Re: [asterisk-users] Asterisk GUI
GUI bad! CLI good! PaulH On Thu, 2007-06-14 at 11:40 -0700, bilal ghayyad wrote: Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Regards Bilal Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ --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] SOLVED -- Accessing Voicemail(Asterisk) -- Remotely either from Cell or Landline
Ya, its not to fancy way but just the easy way which was hidden from my
thoughts/
To access Vm remotely, dial in the number the punch in ur ext at the IVR
greet, the it rings ur ext, then when u get ur VM greet sorry xyz not
available etc.. punch in *followed by VM password, thats it
--
Deepak
Deepak Naidu [EMAIL PROTECTED] wrote:
Hi, I was wondering if we can check the voicemails remotely from a cell or a
landline number.
We have SIP 3 Digit Extensions connected to Asterisk server.
If users are away from Desk need to access voicemails can they dial in to
Asterisk PBX check their messages.
I know one can check through web link even have mailed. Aslo I have checked
regarding DISA, but I am not kind of OK in using DISA now for just voicemails.
Is their any other ways. I am using Free PBX so can I do any thing from
FreePBX to manager it, if not backend configs are fine.
--
Deepak
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Re: [asterisk-users] Asterisk GUI
Paul Hales wrote: GUI bad! CLI good! PaulH Really...? So explain why every major PBX manufacturer has GUI of some sort? Surely they would have had CLI only if GUI is bad!!! Senad www.bicomsystems.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
Re: [asterisk-users] Record CDR in a Oracle database
On 12 Jun 2007, at 20:04, Everton Goularth wrote:
Hello All,
How can I do to record my asterisk's CDR in a Oracle database?
I have to use unixODBC?
Can anybody send me a step to step to do this configuration?
Thank's All
Everton Goularth
I've just been doing some work on getting realtime to talk to a
remote oracle XE (not cdr though)
Here's what I needed:
1) oracle running somewhere (I used XE - its free - as in lunch :-) )
2) unix odbc installed (I used unixODBC-2.2.11-x86-linux-rh9.tar.gz )
3) asterisk built for odbc
4) oracle 'instantclient' for linux (oracle-instantclient-
basic-10.2.0.3-1.i386.rpm
downloadable from oracle.com)
5) a working sqlnet connection between my asterisk and oracle boxes
I used sqlplus to test it. (oracle-instantclient-
sqlplus-10.2.0.3-1.i386.rpm from oracle.com)
Make sure you configure a tnsnames.ora somewhere.
6) odbc driver for oracle - (I used instantclient-odbc-
linux32-10.2.0.3-20061115.zip from oracle.com)
Make sure isql will talk to your oracle. I needed to set the
following environment vars:
ORACLE_SID=XE
TNS_ADMIN=/home/thp
PATH=/usr/local/bin:/usr/lib/oracle/client/10.2.0.3/bin:${PATH}
LD_LIBRARY_PATH=/usr/local/lib/:/usr/lib/oracle/10.2.0.3/client/lib/:/
home/thp/instantclient_10_2
And I had to edit /usr/local/etc/odbc.ini to set my Oracle SID
ServerName = XE
7) now configure asterisk to use odbc (follow the advice in www.voip-
info.org)
8) create suitable users/tables in oracle (follow the advice in
www.voip-info.org) remembering to
use VARCHAR2 instead of varchar
9) make sure that the environment vars above are set, then start
asterisk.
use netstat on your oracle system to check for an active connection
to 1521 from your asterisk.
- you should be good to go.
I've only used it for realtime iaxusers not for anything else yet. I
also have no clue how robust it is,
but I'll no-doubt find out soon !
Good luck..
Tim.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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[asterisk-users] Chanspy
I am developing a Call Center with AMI, I have already redirect and create a call, but I want to implement a functionality so I can supervise the call (spy). I have been studying the function chanspy of asterisk. Does any one know how to execute the function Chanspy(SIPXXX) throw AMI? Best Regards Carlos ___ --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] Chanspy
On 6/16/07, Carlos Garcia Mujica [EMAIL PROTECTED] wrote: I am developing a Call Center with AMI, I have already redirect and create a call, but I want to implement a functionality so I can supervise the call (spy). I have been studying the function chanspy of asterisk. Does any one know how to execute the function Chanspy(SIPXXX) throw AMI? You can do manager Originate, and send that call to context where you have ChanSpy. Regards, Atis ___ --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] ivr testing script
Hello All, I want to run a bash script on my asterisk-now machine that will test our IVR at another site. I need it to dial the number and, if it answers, listen for a few seconds and then hang up. If there are return codes from the commands that can help me determine the status of the line, I can then send an email to alert if there was busy tone or no answer. I've fiddled with a bash script to send sip show peers and parse it, but that's about the most of my experience interfacing to asterisk. Is there a way to do what I want? Thanks, Ben W ___ --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] (no subject)
Hi,
I am facing some issues while using MixMonitor and StopMonitor. My
extensions logic is attached below:
exten = s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b)
exten = s,2,Dial(SIP/101,13)
exten = s,3,StopMonitor()
exten = s,4,NoOp(Dial Status: ${DIALSTATUS})
exten = s,5,Goto(sss-${DIALSTATUS},1)
exten = sss-NOANSWER,1,VoiceMail([EMAIL PROTECTED])
exten = sss-NOANSWER,2,Goto(salesivr,s,4)
As evident from the dialplan I only want to record the call when
Dial(SIP/101,13) is successful.
