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(${CALLERIDNUM})
-------------------------------------------------------------
;
; /etc/asterisk/voicemail.conf
;
[general]
format=wav49|gsm|wav
[johnhome]
5702 = 5702,John Coll,john
5703 = 5703,John Coll,john
----- Original Message ----- From: "John Coll" <john.coll@csoft.co.uk> To: <asterisk-users@lists.digium.com> Sent: Monday, January 05, 2004 9:07 AM Subject: [Asterisk-Users] Newbie - MWI> Sorry for the partial post a moment ago > > With help I got two phones communicating - PCMA/PCMU was the problem. > > Next stpe is to try voicemail. VM works fine, I can leave a mesage andthen> retrieve it - but no MWI on the phone and no stutter dialtone. > > I promise I've spent the requisite 4 hours + on google etc. but havereally> no further ideas. > > The setup is 2 Grandstream phones (latest firmware) and an asterisk on a > LAN. The cofig files I am using are shown below. Any suggestions would be > appreciated..... > > john > > ------------------------------------------------------------- > ; > ; liza:/etc/asterisk/sip.conf > ; > [general] > port = 5060 > bindaddr = 0.0.0.0 > externip = 10.0.1.198 > > [5702] > type=friend > host=dynamic > context=johnhome > reinvite=no > canreinvite=no > qualify=300 > callerid="John workroom #1" <5702> > mailbox=5702 > disallow=all > allow=ulaw > allow=alaw > ; dtmfmode=rfc2834 > dtmfmode=info > username=5702 ; not convinced this is needed > nat=yes > > > [5703] > ---- same as above in effect > > ------------------------------------------------------------- > ; > ; liza:/etc/asterisk/extensions.conf > ; > [general] > static=yes > writeprotect=no > ; > [globals] > CONSOLE=Console/dsp > > [johnhome] > exten => 5702,1,Dial(SIP/5702,20,Ttr) > exten => 5702,2,Voicemail(u5702) > exten => 5702,102,Voicemail(b5702) > exten => 5702,103,Hangup > > exten => 5703,1,Dial(SIP/5703,20,Ttr) > exten => 5703,2,Voicemail(u5703) > exten => 5703,102,Voicemail(b5703) > exten => 5703,103,Hangup > > exten => 88,1,VoicemailMain(${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 mailbox=5702@johnhome Paul
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 mailbox=5702@johnhome 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:asteriskuser@aktzero.com] Sent: Sunday, January 04, 2004 7:32 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Newbie - MWI <lots of snips>> > ------------------------------------------------------------- > > ; > > ; 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 mailbox=5702@johnhome > > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
hi,>Hi there, > >I stumbled on this list mostly by accident. I came across Asterisk * as a >means to help me get a better handle on my soaring telephone costs. Each >month I look at my phone bills and my stomach just turns because I can not >find any competition to Verizon which is the local anointed phone company >around here. > >Since I am a neophyte at all this I was wondering if some kind soul would >confirm/disconfirm my assumptions about this software called Asterisk *. > >1) Am I correct to assume that there is a way to dump Verizon and strictly >go VOIP in a SOHO situation? > >Yes>2) Can 1-800 numbers terminate to a VOIP assigned number? > > >yes, nufone has tollfree numbers (including 1800 numbers I beleive) for 2.9 cents incoming, and 2.9 cents outgoing nationwide.>3) With VOIP am I under the assumption that one must also purchase licenses >for such service to work. > >I am not sure, if your a bueisness SELLING voip server, you may need it, but if your just the "end user" then I am pretty sure you dont need a licence>4) Who are the companies I can purchase VOIP service from? I need numbers >in my local area code, plus I need some kind of unlimited VOIP service Asia >- mainly to Taiwan. > >Nufone.net -have to call them up/talk to JerJer in irc [irc.freenode.net #asterisk] connect.voicepulse.com I think there is a few other ones as well. Now for Asia, thats another thing, I dont know of any other voip providers with unlimited asia service..>5) Am I being unrealistic in my savings by implementing an Asterisk * PBX in >our SOHO situation. > >Alot cheaper than buying a commercial system... Join the IRC, there are (hopefully) alot of people around that can help you with some of the info you need!> >Thanks for helping this person out. > >Regards, > >Charles Alvis > > > >--- >[This E-mail scanned for viruses by Virus Hunter at itechgroup.com] > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users > > >DH