SipDiscount
Created by: damianmontero,Last modification on Tue 28 of Mar, 2006 [11:46 UTC] by enzo
SIPDiscount.com
Website: sipdiscount.com
To sign up: You'll have to download their program
Their hard to find FAQ and Instructions Page
Sipdiscount has been charging for free calls again (Update 26 Feb 2006):
Over and over again (atleast since February 09, 2006) Sipdiscount has been charging their customers for free calls, e.g. to Canada and the USA. Both servers (sip.sipdiscount.com and sip1.sipdiscount.com) have been manipulated, so the caller no longer hears the message how much will be charged for the call. Furthermore, Sipdiscount has not updated their rates on their website. Therefore, customers are illegaly being charged. The only way to find out is to watch your call history on their website, free calls will be charged half a cent per minute. Even after multiple attempts, Sipdiscount was unable to react on customer complaints and has still not updated their rates, notified their customers or credited for the false billing.
- UPDATE: Since Feb 24 everything seems normal again. I will notify everyone should they do it again, especially since this was not the first time that this happened. As verification here is a list of FREE calls that I made and which were inacceptably billed, check your call history during these days and check if you were billed too. Should it happen again you can expect an update posting faster than they can turn it off.
Free calls charged:
2006-02-09 02:52:28 +1703xxxxxxx 00:00:04 € 0.0050
2006-02-09 02:52:05 +1319xxxxxxx 00:30:04 € 0.1550
2006-02-09 03:22:12 +1319xxxxxxx 00:09:11 € 0.0500
2006-02-09 03:33:49 +1800xxxxxxx 00:00:10 € 0.0050 <- even 800 numbers, yeah!
2006-02-09 03:35:09 +1319xxxxxxx 00:02:30 € 0.0150
2006-02-09 04:08:39 +1514xxxxxxx 00:00:04 € 0.0050
2006-02-09 04:16:55 +1514xxxxxxx 00:00:56 € 0.0050
2006-02-09 04:32:25 +1514xxxxxxx 02:04:02 € 0.6250
2006-02-10 01:36:05 +1319xxxxxxx 00:02:21 € 0.0150
2006-02-10 01:38:44 +1319xxxxxxx 00:00:41 € 0.0050
2006-02-10 02:13:33 +1319xxxxxxx 00:01:04 € 0.0100
2006-02-10 02:57:43 +1514xxxxxxx 00:00:11 € 0.0050
2006-02-10 02:58:06 +1514xxxxxxx 00:56:00 € 0.2800
2006-02-10 04:16:18 +1514xxxxxxx 00:03:52 € 0.0200
2006-02-15 17:56:15 +1319xxxxxxx 00:01:12 € 0.0100
2006-02-19 00:10:36 +1319xxxxxxx 00:01:38 € 0.0100
2006-02-19 00:12:44 +1514xxxxxxx 00:01:10 € 0.0100
2006-02-19 00:15:17 +1514xxxxxxx 00:00:21 € 0.0050
2006-02-19 00:24:00 +1514xxxxxxx 00:53:19 € 0.2700
2006-02-19 14:17:50 +1514xxxxxxx 00:21:39 € 0.1100
2006-02-19 19:17:59 +1514xxxxxxx 00:00:13 € 0.0050
2006-02-19 19:18:44 +1514xxxxxxx 02:53:05 € 0.8700
2006-02-21 03:26:52 +1514xxxxxxx 00:00:20 € 0.0050
2006-02-21 03:27:26 +1514xxxxxxx 00:00:08 € 0.0050
2006-02-21 04:15:07 +1514xxxxxxx 00:02:18 € 0.0150
2006-02-21 03:27:41 +1514xxxxxxx 00:49:55 € 0.2500
2006-02-21 04:17:45 +1514xxxxxxx 01:37:48 € 0.4900
2006-02-22 02:56:20 +1514xxxxxxx 00:00:10 € 0.0050
2006-02-22 02:56:45 +1514xxxxxxx 00:02:07 € 0.0150
Look at how fascinating this is:
2006-02-22 11:48:51 +49203364xxxx 00:03:02 FREE!
