I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI> sip show registry
Host dnsmgr Username Refresh
State Reg.Time
sip3.voipvoip.com:5060 N 1112530146 105
Registered Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI>
Here is the dial plan:
[incoming]
exten => 17036361355,1,Playback(beep)
exten => 17036361355,2,SayDigits(${EXTEN})
exten => 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle mobile phone.
;exten => s,1,Answer()
;exten => s,n,Dial(SIP/17037171234,150,r,t,)
[general]
register =>1112530146:albany!@#123 at sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming
; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=sip3 at voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite
Thoughts please? I think something to do w/ "incoming" is
incorrect.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130408/6c5c6a5a/attachment.htm>
On Monday 08 April 2013, Thomas Perron wrote:> I am trying to make sure my DID and SIP account details are working > properly and engaging the extensions.conf and dial plan. > > I have a successful SIP session registered: > > Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922) > Asterisk*CLI> sip show registry > Host dnsmgr Username Refresh > State Reg.Time > sip3.voipvoip.com:5060 N 1112530146 105 > Registered Mon, 08 Apr 2013 06:02:09 > 1 SIP registrations. > Asterisk*CLI> > > Here is the dial plan: > [incoming] > exten => 17036361355,1,Playback(beep) > exten => 17036361355,2,SayDigits(${EXTEN}) > exten => 17036361355,3,Goto(testdtmf|s|1 > ;Ring on Elle mobile phone. > ;exten => s,1,Answer() > ;exten => s,n,Dial(SIP/17037171234,150,r,t,) > > > [general] > register =>1112530146:albany!@#123 at sip3.voipvoip.com/1112530146 > registertimeout=20 > context=incoming > allowoverlap=no > bindport=5060 > bindaddr=192.168.1.10 > srvlookup=no > ;context=incoming > > ; The SIP provider > [voipvoip.com] > canreinvite=no > username=1112530146 > fromuser=1112530146 > secret=albany!@#123 > context=incoming > type=friend > fromdomain=sip3 at voipvoip.com > host=69.90.209.57 > dtmfmode=rfc2833 > disallow=all > allow=alaw > allow=ulaw > nat=force_rport > insecure=port,invite > > Thoughts please? I think something to do w/ "incoming" is incorrect.You only have one extension, "17036361355" in the [incoming] context in your dialplan. Are you sure that "17036361355" is exactly what the SIP provider are actually sending to your end ? I'd put an "s" extension with a NoOp(${EXTEN}) in there, just to catch the actual extension number they were sending. -- AJS Answers come *after* questions.
Jacob.E.Miles at L-3Com.com
2013-Apr-08 12:08 UTC
[asterisk-users] extensions.conf / test DID
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI> sip show registry
Host dnsmgr Username Refresh
State Reg.Time
sip3.voipvoip.com:5060 N 1112530146 105
Registered Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI>
Here is the dial plan:
[incoming]
exten => 17036361355,1,Playback(beep)
exten => 17036361355,2,SayDigits(${EXTEN})
exten => 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle mobile phone.
;exten => s,1,Answer()
;exten => s,n,Dial(SIP/17037171234,150,r,t,)
[general]
register =>1112530146:albany!@#123 at sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming
; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=sip3 at voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite
Thoughts please? I think something to do w/ "incoming" is
incorrect.
[incoming]
exten => 17036361355,1,Playback(beep)
exten => 17036361355,2,SayDigits(${EXTEN})
exten => 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle mobile phone.
;exten => s,1,Answer()
;exten => s,n,Dial(SIP/17037171234,150,r,t,)
As well doesn't the Goto need to closing ")"?
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130408/24956ec0/attachment.htm>
On Mon, 8 Apr 2013, Thomas Perron wrote:> I am trying to make sure my DID and SIP account details are working > properly and engaging the extensions.conf and dial plan.If you jack up logging, you may see a message on the console like: looking for x in y where x is the extension and y is the context. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000