context? Is it a must to have username(test) and the sip context [test] the same? what the different with username sip context? can username variable be alphanumeric, like using email address as username?default sip as below : [test] disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm type=friend username=test secret=1234 host=dynamic dtmfmode=rfc2833; defaultip= 10.10.10.17 qualify=yes mailbox=2000 nat=yes context=extensions accountcode=test canreinvite=no thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call back features
anyone have information on how the call back features work with asterisk? I means the dial plan or what so ever. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming PSTN calls not routing to Asterisk? (using Sipura 3000)
Hi John, Your first question, I am not sure why but for this part i can explain abit Also, on a side note, I have a context called [home] which each SIPPhone is associated with.Do I need to specify each extension inthere?SIP user can register as name as well . Doesnt means to have number. Example in sip.conf [john]type=friendusername=johnhost=dynamiccontext=incomingsecret=6769dtmfmode=rfc2833disallow=allallow=ulawinsecure=veryThen in extensions.conf , you can have any number to ring this john sip user phone. Example : exten =9XXX,1,Dial(SIP/john) ; any number start with 9 end with 7 digit behinds. or you can also exten = 9XXX,2,Hangupexten = s,1,Dial(SIP/john) ; starting of the incoming call will ring John phone. exten = s,2,HangupHope my explaination is clear or fullfill your needsthanksOn 6/17/06, John Klimek [EMAIL PROTECTED] wrote:Incoming calls from my Sipura 3000 don't seem to be correctly routing to Asterisk (or something?)Here is my Asterisk configuration for my incoming PSTN line:Code:[1000]type=friendhost=dynamiccontext=incomingsecret=6769dtmfmode=rfc2833disallow=all allow=ulawinsecure=veryInside of extensions.conf, I have this:Code:[incoming]exten = s,1,Answer( )exten = s,2,Background(enter-ext-of-person)When I call my PSTN line, my Sipura 3000 seems to successfully answer it because the line rings once, but then immediately switches to asecond dial tone. Shouldn't my incoming call be answered and then haveenter-ext-of-person played to them?What could be causing this? Also, on a side note, I have a context called [home] which each SIPPhone is associated with.Do I need to specify each extension inthere?For example:exten = 50,1,Dial(SIP/50)exten = 50,2,Hangup exten = 21,1,Dial(SIP/21)exten = 21,2,HangupCan't I just setup a default system where any two-digit number isassumed to be an extension and it is automatically tried?Thanks for any help!! ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold troubleshooting
Did you install the sound packages such as mpg123-0.59r-1.i386.rpm ? Can download from http://rpm.pbone.net/index.php3/stat/4/idpl/516450/com/mpg123-0.59r-1.i386.rpm.html good luck!On 6/16/06, kharris [EMAIL PROTECTED] wrote: Can anyone point me in the direction for resources for troubleshootingno MusicOnHold with Asterisk version 1.2.9.1 and Asterisk Addons version1.2.3?ThanksKarl___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to hang the zap channel
I had the same problem. I change some variable in zapata.conf such as in [defaults] context : 1. busydetect=yes2. busycount=43. hanguponpolarityswitch=yes ; some said need this variables4. rxgain= 1.0 5. txgain=1.0not sure which one effect it...but tried it...On 6/16/06, Steven Ringwald [EMAIL PROTECTED] wrote:soft hangup Zap/4-1SteveBartosz Wegrzyn - asterisk wrote: After all users disconnect the Zap channel is still connected to pSTN call. asterisk1*CLI show channels ChannelLocation State Application(Data) Zap/4-1 [EMAIL PROTECTED] UpMeetMe(500|xApMs|1234) Zap/pseudo-141305407 [EMAIL PROTECTED]:1Rsrvd (None) 2 active channels 1 active call I wish I could shutdown that channel when all users disconnect. Maybe I am doing something wrong. Maybe the extensions design is wrong. Thanks On Thu, Jun 15, 2006 at 10:37:23AM -0500, Bartosz Wegrzyn - asterisk wrote: in which extension, the thing is that when every (voip) user disconnects , the zap channel is still connected to the conference, How about a nice little show channels ? -- Tzafrir Cohensip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED]http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open source sip softphone (Window OS version )
yes. you can use xten http://www.xten.net/index.php?menu=download. free to download.On 6/15/06, Asterisk guy [EMAIL PROTECTED] wrote: are there any open source sip softphone (Window OS version )?___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] click to call features on asterisk
