Sistemista WebSolvingJaa
2005-Jul-15 09:07 UTC
[Asterisk-Users] [Aserisk-Users]no audio inside the net
Hi list, i've problems with my * server and the 4 phones which are
linked to it. i've 2 grandstream bt100 with the firmware upgraded to
101, a wi-fi phone (i don't know its brand) and another ip phone i
don't know its brand. with this sip.conf :
[general]
port = 5060
bindaddr = 192.168.100.229
context = default ;x changed from default to sip
localnet = 192.168.100.0/24
srvlookup = yes
allow=all
[2001] ;grandstream 2
type=friend
username=2001
secret=1945
canreinvite=yes
reinvite=yes
host=dynamic
dtmfmode=rfc2833
qualify=yes
;mailbox=2001
nat=1
allow=all
[2002] ; soft phone
type=friend
username=2002
secret=1945
canreinvite=yes
reinvite=yes
host=dynamic
dtmfmode=rfc2833
qualify=200
mailbox=2002
nat=1
allow=all
[2010]; wi-fi phone
type=friend
username=2010
secret=1945
nat=1
host=dynamic
dtmfmode=rfc2833
canreinvite=yes
reinvite=yes
qualify=200
allow=all
[2011] ; ip-phone no brand
type=friend
username=2011
secret=1945
nat=1
host=dynamic
dtmfmode=rfc2833
canreinvite=yes
reinvite=yes
qualify=yes
allow=all
[2012] ;grandstream1
type=friend
username=2012
secret=1945
nat=1
host=dynamic
dtmfmode=rfc2833
canreinvite=yes
reinvite=yes
qualify=yes
allow=all
*****************************
and with this extensions.conf file:
[general]
static=yes
writeprotect=yes
autofallthrough=yes
[globals]
CONSOLE=Console/dsp ; Console interface for demo
CONSOLE=Zap/1
CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
[dundi-e164-local]
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
switch => DUNDi/e164
[dundi-e164-lookup]
include => dundi-e164-local
include => dundi-e164-switch
[macro-dundi-e164]
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
[trunkint]
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
ignorepat => 9
include => local
include => trunkld
[local]
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
[macro-stdexten];
exten => s,1,Dial(${ARG2},20) ; Ring
the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they
press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy,
send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If
they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ;
Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If
they press *, send the user into VoicemailMain
[demo]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,SetVar(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,SetVar(TIMEOUT(response)=10) ; Set Response Timeout
to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory
message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some
instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,SetVar(LANGUAGE()=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the
congratulations
exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip) ; "Please hold
while..."
; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(u1234) ; Unless busy
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the
demo"
exten => #,n,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try
again"
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default) ; Call
the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo
site
exten => 500,n,Goto(s,6) ; Return to the start over message.
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
[default]
include => from-sip
exten => 1000,1,Dial,Zap/1|20 ; Exten 1000 dials Zap channel 1 with a
ring time option of 20 secs, which is the analog telephone plugged
into the first port of the TDM400P.
exten => 1000,2,Voicemail,u1000
exten => 1000,3,Hangup
exten => 1000,102,Voicemail,b1000
exten => 1000,103,Hangup
exten => 2000,1,Dial,Zap/2|20
exten => 2000,2,Voicemail,u2000
exten => 2000,3,Hangup
exten => 2000,102,Voicemail,b2000
exten => 2000,103,Hangup
exten => 3000,1,Dial,Zap/3|20
exten => 3000,2,Voicemail,u3000
exten => 3000,3,Hangup
exten => 3000,102,Voicemail,b3000
exten => 3000,103,Hangup
exten => _NXXXXXX,1,Dial(Zap/4/${EXTEN}|20,t)
[incoming]
exten => s,1,Wait(1)
exten => s,2,Dial(Zap/g1|20,t) ; Calls the first available channel in group 1
exten => s,3,Voicemail,u9000 ; Directs caller to unavailable voicemailbox
9000
exten => s,4,Hangup
exten => s,103,Voicemail,b9000 ; Directs caller to busy voicemailbox 9000
exten => s,104,Hangup
[sip-incoming]
exten => _.,1,Wait(1)
exten => _.,2,Playback(demo-thanks)
exten => _.,3,Hangup
[from-sip]
exten => 2010,1,Dial(SIP/2010,20)
exten => 2010,2,Voicemail(u2010)
exten => 2010,102,Voicemail(b2010)
exten => 2010,103,Hangup
exten => 2011,1,Dial(SIP/2011,20)
exten => 2011,2,Voicemail(u2011)
exten => 2011,102,Voicemail(b2011)
exten => 2011,103,Hangup
exten => 2012,1,Dial(SIP/2012,20)
exten => 2012,2,Voicemail(u2012)
exten => 2012,102,Voicemail(b2012)
exten => 2012,103,Hangup
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup
exten => 2002,1,Dial(SIP/2002,20)
exten => 2002,2,Voicemail(u2002)
exten => 2002,102,Voicemail(b2002)
exten => 2002,103,Hangup
[local]
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
include => sip ;x included sip
[sip]
exten => 55,1,VoicemailMain
exten => 2001,1,Dial(SIP/2001,20,tr)
