Re: [Asterisk-Users] NEWBIE: MWI on 7960
Works for me, did you do a sip debug and captured the output? On Tue, 22 Mar 2005 11:13:31 -0500, Friend, George E. <[EMAIL PROTECTED]> wrote: > > > > Situation: > > > > New Install of Asterisk > 7960 w/ SIP 7.4 Image > 7912 w/ SIP040406A > 3 Lines Defined on the 7960 (5104,3100,2100) > > > > Questions (configs are below): > > Why won't the MWI light on the Cisco? I've tried: > > mailbox=2100 > [EMAIL PROTECTED] > [EMAIL PROTECTED] > Does anything look goofy overall? J > > > > Thanks, > > > > George > > > > > > Sip.conf > > > > [2100] > > type=friend > > secret=froboz > > host=dynamic > > context=incoming-wvls > > canreinvite=no > > qualify=no > > callerid="Joe User" <2100> > > mailbox=2100 > > callgroup=1 > > pickupgroup=1 > > username=2100 > > > > -- Duplicate above for all extensions รข > > > > Voicemail.conf (names have been changed to protect the innocent) > > > > [zonemessages] > > eastern=America/New_York|'vm-received' Q 'digits/at' IMp > > central=America/Chicago|'vm-received' Q 'digits/at' IMp > > central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' M > 'hours' > > > > [default] > > include => wvlandsales-voicemail > > include => cookmaninsurance-voicemail > > include => uswf-voicemail > > > > [wvlandsales-voicemail] > > 2000 => 2000, Joe User > > 2100 => 2100, Joe User > > 2101 => 2101, Joe User > > 2102 => 2102, Joe User > > 2103 => 2103, Joe User > > > > [cookmaninsurance-voicemail] > > 3100 => 3100,Joe User > > > > [uswf-voicemail] > > 5101 => 5101, Joe User > > 5102 => 5102, Joe User > > 5103 => 5103, Joe User > > 5104 => 5104, Joe User > > 5105 => 5105, Joe User > > 5106 => 5106, Joe User > > 5107 => 5107, Joe User > > 5115 => 5115, Joe User > > > ___ > 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 > > ___ 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
RE: [Asterisk-Users] Newbie - MWI
There are no guarantees that the voicemail will be in the same context as the extension. By giving you the ability and flexibility of defining everything independently, there's not much you can't do! Remember, the context call in the sip.conf refers to the context in extensions.conf. the "johnhome" at the end of the [EMAIL PROTECTED] refers to the context in voicemail.conf. Maybe I'm missing your point, and I apologize if I am. Sean -Original Message- From: Andrew Thompson [mailto:[EMAIL PROTECTED] Sent: Sunday, January 04, 2004 7:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie - MWI > > - > > ; > > ; liza:/etc/asterisk/sip.conf > > ; > > [general] > > port = 5060 > > bindaddr = 0.0.0.0 > > externip = 10.0.1.198 > > > > [5702] > > type=friend > > host=dynamic > > context=johnhome > > reinvite=no > > canreinvite=no > > qualify=300 > > callerid="John workroom #1" <5702> > > mailbox=5702 > > disallow=all > > allow=ulaw > > allow=alaw > > ; dtmfmode=rfc2834 > > dtmfmode=info > > username=5702 ; not convinced this is needed > > nat=yes > > > > > > ; > > ; /etc/asterisk/voicemail.conf > > ; > > [general] > > format=wav49|gsm|wav > > > > [johnhome] > > 5702 = 5702,John Coll,john > > 5703 = 5703,John Coll,john > > > > John, > > You have your voicemail within the "johnhome" context, so for your sip > config, your phone entry for voicemail should be [EMAIL PROTECTED] > > Paul > Why shouldn't the mailbox definition inherit the context defined on the SIP entry? Why should we have to create each SIP/IAX/(etc) entry, define it's context, and then also define the context it's voicemail is in? [default] has no rights & privelidges that should put it above any other context, does it? Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie - MWI
> > - > > ; > > ; liza:/etc/asterisk/sip.conf > > ; > > [general] > > port = 5060 > > bindaddr = 0.0.0.0 > > externip = 10.0.1.198 > > > > [5702] > > type=friend > > host=dynamic > > context=johnhome > > reinvite=no > > canreinvite=no > > qualify=300 > > callerid="John workroom #1" <5702> > > mailbox=5702 > > disallow=all > > allow=ulaw > > allow=alaw > > ; dtmfmode=rfc2834 > > dtmfmode=info > > username=5702 ; not convinced this is needed > > nat=yes > > > > > > ; > > ; /etc/asterisk/voicemail.conf > > ; > > [general] > > format=wav49|gsm|wav > > > > [johnhome] > > 5702 = 5702,John Coll,john > > 5703 = 5703,John Coll,john > > > > John, > > You have your voicemail within the "johnhome" context, so for your sip > config, your phone entry for voicemail should be [EMAIL PROTECTED] > > Paul > Why shouldn't the mailbox definition inherit the context defined on the SIP entry? Why should we have to create each SIP/IAX/(etc) entry, define it's context, and then also define the context it's voicemail is in? [default] has no rights & privelidges that should put it above any other context, does it? Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - MWI