Firstly, thanks for the information, but I dont seem to get this SNAP work. I found out that the disadvantage of this is most computer dont come with mozilla, therefore for some non-IT literal is quite troublesome for them. Hmm..hopefully someone can provide me some info on click n call features. thanks in advance. On 6/10/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:You could check out Snap, there is a Firefox extension for it. You won't have to program webpages or anything as the phone numbers areautomatically detected and handled without needing anything extra fromthe web designer.http://www.snapanumber.com On 6/9/06, Sharon Lim [EMAIL PROTECTED] wrote: Hi colin, I am doing on php. But i would glad that you can share the codes as i will explore it. Thanks. On 6/9/06, Colin Anderson [EMAIL PROTECTED] wrote: I have, using Active Server Pages + Flash. See: http://new.landmarkmasterbuilder.com and click on Contact Call Us Online. I can post the .asp and .fla somewhere if someone is interested in it.-Original Message- From: Sharon Lim [mailto: [EMAIL PROTECTED]] Sent: Friday, June 09, 2006 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] click to call features on asterisk Hi there, anyone in the community has manage to configure click to call features? Care to share. I have tried on this manual , seem got some software error like http://www.voip-info.org/wiki/view/Asterisk+click+to+call Software error: Unable to determine call statusMessage: Originate with 'Exten' requires 'Context' and 'Priority' For help, please send mail to the webmaster ([EMAIL PROTECTED]), giving this error message and the time and date of the error. Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] conference
each ip phone need to register to be able to call the conference. Firstly, you need to create user with username and password under sip.conf, Then you need to create conference room which is meetme in meetme.conf then you need to create extension to point to the conference room in extensions.confafter that, just click on the extension that refer in conference room. for more info, read this http://www.voip-info.org/wiki-Asterisk+cmd+MeetMehope this help. On 6/13/06, Peter Bowyer [EMAIL PROTECTED] wrote:Have you sent this enough times yet?On 13/06/06, Khaled Chehab [EMAIL PROTECTED] wrote: Any one knows how to make a call conference using a voip gateway connected to asterisk. In mean what should I press (extension)to have another line and make the conference . regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Peter BowyerEmail: [EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] click to call features on asterisk
Hi there,anyone in the community has manage to configure click to call features? Care to share.I have tried on this manual , seem got some software error like http://www.voip-info.org/wiki/view/Asterisk+click+to+callSoftware error: Unable to determine call statusMessage: Originate with 'Exten' requires 'Context' and 'Priority' For help, please send mail to the webmaster ([EMAIL PROTECTED]), giving this error message and the time and date of the error. Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] click to call features on asterisk
Hi colin,I am doing on php. But i would glad that you can share the codes as i will explore it. Thanks. On 6/9/06, Colin Anderson [EMAIL PROTECTED] wrote: I have, using Active Server Pages + Flash. See: http://new.landmarkmasterbuilder.com and click on Contact Call Us Online. I can post the .asp and .fla somewhere if someone is interested in it. -Original Message-From: Sharon Lim [mailto:[EMAIL PROTECTED]]Sent: Friday, June 09, 2006 6:37 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] click to call features on asteriskHi there,anyone in the community has manage to configure click to call features? Care to share.I have tried on this manual , seem got some software error likehttp://www.voip-info.org/wiki/view/Asterisk+click+to+call Software error: Unable to determine call statusMessage: Originate with 'Exten' requires 'Context' and 'Priority' For help, please send mail to the webmaster ([EMAIL PROTECTED]), giving this error message and the time and date of the error. Thanks in advance. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reserving a conference room
Hi there, Try use Web-MeetMe http://www.voip-info.org/wiki/view/MeetMe-Web-Control . I have tried to install but havent had time to configure. The latest version has a Conference schedulling..maybe this will helpwhen you get it working maybe can email me the configuration details...thanks..have a great day! On 6/9/06, Eric Bishop [EMAIL PROTECTED] wrote: Hi all, We have executives who use conference rooms. The typical scenario is that one of them will organise a conference a few hours in advance and email everyone the details, however is there anyway the they can reserve a conference room number? For example if they organise a conference in room 123 at 4pm and send out all the details then someone else in the meantime dynamically creates conference room 123. This is what we are trying to avoid. I know we can set up a static conference room and pin, but we would like to do it with dynamic conference rooms. thanks. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about SIP or IAX2 or both for Asterisk.
