RE: [asterisk-users] VoiceMail Access
Am Montag, den 21.05.2007, 23:16 -0500 schrieb Mike Hammett: If it is easy, could you enlighten me? I have another thread on caller ID matching, but I haven't received any positive responses. In the context where your internal calls usually are handled, like this (my internal phones have SIP accounts like sip501 with a number between 501 and 599): exten = _5XX,1,Dial(SIP/sip${EXTEN}) insert lines beforehand that check the caller id against the extension: exten = _5XX,1,GotoIf($[0${EXTEN}=0${CALLERID(num)}?voicemail,1) exten = _5XX,2,Dial(SIP/sip${EXTEN}) exten = voicemail,1,VoiceMailMain(${CALLERID(num)},s) If you want them to have to enter their voicemail password although calling from their own phone, remove the s (IIRC) - there are docs on the voip-info.org wiki about all this. Make sure that noone can fake a callerid when coming into that context... the nice thing about it having it like this is that users with a softphone can call voicemail instead of a number. About the key to be pressed when calling one's own voicebox from abroad: You can use the voicemail.conf settings exitcontext and operator for this. I do not currently, so caveat emptor: **voicemail.conf exitcontext=voicemailout operator=yes **extensions.conf [voicemailout] exten = a,1,VoiceMailMain() exten = o,1,VoiceMailMain() This way the users should be redirected to the voicemail login prompt when they press * or 0 during the message (Again: beware, I did not test this). They will have to enter the voicebox number and pin. I do not know wether there is a method to get the voicebox number at this point, such that only the pin needs to be entered. Perhaps setting a variable (before calling voicemail(123) happens) would do the trick, but I do not know wether that variable will still exists when jumping to that voicemailout context. BR Anselm ___ --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 Access
I was looking at the ILECs' web sites to determine how their users access voicemail. I looked at ATT, Verizon, Qwest, and Embarq. They supported one or a combination of the following for calling from your phone: *98 #55 Toll free number Your number A varying phone number, based on your number's location. Calling from anywhere else, they supported: Hitting star when you hear your greeting when calling yourself Toll free number What method should I use for my users checking their voicemail? Can Asterisk voicemail be made to accept hitting * during the greeting to enter the voicemail system? If they call their own number, how do I get Asterisk to recognize that and take them to the voicemail system? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.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] VoiceMail Access
I created a *9 extension which executes VoiceMailMain with the callerid number as the argument. Then of course the voicemail box just has to be the same as the phone number. Then we just have another DID for outside access. * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * Mike Hammett wrote: I was looking at the ILECs' web sites to determine how their users access voicemail. I looked at ATT, Verizon, Qwest, and Embarq. They supported one or a combination of the following for calling from your phone: *98 #55 Toll free number Your number A varying phone number, based on your number's location. Calling from anywhere else, they supported: Hitting star when you hear your greeting when calling yourself Toll free number What method should I use for my users checking their voicemail? Can Asterisk voicemail be made to accept hitting * during the greeting to enter the voicemail system? If they call their own number, how do I get Asterisk to recognize that and take them to the voicemail system? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.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
Re: [asterisk-users] VoiceMail Access
Mike Hammett wrote: I was looking at the ILECs’ web sites to determine how their users access voicemail. What method should I use for my users checking their voicemail? Can Asterisk voicemail be made to accept hitting * during the greeting to enter the voicemail system? If they call their own number, how do I get Asterisk to recognize that and take them to the voicemail system? Mike, A common approach is to use the caller id in combination with some digit sequence. For my systems, I've just used 555 as the VM extension. exten=555,1,VoicemailMain(${CALLERID(num)}) For access to the VM from outside the system, I've used an AGI script to query a database to validate the user. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail Access