After that I disable recording by issuing the StopMonitor command. Now
the problem is that when the status of dial is NOANSWER the voicemail
recording is also recorded and saved.
It is only after I hangup that I see the following print on the console
End MixMonitor Recording SIP/192.168.0.10.172-081c67c0
I want monitor to be disabled on priority s,3. Can someone please
point out what I am doing wrong here.
Regards,
Asif
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[asterisk-users] MixMonitor Problem
Hi,
I am facing some issues while using MixMonitor and StopMonitor. My
extensions logic is attached below:
exten = s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b)
exten = s,2,Dial(SIP/101,13)
exten = s,3,StopMonitor()
exten = s,4,NoOp(Dial Status: ${DIALSTATUS})
exten = s,5,Goto(sss-${DIALSTATUS},1)
exten = sss-NOANSWER,1,VoiceMail([EMAIL PROTECTED])
exten = sss-NOANSWER,2,Goto(salesivr,s,4)
As evident from the dialplan I only want to record the call when
Dial(SIP/101,13) is successful.
After that I disable recording by issuing the StopMonitor command. Now
the problem is that when the status of dial is NOANSWER the voicemail
recording is also recorded and saved.
It is only after I hangup that I see the following print on the console
End MixMonitor Recording SIP/192.168.0.10.172-081c67c0
I want monitor to be disabled on priority s,3. Can someone please
point out what I am doing wrong here.
Regards,
Asif
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Re: [asterisk-users] Asterisk GUI
On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote: Paul Hales wrote: GUI bad! CLI good! PaulH Really...? So explain why every major PBX manufacturer has GUI of some sort? Surely they would have had CLI only if GUI is bad!!! Senad Senad - it is really to cover the inability of 'average' people to understand CLI. Any 4 year old can run a GUI and that is why the skill level of people programming phone systems has gone down hill so much. It really is a profit driven decision by management to get cheaper employees and make more money. Brett ___ --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] ipv6 on Asterisk
Hi guys, Does anybody try to install IPV6 support on asterisk?I just found a patch for that but it is released on 2005,I have no idea if there is new version to support ipv6 or new patches,please advise.Thanks a lot. ___ --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] Asterisk GUI
Brett Crapser wrote: On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote: Paul Hales wrote: GUI bad! CLI good! PaulH Really...? So explain why every major PBX manufacturer has GUI of some sort? Surely they would have had CLI only if GUI is bad!!! Senad Senad - it is really to cover the inability of 'average' people to understand CLI. CLI is useful for small/simple dial tone installations. Anything above that even very competent administrator will make syntax/logical errors. GUIs do not make such mistakes and in addition do allow TRAINED average person to make changes by them selves. Any 4 year old can run a GUI and that is why the skill level of people programming phone systems has gone down hill so much. I have never heard of 4 year old been allowed to play with any companies phone systems !!! Remember no dial tone, no customers. As for skill level, I agree a lot training and patience needs to be invested into end users/resellers using GUI administration let alone CLI. Senad ___ --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] can ENUMLOOKUP query multiple DNS servers without having to replicate the same code for each server?
[EMAIL PROTECTED] wrote:
Hi all,
Does ENUMLOOKUP can query multiple DNS servers without having to
replicate the same code in which the only thing replaced is the server?
the enumlookup dialplan function (as opposed to the application) never
cares about your enum.conf file. The trick is to use separate enum
domains, and test them all in your dialplan using
${ENUMLOOKUP(+${ARG1:2:},sip,c,yourdomain.local) or something in a loop.
If I use ENUMLOOKUP(${exten}), Asterisk will parse enum.conf file to
find the list of DNS servers in order of preference to be queried, but,
I pretend to use something like this: ${ENUMLOOKUP(+${ARG1:2:},sip,c)
which does not seam to care about the existence of enum.conf file! May I
force Asterisk to care about the servers I wrote in enum.conf?
To let you understand better, I wish to use just a block of code that is
able to query multiple DNS servers, instead of repeating like in the
following example the same code for each DNS server I wish to lookup for:
; Start first with e164.arpa zone:
exten = _X.,1,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c)}|counter=0)
exten = _X.,2,GotoIf($[${counter}${sipcount}]?3:6)
exten = _X.,3,Set(counter=$[${counter}+1])
exten = _X.,4,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,${counter})})
exten = _X.,5,GotoIf($[${counter}${sipcount}]?3:6)
; ...then also try e164.org:
exten = _X.,6,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c)}|counter=0)
exten = _X.,7,GotoIf($[${counter}${sipcount}]?8:11)
exten = _X.,8,Set(counter=$[${counter}+1])
exten = _X.,9,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,${counter})})
exten = _X.,10,GotoIf($[${counter}${sipcount}]?8:11)
; ...in case of no route by IP, then send out PRI:
exten = _X.,11,Dial(Zap/g1/${EXTEN})
Regards,
Ricardo.