Hi there, SIP more commanly used and it is a openstandard. Meanwhile IAX2 is a protocol on asterisk. I dont think it will effect the cpu resources cause they are bid with the same codecs like G711 and etc..so if you used SIP or IAX2 it also refer to the same codecs...so dont think it will take a lots of cpu resources. Correct me if i am wrong. Thanks. Have a nice day!On 6/9/06, Lan [EMAIL PROTECTED] wrote: Dear Friends and Supporters! I have successfully installed the asterisk at home, and it is working perfectly with sip and iax2 users. However, I am wondering that should we use only sip or only iax2? If we use both of them, does it take a lots of cpu resources to translate between sip with iax2 users? Your opinions will be very appreciated! Regards, Lan. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme conf
Hmm...any idea where to define the context of a conference? Cause from my understanding, [rooms] context is a default. ThanksOn 5/18/06, Gavin Henry [EMAIL PROTECTED] wrote: quote who=Sharon Lim hi there, i am wondering can meetme.conf able to support diffferent context. Cause currently, it has [rooms] context. ] is it possible to have same conference number with different context? thanksTry it and see ;-)--Kind Regards,Gavin Henry.Open Source. Open Solutions(tm).http://www.suretecsystems.com/___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme conf
hi there, i am wondering can meetme.conf able to support diffferent context. Cause currently, it has [rooms] context. ]is it possible to have same conference number with different context?thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web Admin
Are you looking for an web interface that write to asterisk config files? if yes, you can look at freepbx.org . On 5/11/06, Kerry Garrison [EMAIL PROTECTED] wrote: You could install any number of interfaces but it does not come with one. Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED] http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Francisco SalinasSent: Wednesday, May 10, 2006 10:05 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Web Admin I am planning to deploy Asterisk Business edition. Does this edition have a web module administration? Thanks ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with TDM400P and FXO modules
I am sorry cause i post this questions is not related to your problem, but i am having problem detecting my TDM400P which is a TDM400P problem. I manage to installed the card with compiling with zaptel and it got 2FXS and 2FXO. I am having problem while reboot or restart the system. Kudzu seem to detect TDM400P network card which is Tiger Jet Network Modem /ISDN hardware removed. Is quiet confusing cause if i remove the hardware configuration is still working and something the card no circuits flow. I am using centos 4.2, asterisk , zaptel, libpri 1.2.2 and runnning freepbx as interface to asterisk. When i run ztcfg -v , output as below :Zaptel Configuration== Channel map:Channel 01: FXO Kewlstart (Default) (Slaves: 01)Channel 02: FXO Kewlstart (Default) (Slaves: 02)Channel 03: FXS Kewlstart (Default) (Slaves: 03)Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured.Thanks in advance. On 5/9/06, Rich Adamson [EMAIL PROTECTED] wrote:Bogdan Tocu wrote: The outline is like this : Board 1 channels 1-4 # channels 1-4 Port 1 unused Port 2 unused Port 3 - FXO module - not working Port 4 - FXO module - working ok Board 2 - 4 FXO modules - all working ok# channels 5-8 Board 3 - 4 FXO modules - none works #channels 9-12 Any ideeas?Which part of the previous post did you not understand?You wrote the entries in /etc/zaptel.conf assuming that what you arecalling Board 1 really is Board 1, and its not. Its Board 3 using your numbering scheme.I don't know of any way to determine exactly how three identical boardsare numbered, so you'll have to experiment to determine which boardholds channels 1-4, which has channels 5-8, and which has channels 9-12. Its obvious from the error message that you posted that channel 9 and 10correspond to what you are calling Board 1 (since there are no modulesin the first two positions of that board).So, change your /etc/zaptel.conf and zapata.conf to address the emptychannel 9 and 10 slots.Once you get asterisk to run, then (and only then) you can place a callto each pstn line and see which Zap channel corresponds to which boardby watching the CLI. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 process running concurrent in dialplan
Hi there, I am new to asterisk, I am trying to write a dialplan with 2 process running concurrently. Current dialplan only able to execute process with priority example exten =100,1,Answer()exten =100,2,Musiconhold() exten =100,3,Hangupis it possible to have process musiconhold/background and dial process together.? I am looking into a dialplan where once call connected then the musiconhold or background still running. Mean i need 2 process running at a same time. Is it possible?Thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
