Crazy Boy
2006-Oct-25 01:54 UTC
[asterisk-users] Call is not coming through sipgate.co.uk+Asterisk
Hi,
I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I
got registered with sipgate.co.uk and got the UK phone number i.e., 0207100xxxx.
I configured my Asterisk server with 0207100xxxx. When I made a call to this
number from outside phone, my XLite extension is not ringing. Its directly going
to Voicemail or telling that "person is unavailable". When I made a
call, Asterisk console is also not showing anything. But, sipgate website is
showing my calls list. I thought that When I made a call from outside to my
number, call is going to sipgate.co.uk and its not routing to my server. When I
execute "sip show registry", its not displaying anything.
Here I am giving my configuration details:
My sip.conf file contents:
[general]
port = 5060
bindaddr = 0.0.0.0
qualify=no
disable=all
allow=alaw
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
[250]
type=friend
username=250
secret=danny
callerid="Danny"
host=dynamic
context=demo
register => 100xxxx:password@sipgate.co.uk/100xxxx
[sipgate4]
type=friend
disallow=all
allow=alaw
allow=ulaw
fromuser=100xxxx
authuser=100xxxx
secret=password
username=100xxxx
host=sipgate.co.uk
context=demo
dtmfmode=info
fromdomain=sipgate.co.uk
insecure=very
nat=yes
canreinvite=no
callerid="Danny" <0207100xxxx>
My Extensions.conf file contents:
[demo]
exten => 250,1,Dial(SIP/250,20)
exten => 250,2,Voicemail(u250)
exten => 250,3,Voicemail(b250)
exten => 250,4,Hangup
exten => _0207.,1,SetCallerID("" <100xxxx>|a)
;Outgoing
exten => _0207.,2,Dial(SIP/${EXTEN:4}@sipgate4,40,tr)
exten => 100xxxx,1,Dial(SIP/250,30,tr)
;Incoming
Am I have to install any other libraries?
Anything wrong in the above configuration?
Looking forward to your response. Thanks in advance.
Regards,
Chandra.
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Dovid B
2006-Oct-25 04:52 UTC
[asterisk-users] Call is not coming through sipgate.co.uk+Asterisk
Are you behind NAT. Any firewall's ?
----- Original Message -----
From: Crazy Boy
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 25, 2006 10:54 AM
Subject: [asterisk-users] Call is not coming through sipgate.co.uk+Asterisk
Hi,
I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system.
I got registered with sipgate.co.uk and got the UK phone number i.e.,
0207100xxxx.
I configured my Asterisk server with 0207100xxxx. When I made a call to this
number from outside phone, my XLite extension is not ringing. Its directly going
to Voicemail or telling that "person is unavailable". When I made a
call, Asterisk console is also not showing anything. But, sipgate website is
showing my calls list. I thought that When I made a call from outside to my
number, call is going to sipgate.co.uk and its not routing to my server. When I
execute "sip show registry", its not displaying anything.
Here I am giving my configuration details:
My sip.conf file contents:
[general]
port = 5060
bindaddr = 0.0.0.0
qualify=no
disable=all
allow=alaw
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
[250]
type=friend
username=250
secret=danny
callerid="Danny"
host=dynamic
context=demo
register => 100xxxx:password@sipgate.co.uk/100xxxx
[sipgate4]
type=friend
disallow=all
allow=alaw
allow=ulaw
fromuser=100xxxx
authuser=100xxxx
secret=password
username=100xxxx
host=sipgate.co.uk
context=demo
dtmfmode=info
fromdomain=sipgate.co.uk
insecure=very
nat=yes
canreinvite=no
callerid="Danny" <0207100xxxx>
My Extensions.conf file contents:
[demo]
exten => 250,1,Dial(SIP/250,20)
exten => 250,2,Voicemail(u250)
exten => 250,3,Voicemail(b250)
exten => 250,4,Hangup
exten => _0207.,1,SetCallerID("" <100xxxx>|a)
;Outgoing
exten => _0207.,2,Dial(SIP/${EXTEN:4}@sipgate4,40,tr)
exten => 100xxxx,1,Dial(SIP/250,30,tr)
;Incoming
Am I have to install any other libraries?
Anything wrong in the above configuration?
Looking forward to your response. Thanks in advance.
Regards,
Chandra.
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Brian Candler
2006-Oct-25 07:01 UTC
[asterisk-users] Call is not coming through sipgate.co.uk+Asterisk
On Wed, Oct 25, 2006 at 01:54:43AM -0700, Crazy Boy wrote:> My sip.conf file contents:...> [250] > type=friend > username=250 > secret=danny > callerid="Danny" > host=dynamic > context=demo > register => 100xxxx:password@sipgate.co.uk/100xxxx...> My Extensions.conf file contents: > [demo] > exten => 250,1,Dial(SIP/250,20) > exten => 250,2,Voicemail(u250) > exten => 250,3,Voicemail(b250) > exten => 250,4,Hangup > exten => _0207.,1,SetCallerID("" <100xxxx>|a) > ;Outgoing > exten => _0207.,2,Dial(SIP/${EXTEN:4}@sipgate4,40,tr) > exten => 100xxxx,1,Dial(SIP/250,30,tr) > ;Incoming > Am I have to install any other libraries?No. In the first case, getting incoming calls to work is easy. Start with a configuration which has nothing to do with sipgate in it. At the top of sip.conf you should have a [general] section, and you can put the registration statement there, i.e. [general] register => 100XXXX:PPPPPPPP@sipgate.co.uk/101 context=default In this case, incoming calls to your sipgate.co.uk PSTN number will ring as 101 in context 'default'. I've just tested this with a sipgate.co.uk and it works fine. (I actually have two accounts, with two register statements, pointing at two different extensions) Now, getting outbound to work is a little harder. You need a new entry in sip.conf to place outbound calls. My first attempt was: [sipgate-out] type=peer host=sipgate.co.uk username=100XXXX secret=PPPPPPPP fromuser=100XXXX fromdomain=sipgate.co.uk With the correct extensions.conf config (see below), outbound calls worked. Unfortunately, doing this stopped incoming calls from working; they are rejected with "401 unauthorised" because Asterisk now explictly matches this SIP entry for incoming calls from sipgate.co.uk, in preference to [general] So what I eventually ended up with was: [sipgate-out] type=friend host=sipgate.co.uk username=100XXXX secret=PPPPPPPP fromuser=100XXXX fromdomain=sipgate.co.uk insecure=invite ;context=default ; not required because I have this in [general] still I'm not sure if this is the best way to go, but it does seem to work. I tried moving the "register" lines under [sipgate-out] and Asterisk no longer registered. Perhaps "register" doesn't work for friend entries? Finally, you need a rule in extensions.conf to route outbound calls via this link, in whichever context(s) your local phone(s) sit where you want to allow outbound calls. For example: [internal] exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate-out,15,r) exten => _9.,2,Congestion() exten => _9.,102,Congestion() This will match all numbers which begin with 9, and route them via sipgate, stripping off the leading 9. Regards, Brian. P.S. All my testing was with SVN trunk, which is close to 1.4. Behaviour may be different in earlier versions.