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--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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Re: [asterisk-users] Asterisk GUI
On Sat, Jun 16, 2007 at 08:55:24PM +0100, Senad Jordanovic wrote: Brett Crapser wrote: On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote: Paul Hales wrote: GUI bad! CLI good! PaulH Really...? So explain why every major PBX manufacturer has GUI of some sort? Surely they would have had CLI only if GUI is bad!!! Senad Senad - it is really to cover the inability of 'average' people to understand CLI. CLI is useful for small/simple dial tone installations. Anything above that even very competent administrator will make syntax/logical errors. Hence automation is required. Automation does not imply GUI. Bad GUIs get in the way of automation. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Asterisk GUI
Brett, The demand for asterisk GUI's could be that the world primarily consists of four year olds (as you put it - I call them customers) and not geeks with pocket protectors and Vi skills to tame all tasks. When you realize that IP Telephony/Asterisk was restricted to such a small band of users when it was pure coding with Vi and .conf files and now with GUI's like Trixbox you have a much wider base of users experimenting and implementing. Of course that's not to say that Trixbox is the be-all and end-all..personally I think that there is a hell of a lot missing (/wrong) with the Trixbox/Fonality product and a lot that could be/should be done differently/better. but that's for another email. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brett Crapser Sent: Saturday, 16 June 2007 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk GUI On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote: Paul Hales wrote: GUI bad! CLI good! PaulH Really...? So explain why every major PBX manufacturer has GUI of some sort? Surely they would have had CLI only if GUI is bad!!! Senad Senad - it is really to cover the inability of 'average' people to understand CLI. Any 4 year old can run a GUI and that is why the skill level of people programming phone systems has gone down hill so much. It really is a profit driven decision by management to get cheaper employees and make more money. Brett ___ --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] Convert or listen to .sln file
Hi, How do I listen to .sln audio file or convert it to some format that can listened to? Sox does not seems to support .sln as an input file. Thanks. - Andrew ___ --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] Where an extension really is (DUNDi woes)
Kyle Sexton wrote: On 6/15/07, *Anthony Francis* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Kyle Sexton wrote: I have two servers setup to do DUNDi lookups against each other. The scenario is that on server A, I have a wildcard match for extensions 64XX that rings to a local extension on the server. On server B I have a 6442 real extension that I would like to have ring if called. It seems that DUNDi is matching on the 64XX and not searching out to see if there is a *more* exact match than the pattern match. Is there any way to get around this? I don't think I am incorrect in saying that dundi doesn't look for externally that which it knows about locally. I think thats pretty standard of routing protocols. I was afraid of that. It just means I have to explicitly list every number in the DID range (so hundreds of extensions). I was hoping DUNDi would make the dialplan simpler. :( it will, look into the regexten for your sip accounts. Now have dundi lookup a number in that extension... Now, you might need to abstract extension numbers from sip accounts, but that only has advantages anyway... -- Kyle Sexton ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Convert or listen to .sln file
On 6/17/07, Andrew Pogrebennyk [EMAIL PROTECTED] wrote: Hi, How do I listen to .sln audio file or convert it to some format that can listened to? Sox does not seems to support .sln as an input file. Thanks. Sox defineately supports it. It just can't guess it's type from extension (you can try .slin or .raw). I have parameter -t raw -s -w in my config file for slin, but i don't remember by heart what was -s -w, so you might try variations (or read man page :). The same goes for u-law and a-law. Asterisk recognizes file extensions .ulaw and .alaw, but for sox i use -t ul and -t al. Regards, Atis ___ --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] Convert or listen to .sln file
Atis wrote: Sox defineately supports it. It just can't guess it's type from extension (you can try .slin or .raw). I have parameter -t raw -s -w in my config file for slin, but i don't remember by heart what was -s -w, so you might try variations (or read man page :). The same goes for u-law and a-law. Asterisk recognizes file extensions .ulaw and .alaw, but for sox i use -t ul and -t al. Regards, Atis It works, thank you! - Andrew ___ --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] VPN on Asterisk
Hi, Greetings to All, Im looking for some help on configuring VPN on the Asterisk PBX that I have hosted in US. Im currently in Middle East and as everyone knows some countries here has taboo to VOIP. Im not able to get phy phones registered to my PBX as they are blocking SIP and IAX2. Hence im looking for a VPN solution. For this first i need to setup VPN on my server .. Am i right? Well if anyone has experience in the whole setup how to make it run, a guide would be much appreciated with some pointer to equipment that are wel suited for the setup. Thanks in advance. Danny ___ --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