exten => 2001,2,VoiceMail,u2001
exten => 2001,102,VoiceMail,b2001
exten => 2002,1,Dial(SIP/2002,20,tr)
exten => 2002,2,VoiceMail,u2002
exten => 2002,102,VoiceMail,b2002
exten => 2003,1,Dial(SIP/2003,20,tr)
exten => 2003,2,VoiceMail,u2003
exten => 2003,102,VoiceMail,b2003
exten => 2004,1,Dial(SIP/2004,20,tr)
exten => 2004,2,VoiceMail,u2004
exten => 2004,102,VoiceMail,b2004
exten => 2010,1,Dial(SIP/2010,20,tr)
exten => 2010,2,VoiceMail,u2010
exten => 2010,102,VoiceMail,b2010
exten => 2011,1,Dial(SIP/2011,20,tr)
exten => 2011,2,VoiceMail,u2011
exten => 2011,102,VoiceMail,b2011
exten => 2012,1,Dial(SIP/2012,20,tr)
exten => 2012,2,VoiceMail,u2012
exten => 2012,102,VoiceMail,b2012
exten => 2022,1,Dial(SIP/2022,20,tr)
exten => _1XXX,1,Dial(IAX/asterisk2:1945@192.168.1.30/${EXTEN}@local)
**********************************
from 2012 to 2011 it's all right in both ways
from 2012 to 2010 no audio from 2012
from 2012 to 2001 no audio in both ways
from 2011 to 2012 no audio in both ways
from 2011 to 2010 no audio in both ways
from 2011 to 2001 no audio in both ways
from 2001 to 2010 no audio in both ways
from 2001 to 2011 no audio in both ways
from 2001 to 2012 no audio in both ways
from 2010 to 2001 no audio in both ways
from 2010 to 2011 no audio in both ways
from 2010 to 2012 no audio in both ways
2002 can't login in the server.
so, anybody can suggest me something make this net work??
Kanuri, Seshu (Company IT)
2005-Jul-15 10:32 UTC
[Asterisk-Users] [Aserisk-Users]no audio inside the net
1) reinvite=yes is incorrect syntax? Check the info here:
http://voip-info.org/wiki-Asterisk+sip+canreinvite
You can keep canrenvite=yes, but why do you want that?
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go
directly from
; the caller to the callee. Some
devices do not
; support this (especially if one of
them is
; behind a NAT). So use canreinvite=no
2) Use nat=yes or nat=auto for correct evaluation of NAT by Asterisk.
3) qualify=yes may be used as qualify=800
Seshu
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Sistemista
WebSolvingJaa
Sent: Friday, July 15, 2005 12:08 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] [Aserisk-Users]no audio inside the net
Hi list, i've problems with my * server and the 4 phones which are
linked to it. i've 2 grandstream bt100 with the firmware upgraded to
101, a wi-fi phone (i don't know its brand) and another ip phone i don't
know its brand. with this sip.conf :
[general]
port = 5060
bindaddr = 192.168.100.229
context = default ;x changed from default to sip localnet 192.168.100.0/24
srvlookup = yes allow=all
[2001] ;grandstream 2
type=friend
username=2001
secret=1945
canreinvite=yes
reinvite=yes
host=dynamic
dtmfmode=rfc2833
qualify=yes
;mailbox=2001
nat=1
allow=all
[2002] ; soft phone
type=friend
username=2002
secret=1945
canreinvite=yes
reinvite=yes
host=dynamic
dtmfmode=rfc2833
qualify=200
mailbox=2002
nat=1
allow=all
[2010]; wi-fi phone
type=friend
username=2010
secret=1945
nat=1
host=dynamic
dtmfmode=rfc2833
canreinvite=yes
reinvite=yes
qualify=200
allow=all
[2011] ; ip-phone no brand
type=friend
username=2011
secret=1945
nat=1
host=dynamic
dtmfmode=rfc2833
canreinvite=yes
reinvite=yes
qualify=yes
allow=all
[2012] ;grandstream1
type=friend
username=2012
secret=1945
nat=1
host=dynamic
dtmfmode=rfc2833
canreinvite=yes
reinvite=yes
qualify=yes
allow=all
*****************************
and with this extensions.conf file:
[general]
static=yes
writeprotect=yes
autofallthrough=yes
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
CONSOLE=Zap/1
CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel
username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
[dundi-e164-local]
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
switch => DUNDi/e164
[dundi-e164-lookup]
include => dundi-e164-local
include => dundi-e164-switch
[macro-dundi-e164]
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
[iaxtel700]
exten =>
_91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
[trunkint]
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
ignorepat => 9
include => local
include => trunkld
[local]
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
[macro-stdexten];
exten => s,1,Dial(${ARG2},20) ; Ring
the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they
press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy,
send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If
they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ;
Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If
they press *, send the user into VoicemailMain
[demo]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,SetVar(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,SetVar(TIMEOUT(response)=10) ; Set Response Timeout
to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory
message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some
instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,SetVar(LANGUAGE()=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the
congratulations
exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip) ; "Please hold
while..."