On Mon, May 21, 2007 at 03:10:23PM -0400, Lee Jenkins wrote: Mike Hammett wrote: I was looking at the ILECs? web sites to determine how their users access voicemail. What method should I use for my users checking their voicemail? Can Asterisk voicemail be made to accept hitting * during the greeting to enter the voicemail system? If they call their own number, how do I get Asterisk to recognize that and take them to the voicemail system? A common approach is to use the caller id in combination with some digit sequence. For my systems, I've just used 555 as the VM extension. exten=555,1,VoicemailMain(${CALLERID(num)}) For access to the VM from outside the system, I've used an AGI script to query a database to validate the user. It's also quite easy to set-up if you call your own extension number from your extension it goes into voicemail for you extension. You can have another number as above to access voicemail from another extension. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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] VoiceMail Access
If it is easy, could you enlighten me? I have another thread on caller ID matching, but I haven't received any positive responses. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Monday, May 21, 2007 5:55 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoiceMail Access On Mon, May 21, 2007 at 03:10:23PM -0400, Lee Jenkins wrote: Mike Hammett wrote: I was looking at the ILECs' web sites to determine how their users access voicemail. What method should I use for my users checking their voicemail? Can Asterisk voicemail be made to accept hitting * during the greeting to enter the voicemail system? If they call their own number, how do I get Asterisk to recognize that and take them to the voicemail system? A common approach is to use the caller id in combination with some digit sequence. For my systems, I've just used 555 as the VM extension. exten=555,1,VoicemailMain(${CALLERID(num)}) For access to the VM from outside the system, I've used an AGI script to query a database to validate the user. It's also quite easy to set-up if you call your own extension number from your extension it goes into voicemail for you extension. You can have another number as above to access voicemail from another extension. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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
Re: [asterisk-users] voicemail access thru apache on another server
THanks RR, Am trying it right now. But getting into all kinds of trouble!! ranging from SQL Alloc failed - to seg faults. I honestly donno anything abt odbc(n i seriously dont have the time to rnd on that one, right now) Can you or anyone else paste their config files for(related to voicemail odbc storage) voicemail.conf odbc.ini odbcinst.ini res_odbc.conf res_mysql.conf (i dont think this should change , was using realtime for storing voicemail users and sip users,etc, perfectly) how abt extconfig.conf (i guess this too duznt change). all help really appreciated Ben. RR wrote: have a look at Wiki for asterisk + odbc storage. The database for storing entire voicemail messages can be stored on a local or a remote database. Then you can do whatever you want with it. You will have to recompile asterisk by turning on ODBC storage. It's all there on the Wiki ___ --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] voicemail access thru apache on another server
okk.. got it working. the problem was that I had started out with Realtime, using Mysql. Seems u can't use mysql and then put in odbc solely for voicemail storage. res_odbc.conf entry decides that u r gonna use odbc for everything. so had to replace mysql stuff with odbc in the conf files. Now, how on earth do i read the recordings and play them out thru a browser!! pheww. it never ends!!! Benjamin Jacob wrote: THanks RR, Am trying it right now. But getting into all kinds of trouble!! ranging from SQL Alloc failed - to seg faults. I honestly donno anything abt odbc(n i seriously dont have the time to rnd on that one, right now) Can you or anyone else paste their config files for(related to voicemail odbc storage) voicemail.conf odbc.ini odbcinst.ini res_odbc.conf res_mysql.conf (i dont think this should change , was using realtime for storing voicemail users and sip users,etc, perfectly) how abt extconfig.conf (i guess this too duznt change). all help really appreciated Ben. RR wrote: have a look at Wiki for asterisk + odbc storage. The database for storing entire voicemail messages can be stored on a local or a remote database. Then you can do whatever you want with it. You will have to recompile asterisk by turning on ODBC storage. It's all there on the Wiki ___ --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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail access thru apache on another server
