This is a short tutorial on how to test the cnxcc module with live traffic and from any softphone. These are the things that you need:
– Register from any softphone:
* server: cnxcc.caruizdiaz.com:5070
* username: anything between 5000-9999. Actually anything will work, but I’d rather use those numbering ranges
* password: anything. It is not required
* credit: you will get 1$ credit every time you register. If you place a call and run out of credit, just register again and you will get +1$ on your account.
– Place a call: you can actually call to any number with the following pattern: “^09\d{8}$”. This basically means that you can call to a number starting with “09”, that has 8 digits afterwards. Example: 0981223344
– You will run out of credit: after a little more than 10 seconds, you will run out of credit and your call will be terminated. Don’t worry, you will receive a SIP MESSAGE on you UA, if it supports it, telling you what just happened.
– Recharge your line: if you want more credit, just re-REGISTER from your UA
– Check you credit: by texting “credit” to destination “service“, from your UA that supports SIP MESSAGE.
– Check the web: you can actually hang up somebody else’s call, or your call, from the web interface. It looks nice and you can log in using “guest” as username and “123456” password.
– Call each other: you have the option to call other registered users, but for this, you have to know the number of the other person, or guess it randomly 😉
– Duplicated username: you may have chosen the same username that someone else did. If that’s the case, your call will last less than 10 seconds if you call simultaneously, or not even start if someone already exhausted the credit. In the latter case, you will receive a SIP MESSAGE telling you that you have no credit left ;).
These are the files involved in this project. I hope you find them useful.
The demo has no media relay configured, so, you won’t have audio unless you are calling each other on the same network.
hello, is this setup still live?
cnxcc.caruizdiaz.com:5070
?
thanks from hamburg,
rené
it is not, but I can put it online again if you want to give it a try 🙂
I would love to including the web-interface. Thanks!
At the moment I get the forbidden error, while calling from my SNOM
Sent to udp:107.170.102.188:5070 at 11/8/2014 18:43:24:645 (1223 bytes):
INVITE sip:[email protected]:5070;user=phone SIP/2.0
Via: SIP/2.0/UDP 79.202.219.252:3073;branch=z9hG4bK-2bmj1apewjwz;rport
From: “7777” ;tag=63bl2ztb9e
To:
Call-ID: acf2e8539b6f-9yjsas1gfru7
CSeq: 1 INVITE
Max-Forwards: 70
Contact: ;reg-id=1
X-Serialnumber: 000413412891
P-Key-Flags: resolution=”31×13″, keys=”4″
User-Agent: snom870/8.7.3.19
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 402
v=0
o=root 1110839919 1110839919 IN IP4 79.202.219.252
s=call
c=IN IP4 79.202.219.252
t=0 0
m=audio 20028 RTP/AVP 9 0 8 99 108 18 101
a=direction:both
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
Received from udp:107.170.102.188:5070 at 11/8/2014 18:43:24:768 (368 bytes):
SIP/2.0 100 trying — your call is important to us
Via: SIP/2.0/UDP 79.202.219.252:3073;branch=z9hG4bK-2bmj1apewjwz;rport=3073
From: “7777” ;tag=63bl2ztb9e
To:
Call-ID: acf2e8539b6f-9yjsas1gfru7
CSeq: 1 INVITE
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0
Received from udp:107.170.102.188:5070 at 11/8/2014 18:43:24:787 (867 bytes):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 79.202.219.252:3073;branch=z9hG4bK-2bmj1apewjwz;rport=3073
From: “7777” ;tag=63bl2ztb9e
To: ;tag=F6Zgg9m8FBr8e
Call-ID: acf2e8539b6f-9yjsas1gfru7
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.2.24+git~20140513T183205Z~9af707ed20~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Proxy-Authenticate: Digest realm=”cnxcc.caruizdiaz.com”, nonce=”9c55186e-2176-11e4-af84-1dbd4e65c7e3″, algorithm=MD5, qop=”auth”
Content-Length: 0
Sent to udp:107.170.102.188:5070 at 11/8/2014 18:43:24:790 (432 bytes):
ACK sip:[email protected]:5070;user=phone SIP/2.0
Via: SIP/2.0/UDP 79.202.219.252:3073;branch=z9hG4bK-2bmj1apewjwz;rport
From: “7777” ;tag=63bl2ztb9e
To: ;tag=F6Zgg9m8FBr8e
Call-ID: acf2e8539b6f-9yjsas1gfru7
CSeq: 1 ACK
Max-Forwards: 70
Contact: ;reg-id=1
Content-Length: 0
Sent to udp:107.170.102.188:5070 at 11/8/2014 18:43:24:797 (1497 bytes):
INVITE sip:[email protected]:5070;user=phone SIP/2.0
Via: SIP/2.0/UDP 79.202.219.252:3073;branch=z9hG4bK-qrc6rnlixgdf;rport
From: “7777” ;tag=63bl2ztb9e
To:
Call-ID: acf2e8539b6f-9yjsas1gfru7
CSeq: 2 INVITE
Max-Forwards: 70
Contact: ;reg-id=1
X-Serialnumber: 000413412891
P-Key-Flags: resolution=”31×13″, keys=”4″
User-Agent: snom870/8.7.3.19
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest username=”7777″,realm=”cnxcc.caruizdiaz.com”,nonce=”9c55186e-2176-11e4-af84-1dbd4e65c7e3″,uri=”sip:[email protected]:5070;user=phone”,qop=auth,nc=00000001,cnonce=”459f5447″,response=”6ce18ab860602da0e2a105a35a08c185″,algorithm=MD5
Content-Type: application/sdp
Content-Length: 402
v=0
o=root 1110839919 1110839919 IN IP4 79.202.219.252
s=call
c=IN IP4 79.202.219.252
t=0 0
m=audio 20028 RTP/AVP 9 0 8 99 108 18 101
a=direction:both
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
Received from udp:107.170.102.188:5070 at 11/8/2014 18:43:24:922 (368 bytes):
SIP/2.0 100 trying — your call is important to us
Via: SIP/2.0/UDP 79.202.219.252:3073;branch=z9hG4bK-qrc6rnlixgdf;rport=3073
From: “7777” ;tag=63bl2ztb9e
To:
Call-ID: acf2e8539b6f-9yjsas1gfru7
CSeq: 2 INVITE
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0
Received from udp:107.170.102.188:5070 at 11/8/2014 18:43:24:940 (717 bytes):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 79.202.219.252:3073;branch=z9hG4bK-qrc6rnlixgdf;rport=3073
From: “7777” ;tag=63bl2ztb9e
To: ;tag=gFS9H45BDmeUa
Call-ID: acf2e8539b6f-9yjsas1gfru7
CSeq: 2 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.2.24+git~20140513T183205Z~9af707ed20~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
Sent to udp:107.170.102.188:5070 at 11/8/2014 18:43:24:943 (432 bytes):
ACK sip:[email protected]:5070;user=phone SIP/2.0
Via: SIP/2.0/UDP 79.202.219.252:3073;branch=z9hG4bK-qrc6rnlixgdf;rport
From: “7777” ;tag=63bl2ztb9e
To: ;tag=gFS9H45BDmeUa
Call-ID: acf2e8539b6f-9yjsas1gfru7
CSeq: 2 ACK
Max-Forwards: 70
Contact: ;reg-id=1
Content-Length: 0
Ok, I will put it online again, today, in a couple of hours.
I’ll let you know.
The demo is online again
hello, Need some paid help withthe cnxx. Would be really helpfull.