How to have dynamic extensions

Asterisk Asterisk Support
1

I have recently swapped VoIP providers. So far so good, but I have ran into one little snag (and I’m honestly not really sure how to ask / Google this).

With my old provider I could have inside my extensions.conf inbound context each of the landing point numbers as such:

exten => _6123456779,1,Goto(1300s,${EXTEN},1) exten => _6123456773,1,Goto(1300s,${EXTEN},1)
That worked great, but with my new provider, all the calls go to the /extension in the register function or to extension s.

Old providers sip.conf:

;This comes through as the phone number dialed register => username:password@10.10.10.10 [CL] type=peer host=10.10.10.10 context=from-cl qualify=yes disallow=all allow=ulaw nat=yes canreinvite=no dtmfmode=rfc2833 insecure=port,invite username=username authuser=username secret=password

New providers sip.conf:

[code];This comes through as extension s

register => phonenumber:password@trunk.engin.com.au

[CL]
auth=phonenumber
username=phonenumber
host=trunk.engin.com.au
secret=password
type=peer
insecure=invite,port
nat=yes
qualify=yes
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
defaultexpirey=1800
maxexpirey=3600
context=from-cl[/code]
So how do I re-enable this functionality?

Or do I have to do:

register => 6123456770,1:password@trunk.engin.com.au/6123456770 register => 6123456770,1:password@trunk.engin.com.au/6123456771 register => 6123456770,1:password@trunk.engin.com.au/6123456772
etc,etc?

This is the sip INVITE:

[code]<— SIP read from UDP:10.10.10.10:5060 —>
INVITE sip:s@10.10.10.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bKl7qoae00agdqjdh18ks0.1
From: sip:0466666666@voice.mibroadband.com.au;user=phone;tag=SDl4lp901-1922148923-1430721154048-
To: "Full Name"sip:6123456771@voice.mibroadband.com.au
Call-ID: SDl4lp901-8c3fdacbe5fbc40c18c775a59025a687-jm6gpa0020
CSeq: 248522241 INVITE
Contact: sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@10.10.10.10:5060;transport=udp
P-Called-Party-ID: sip:6123456771@voice.mibroadband.com.au
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,multipart/mixed
Min-SE: 60
Session-Expires: 1800;refresher=uas
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 327

v=0
o=BroadWorks 525881593 1 IN IP4 10.10.10.10
s=-
c=IN IP4 10.10.10.10
t=0 0
m=audio 17968 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:30
a=sqn: 0
a=cdsc:1 image udptl t38
<------------->[/code]

2

I figured a work around, I’m not terribly happy with it but using this dialplan I can translate the s into a proper extension with the following:

[from-cl] exten => s,1,NoOp(${SIP_HEADER(To)}) exten => s,n,Set(DID=${SIP_HEADER(To)}) exten => s,n,Set(DID=${CUT(DID,:,2)}) exten => s,n,Set(DID=${CUT(DID,@,1)}) exten => s,n,Goto(from-pstn,${DID},1)

3

This is a problem with your ITSP, and I think you have found the best workround.

Are you really sure you need insecure=port?

4

Honestly, I have no idea. I knew before I had put in insecure=invite,port it didn’t work at all. But it might only require invite. Should I try without port?

Also, what exactly does it do?

5

insecure=invite is the old alternative to remotesecret, for when only one side authenticates It stops passwords being checked on incoming calls.

insecure=port causes Asterisk to match calls based only on the IP address, and may be needed in some NAT cases, but usually isn’t. Both remove security checks. ITSP like giving options that are least likely to fail to make calls, not ones that are secure.