Would someone be so kind as to give me a few guidelines for "dial-in" access to asterisk so that the user can access voicemail, based on the below email. I think that is what is happening, the caller is leaving a message which is being saved on Asterisk buts ince the users are registered with ser they cant access the voicemail. I already have an account for each user in sip.conf,extensions.conf and voicemail.conf on Asterisk as suggested in the archives.
Thanks a million, Aisling.
---- Original Message ---- From: blairs@isc.upenn.edu To: ashling.odriscoll@cit.ie Subject: Re: [Serusers] FW: SER Asterisk Voicemail Date: Mon, 14 Feb 2005 13:22:38 -0500
If the message is only sent as an email attachment (delete=yes,attach=yes) then the user must listen to it by playing the attached wav file on their pc.
If the message is saved on the Asterisk server then you need to provide "dial-in" access to Asterisk that sends the caller to VoiceMailMain. From there they can access their mailbox and manage messages.
_Steve
Aisling O'Driscoll wrote:
Any more ideas on my below mail? If a user is registered with SER
and
leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI
method??
Thnaksm Aisling.
---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW: SER Asterisk Voicemail Date: Thu, 10 Feb 2005 16:45:53 -0000
Hi all,
I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message.
Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java Rockx using sipsak for sending mwi sip notify messages to the phone but is there a simpler way which I am blindly ignoring??
Thank you in advance, Aisling.
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ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001fax: 215-898-9348
sip:blairs@upenn.edu
-------------------Legal Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.
-------------------Legal Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.
Aisling:
You need a phone number that "connects" to your Asterisk server. This will become the "lead" number for voicemail. One way is to purchase an analog line card and plug it into an existing phone jack. The other way is to dedicate a number which is sent from the PSTN to your VoIP environment via a gateway. The latter is how I do it.
In either case when someone calls this "lead" number the call needs to get to Asterisk. I'll assume you have this much or can build this much based on your environment.
The next thing you will need is an entry in the extensions.conf file that sends calls to this lead number to voicemailmain. You may also need special treatment in sip.conf depending upon your environment. Here is what I do in extensions.conf to provide a menu of choice for people calling the "lead" number.
exten => 68000,1,Goto(pennmainmenu,s,1)
[pennmainmenu] exten => s,1,Ringing ; 2 seconds of ringback exten => s,2,Answer exten => s,3,Playback(upenn/welcome-2-penn) ; Play general welcome exten => s,4,Playback(upenn/welcome-N-listen) ; Play please listen menu options may have changed exten => s,5,Wait,1 ; Wait a second, just for fun exten => s,6,Background(upenn/press1) ; Play press 1 to access voicemail system exten => s,7,Background(upenn/press2) ; Play press 2 for local weather report exten => s,8,Background(upenn/press3) ; Play press 3 for local date and time exten => s,9,Background(upenn/press4) ; Play press 4 to access the Penn Zoo exten => s,10,Background(upenn/press5) ; Play press 5 for echo test exten => s,11,Background(upenn/press6) ; Play press 6, ask for pin, instruct, record, playack. ; ; Menu option 1 ; exten => 1,1,VoicemailMain ; Goto Voice Mail Main menu exten => 1,2,Hangup ; Hang them up.
Pressing one gets the caller to the exten => 1,1,VoicemailMain statement which is what gets them to the voicemail maim menu.
-Steve
Aisling O'Driscoll wrote:
Would someone be so kind as to give me a few guidelines for "dial-in" access to asterisk so that the user can access voicemail, based on the below email. I think that is what is happening, the caller is leaving a message which is being saved on Asterisk buts ince the users are registered with ser they cant access the voicemail. I already have an account for each user in sip.conf,extensions.conf and voicemail.conf on Asterisk as suggested in the archives.
Thanks a million, Aisling.
---- Original Message ---- From: blairs@isc.upenn.edu To: ashling.odriscoll@cit.ie Subject: Re: [Serusers] FW: SER Asterisk Voicemail Date: Mon, 14 Feb 2005 13:22:38 -0500
If the message is only sent as an email attachment (delete=yes,attach=yes) then the user must listen to it by playing the attached wav file on their pc.
If the message is saved on the Asterisk server then you need to provide "dial-in" access to Asterisk that sends the caller to VoiceMailMain. From there they can access their mailbox and manage messages.
_Steve
Aisling O'Driscoll wrote:
Any more ideas on my below mail? If a user is registered with SER
and
leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI
method??
Thnaksm Aisling.
---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW: SER Asterisk Voicemail Date: Thu, 10 Feb 2005 16:45:53 -0000
Hi all,
I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message.
Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java Rockx using sipsak for sending mwi sip notify messages to the phone but is there a simpler way which I am blindly ignoring??
Thank you in advance, Aisling.
-------------------Legal
Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended
only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
--
ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001fax: 215-898-9348
sip:blairs@upenn.edu
-------------------Legal Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.
-------------------Legal Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.
-
Aisling O'Driscoll -
Steve Blair