; (but skip if channel is not
up) exten => 1234,n,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(u1234) ; Unless busy
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the
demo"
exten => #,n,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try
again"
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default) ; Call
the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo
site
exten => 500,n,Goto(s,6) ; Return to the start over
message.
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
[default]
include => from-sip
exten => 1000,1,Dial,Zap/1|20 ; Exten 1000 dials Zap channel 1 with a
ring time option of 20 secs, which is the analog telephone plugged into
the first port of the TDM400P.
exten => 1000,2,Voicemail,u1000
exten => 1000,3,Hangup
exten => 1000,102,Voicemail,b1000
exten => 1000,103,Hangup
exten => 2000,1,Dial,Zap/2|20
exten => 2000,2,Voicemail,u2000
exten => 2000,3,Hangup
exten => 2000,102,Voicemail,b2000
exten => 2000,103,Hangup
exten => 3000,1,Dial,Zap/3|20
exten => 3000,2,Voicemail,u3000
exten => 3000,3,Hangup
exten => 3000,102,Voicemail,b3000
exten => 3000,103,Hangup
exten => _NXXXXXX,1,Dial(Zap/4/${EXTEN}|20,t)
[incoming]
exten => s,1,Wait(1)
exten => s,2,Dial(Zap/g1|20,t) ; Calls the first available channel in
group 1 exten => s,3,Voicemail,u9000 ; Directs caller to unavailable
voicemailbox 9000 exten => s,4,Hangup exten => s,103,Voicemail,b9000 ;
Directs caller to busy voicemailbox 9000 exten => s,104,Hangup
[sip-incoming]
exten => _.,1,Wait(1)
exten => _.,2,Playback(demo-thanks)
exten => _.,3,Hangup
[from-sip]
exten => 2010,1,Dial(SIP/2010,20)
exten => 2010,2,Voicemail(u2010)
exten => 2010,102,Voicemail(b2010)
exten => 2010,103,Hangup
exten => 2011,1,Dial(SIP/2011,20)
exten => 2011,2,Voicemail(u2011)
exten => 2011,102,Voicemail(b2011)
exten => 2011,103,Hangup
exten => 2012,1,Dial(SIP/2012,20)
exten => 2012,2,Voicemail(u2012)
exten => 2012,102,Voicemail(b2012)
exten => 2012,103,Hangup
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup
exten => 2002,1,Dial(SIP/2002,20)
exten => 2002,2,Voicemail(u2002)
exten => 2002,102,Voicemail(b2002)
exten => 2002,103,Hangup
[local]
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
include => sip ;x included sip
[sip]
exten => 55,1,VoicemailMain
exten => 2001,1,Dial(SIP/2001,20,tr)
exten => 2001,2,VoiceMail,u2001
exten => 2001,102,VoiceMail,b2001
exten => 2002,1,Dial(SIP/2002,20,tr)
exten => 2002,2,VoiceMail,u2002
exten => 2002,102,VoiceMail,b2002
exten => 2003,1,Dial(SIP/2003,20,tr)
exten => 2003,2,VoiceMail,u2003
exten => 2003,102,VoiceMail,b2003
exten => 2004,1,Dial(SIP/2004,20,tr)
exten => 2004,2,VoiceMail,u2004
exten => 2004,102,VoiceMail,b2004
exten => 2010,1,Dial(SIP/2010,20,tr)
exten => 2010,2,VoiceMail,u2010
exten => 2010,102,VoiceMail,b2010
exten => 2011,1,Dial(SIP/2011,20,tr)
exten => 2011,2,VoiceMail,u2011
exten => 2011,102,VoiceMail,b2011
exten => 2012,1,Dial(SIP/2012,20,tr)
exten => 2012,2,VoiceMail,u2012
exten => 2012,102,VoiceMail,b2012
exten => 2022,1,Dial(SIP/2022,20,tr)
exten => _1XXX,1,Dial(IAX/asterisk2:1945@192.168.1.30/${EXTEN}@local)
**********************************
from 2012 to 2011 it's all right in both ways from 2012 to 2010 no audio
from 2012 from 2012 to 2001 no audio in both ways from 2011 to 2012 no
audio in both ways from 2011 to 2010 no audio in both ways from 2011 to
2001 no audio in both ways from 2001 to 2010 no audio in both ways from
2001 to 2011 no audio in both ways from 2001 to 2012 no audio in both
ways from 2010 to 2001 no audio in both ways from 2010 to 2011 no audio
in both ways from 2010 to 2012 no audio in both ways
2002 can't login in the server.
so, anybody can suggest me something make this net work??
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