Hi Benjamin, Am trying to build a system, wherein users can access their profiles, and hence voicemails thru a browser. I am using Apache and am running it on another box and asterisk on another. Am keeping them seperate to not have http traffic on the same box as asterisk. Now, my qs: Is there a way to tell Asterisk to store the msg.txt information in an sql database, so that it's easier to access the voice mail info?? I would wait until the IMAP support in asterisk (currently only in the svn version) is stable instead using sql-db based storage. At least if were talking about an medium installation or bigger. Also, any way to run a script or something, to move a message from INBOX to Old, when a user listens to the message thru the web browser?? Now, how on earth do i read the recordings and play them out thru a browser!! I did write a proof of concept script in php for accessing and manipulating the voicemail folder. It has no locking atm therefore there are some race conditions and its not usuable in a large scale production environment. Youre welcome to add session/locking suport though :} The script enables you to view, read, move, delete and forward voicemails using URLs. ascii based and asterisk realtime authentication is supported. You can use it at least to extract the code how to send an audiofile (if youre using php that is). Should be no problem to convert the scripts to perl or any other script language. Script: http://sip-syndication.com/index.php?option=com_remositoryItemid=26func=selectid=2 Documentation: http://sip-syndication.com/index.php?option=com_contenttask=categorysectionid=5id=21Itemid=47 cheers, Arnd ___ --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] voicemail access thru apache on another server
On Thu, Sep 14, 2006 at 06:14:52PM +0530, Benjamin Jacob wrote: Hello ppl, Am trying to build a system, wherein users can access their profiles, and hence voicemails thru a browser. I am using Apache and am running it on another box and asterisk on another. Am keeping them seperate to not have http traffic on the same box as asterisk. Just to give you another direction: mod_proxy of apache and a locla httpd on the box that runs asterisk... Now, my qs: Is there a way to tell Asterisk to store the msg.txt information in an sql database, so that it's easier to access the voice mail info?? Or use the mysql, postgresql or ODBC storage. Or use the imap storage and some web-based imap client. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [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
[asterisk-users] voicemail access thru apache on another server
Hello ppl, Am trying to build a system, wherein users can access their profiles, and hence voicemails thru a browser. I am using Apache and am running it on another box and asterisk on another. Am keeping them seperate to not have http traffic on the same box as asterisk. Now, my qs: Is there a way to tell Asterisk to store the msg.txt information in an sql database, so that it's easier to access the voice mail info?? Am planning to run another apache on the asterisk box, solely for providing file-playing access(thru hrefs on my first web-server). Also, any way to run a script or something, to move a message from INBOX to Old, when a user listens to the message thru the web browser?? Any ideas good ppl?? thanks in advance. cheerz Ben. ___ --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] voicemail access thru apache on another server
have a look at Wiki for asterisk + odbc storage. The database for storing entire voicemail messages can be stored on a local or a remote database. Then you can do whatever you want with it. You will have to recompile asterisk by turning on ODBC storage. It's all there on the Wiki ___ --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] voicemail access on the Thomson ST2030 ?
for provisioning files to be taken, you have to change the config_sn parameter each time you modify the file, otherwise the phone assumes nothing has changed.2006/5/19, Louis-David Mitterrand [EMAIL PROTECTED]: Hello,After reading all the docs and going through the menus, I still can'tfind the voicemail access button or menu sequence on the ST2030(http://www.voip-info.org/wiki/view/Thomson+ST2030 )Also I can't get phone provisionning through tftp to work. Configurationfiles are loaded but the phone seems to ignore them.Any idea?___--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] voicemail access on the Thomson ST2030 ?
Hello, After reading all the docs and going through the menus, I still can't find the voicemail access button or menu sequence on the ST2030 (http://www.voip-info.org/wiki/view/Thomson+ST2030) Also I can't get phone provisionning through tftp to work. Configuration files are loaded but the phone seems to ignore them. Any idea? ___ --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 access
Hi, my setup [pbx]---[oh323]--[asterisk] calling from the pbx into the voicemail gives following outputin the console -- Executing VoiceMailMain("OH323/R1909", "") in new stackApr 5 19:05:46 DEBUG[11862037]: res_adsi.c:212 __adsi_transmit_messages: No ADSI CPE detected (0) -- Playing 'vm-login' (language 'en')Apr 5 19:05:51 WARNING[11862037]: app_voicemail.c:3238 vm_execmain: Couldn't read username you can see that Asterisk plays the vm-login but the calling party (from pbx) doesn't hear anything. What is the message *NO ADSI CPE detected*? thx in advance... Do you Yahoo!? Better first dates. More second dates. Yahoo! Personals ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users