Thanks Paul very much! john -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Liew Sent: 04 January 2004 22:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie - MWI - Original Message - From: "John Coll" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, January 05, 2004 9:07 AM Subject: [Asterisk-Users] Newbie - MWI > Sorry for the partial post a moment ago > > With help I got two phones communicating - PCMA/PCMU was the problem. > > Next stpe is to try voicemail. VM works fine, I can leave a mesage and then > retrieve it - but no MWI on the phone and no stutter dialtone. > > I promise I've spent the requisite 4 hours + on google etc. but have really > no further ideas. > > The setup is 2 Grandstream phones (latest firmware) and an asterisk on a > LAN. The cofig files I am using are shown below. Any suggestions would be > appreciated. > > john > > - > ; > ; liza:/etc/asterisk/sip.conf > ; > [general] > port = 5060 > bindaddr = 0.0.0.0 > externip = 10.0.1.198 > > [5702] > type=friend > host=dynamic > context=johnhome > reinvite=no > canreinvite=no > qualify=300 > callerid="John workroom #1" <5702> > mailbox=5702 > disallow=all > allow=ulaw > allow=alaw > ; dtmfmode=rfc2834 > dtmfmode=info > username=5702 ; not convinced this is needed > nat=yes > > > [5703] > same as above in effect > > - > ; > ; liza:/etc/asterisk/extensions.conf > ; > [general] > static=yes > writeprotect=no > ; > [globals] > CONSOLE=Console/dsp > > [johnhome] > exten => 5702,1,Dial(SIP/5702,20,Ttr) > exten => 5702,2,Voicemail(u5702) > exten => 5702,102,Voicemail(b5702) > exten => 5702,103,Hangup > > exten => 5703,1,Dial(SIP/5703,20,Ttr) > exten => 5703,2,Voicemail(u5703) > exten => 5703,102,Voicemail(b5703) > exten => 5703,103,Hangup > > exten => 88,1,VoicemailMain(${CALLERIDNUM}) > - > ; > ; /etc/asterisk/voicemail.conf > ; > [general] > format=wav49|gsm|wav > > [johnhome] > 5702 = 5702,John Coll,john > 5703 = 5703,John Coll,john > John, You have your voicemail within the "johnhome" context, so for your sip config, your phone entry for voicemail should be [EMAIL PROTECTED] Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie - MWI
- Original Message - From: "John Coll" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, January 05, 2004 9:07 AM Subject: [Asterisk-Users] Newbie - MWI > Sorry for the partial post a moment ago > > With help I got two phones communicating - PCMA/PCMU was the problem. > > Next stpe is to try voicemail. VM works fine, I can leave a mesage and then > retrieve it - but no MWI on the phone and no stutter dialtone. > > I promise I've spent the requisite 4 hours + on google etc. but have really > no further ideas. > > The setup is 2 Grandstream phones (latest firmware) and an asterisk on a > LAN. The cofig files I am using are shown below. Any suggestions would be > appreciated. > > john > > - > ; > ; liza:/etc/asterisk/sip.conf > ; > [general] > port = 5060 > bindaddr = 0.0.0.0 > externip = 10.0.1.198 > > [5702] > type=friend > host=dynamic > context=johnhome > reinvite=no > canreinvite=no > qualify=300 > callerid="John workroom #1" <5702> > mailbox=5702 > disallow=all > allow=ulaw > allow=alaw > ; dtmfmode=rfc2834 > dtmfmode=info > username=5702 ; not convinced this is needed > nat=yes > > > [5703] > same as above in effect > > - > ; > ; liza:/etc/asterisk/extensions.conf > ; > [general] > static=yes > writeprotect=no > ; > [globals] > CONSOLE=Console/dsp > > [johnhome] > exten => 5702,1,Dial(SIP/5702,20,Ttr) > exten => 5702,2,Voicemail(u5702) > exten => 5702,102,Voicemail(b5702) > exten => 5702,103,Hangup > > exten => 5703,1,Dial(SIP/5703,20,Ttr) > exten => 5703,2,Voicemail(u5703) > exten => 5703,102,Voicemail(b5703) > exten => 5703,103,Hangup > > exten => 88,1,VoicemailMain(${CALLERIDNUM}) > - > ; > ; /etc/asterisk/voicemail.conf > ; > [general] > format=wav49|gsm|wav > > [johnhome] > 5702 = 5702,John Coll,john > 5703 = 5703,John Coll,john > John, You have your voicemail within the "johnhome" context, so for your sip config, your phone entry for voicemail should be [EMAIL PROTECTED] Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie - MWI
John Coll wrote: With help I got two phones communicating - PCMA/PCMU was the problem. Next stpe is to try voicemail. VM works fine, I can leave a mesage and then retrieve it - but no MWI on the phone and no stutter dialtone. I promise I've spent the requisite 4 hours + on google etc. but have really no further ideas. The setup is 2 Grandstream phones (latest firmware) and an asterisk on a LAN. The cofig files I am using are shown below. Any suggestions would be appreciated. ...and voicemail.conf + extensions.conf ? Only sip.conf can't help us debugging for you. Try again! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users