we are trying to handle incoming sip_calls on our asterisk server, but somehow we always end up getting either 403 or 603 (which should be the same?)
what should happen if it works:
user calls service number -> remote asterisk accepts call from pstn and forwards to our asterisk -> our asterisk accepts incoming sip call -> depending on the DDI/DID according to the dialplan our server starts an external call to "whatever_target"(could be an internal ip phone or external pstn number)
what works so far:
if we configure the primary caller phone as an extension, it works properly as our asterisk correctly accepts it and forwards
the big problem with this:
we obivously can not configure incoming extensions as it is a service line random people will call
the error log:
492212XXXXXXXX - being a real phone number behind the service number
43650XXXXXXX - customer phone that is calling the service number
<--- SIP read from UDP:remote-server-ip:5060 --->
INVITE sip:492212XXXXXXXX6@our-server-ip:5060 SIP/2.0
Via: SIP/2.0/UDP remote-server-ip:5060;rport;branch=z9hG4bK-6d70-1406058084-1272-462
Call-ID: 64d6-439-6222014194124-inCGN2-2-remote-server-ip
CSeq: 2 INVITE
Max-Forwards: 70
To: <sip:492212XXXXXXXX6@our-server-ip:5060>
From: "43650XXXXXXX"<sip:43650XXXXXXX@remote-server-ip>;tag=95ffcd055e0f78f7d5d397020e89288df0ec4476
User-Agent: Dialogic-SIP/10.5.3.372 inCGN2 2
Contact: <sip:43650XXXXXXX@remote-server-ip:5060>
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Supported: path, replaces, timer, tdialog
Session-Expires: 1800
Expires: 300
Organization: Dialogic
Authorization: Digest username="", realm="asterisk", nonce="10ecaeef", response="69597a5b260ecf8c217193b054463175", algorithm=MD5, uri="sip:our-server-ip"
Content-Type: application/sdp
Content-Length: 434
v=0
o=Dialogic_SDP 1919001 0 IN IP4 remote-server-ip
s=Dialogic-SIP
c=IN IP4 83.125.45.83
t=0 0
m=audio 8228 RTP/AVP 0 8 18 4 96 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
<------------->
--- (17 headers 19 lines) ---
Sending to remote-server-ip:5060 (no NAT)
Using INVITE request as basis request - 64d6-439-6222014194124-inCGN2-2-remote-server-ip
No matching peer for '43650XXXXXXX' from 'remote-server-ip:5060'
[Jul 22 19:41:24] NOTICE[30280]: chan_sip.c:22518 handle_request_invite: Sending fake auth rejection for device "43650XXXXXXX"<sip:43650XXXXXXX@remote-server-ip>;tag=95ffcd055e0f78f7d5d397020e89288df0ec4476
<--- Transmitting (no NAT) to remote-server-ip:5060 --->
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP remote-server-ip:5060;branch=z9hG4bK-6d70-1406058084-1272-462;received=remote-server-ip;rport=5060
From: "43650XXXXXXX"<sip:43650XXXXXXX@remote-server-ip>;tag=95ffcd055e0f78f7d5d397020e89288df0ec4476
To: <sip:492212XXXXXXXX6@our-server-ip:5060>;tag=as52e8819e
Call-ID: 64d6-439-6222014194124-inCGN2-2-remote-server-ip
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '64d6-439-6222014194124-inCGN2-2-remote-server-ip' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:remote-server-ip:5060 --->
ACK sip:492212XXXXXXXX6@our-server-ip:5060 SIP/2.0
Via: SIP/2.0/UDP remote-server-ip:5060;rport;branch=z9hG4bK-6d70-1406058084-1272-462
Call-ID: 64d6-439-6222014194124-inCGN2-2-remote-server-ip
CSeq: 2 ACK
Max-Forwards: 70
To: <sip:492212XXXXXXXX6@our-server-ip:5060>;tag=as52e8819e
From: "43650XXXXXXX"<sip:43650XXXXXXX@remote-server-ip>;tag=95ffcd055e0f78f7d5d397020e89288df0ec4476
User-Agent: Dialogic-SIP/10.5.3.372 inCGN2 2
Content-Length: 0
sip.conf for incoming calls
name: incoming
defaultuser: 492212XXXXXX
regexten: null
secret: null
context: home
canreinvite: yes
host: remote-server-ip
ipaddr: null
insecure: invite
port: 5060
disallow: all
allow: g729;ilbc;gsm;ulaw;alaw
dtmfmode: rfc2833
fromdomain: our-server-ip (we also tested with remote server ip)
nat: yes
qualify: yes
type: friend
outboundproxy: our-server-ip (we also tested with remote server ip)
allowguest: yes (in the hopes it would allow all calls, it didnt)
exentions configuration:
'297', 'home', '492212XXXXXX', '1', 'Dial', 'SIP/101'
'298', 'home', '492212XXXXXX101', '1', 'Dial', 'SIP/101'
'296', 'home', '_43ZX.', '1', 'Dial', 'SIP/101'
What would be the correct way to implement a general whitelist for calls incoming on "remote-server-ip"?
Do you see any cpt. obvious flaws in the above configuration? I have to admit i increasinglly become blind to seeing errors in this as i tried to switch all the parameters around to find a solution.
Can I somehow append a method of authentication to be appended to incoming calls via the remotesecret/proxy settings?
I have to admit that my previous asterisk experience mostly was about configuring a few inhouse ip-phones and thats it ^^