2006-02-22 12:13:04 +49203364xxxx 00:01:04 € 0.0100
2006-02-22 12:17:03 +49203364xxxx 00:01:22 FREE!
It seems as if they fixed the problem (starting Feb 24), no more calls are billed wrong. Should it stay like this, I will remove the article from the top and change it into a smaller note (bottom of page).
===
Well, this is not quite correct. They still charge for "free destinations" in case of "unfair use" as they call it. By this they mean excessive usage of course.
Instructions
You need a SIP Discount username and password in order to call via their network. Simply sign up and create your login. If you need help configuring your SIP device, please check the manual that came with your SIP phone or router.The setting below should work for outgoing & incoming calls (if you have got a voipin number on your account page).
It has been tested with a french local 01... voipin (DID) number.
Settings
Username: Your SIP Discount username
Password: Your password
SIP/Proxy registrar: sip1.sipdiscount.com (new: see recommended settings)
(Feb 2006: sip.sipdiscount.com can NOT handle IAX: sip1.sipdiscount.com) It declares itself as "User-Agent: (Very nice Sip Registrar Server)"). Note: sip1.sipdiscount.com resolves to the same set of IP addresses as connectionserver.voipstunt.com, and VoipStunt offers many more free destinations (see Finarea SA's page)
Domain/Realm (optional): sipdiscount.com
STUN server: stun.sipdiscount.com
Codecs
- GSM (13.2 kbps) (NOT on sip1.sipdiscount.com)
- G.711 (64 kbps, alaw for incoming calls)
- G.726 (32 kbps) (ONLY on sip1.sipdiscount.com)
- G.729 (8 kbps) (ONLY on sip1.sipdiscount.com)
- G.723 (5.3 & 6.3 kbps) (ONLY on sip1.sipdiscount.com)
Free destinations:
For an updated list, please refer to the page for Finarea SA (or, better, to the provider's official rates page)
Asterisk Settings
In sip.conf:
[general]
context=incoming ; important as incoming from sipdiscount will come to that context
register => {{YOURUSERNAME}}:{{YOURPASSWORD}}@sip1.sipdiscount.com
[sipdiscount]
type=peer
host=sip1.sipdiscount.com
fromdomain=sip1.sipdiscount.com
progressinband=yes
dtmfmode=inband
disallow=all
allow=alaw ; only alaw works with sip1...
;allow=g729 ; but no way to have DMTF with G.729 !
nat=yes
canreinvite=no
qualify=yes
insecure=very
context=incoming
username={{YOURUSERNAME}}
fromuser={{YOURUSERNAME}}
secret={{YOURPASSWORD}}
In extensions.conf:
[incoming]
; incoming from SIPDISCOUNT, which is having no context, only the default one ...
exten => {{YOURUSERNAME}},1,Goto(s,1) ; very important, you arrive as username, not s !
exten => s,1,Answer()
exten => s,2,Dial( ... )
If you live in the US, it might help to add a qualify=yes statement to the above and regularely check your ping time since this provider is based in Europe.
To dial out using this use the line:
Dial(SIP/00{EXTEN}@sipdiscount)
in the appropriate place in your extensions.conf dialplan.
You can also register with SIP using:
register => YOURUSERNAME:YOURPASSWORD@sip1.sipdiscount.com
(But I don't think this is nessesary since they're no dialing into you, unless someone ELSE has a sipDiscount account and dials your NAME.
Update: Feb 24, 2006: It did not work with IAX.
The NumberToCall is always in international format: 00 + country code + area code + local number
(that's why the DIAL line above has 00 in front of the {EXTEN}
Asterisk@Home Settings (Using 2.5)
Note: The calling plan does NOT isolate the mobile numbers in each of these networks.Note2: This is setup to seperate north america and international because in my configuration I have alternate North America calling providers and have multiple trunks in trunk sequence.
The configuration below will allow you to dial out using SipDiscount in north american fashion. e.g: 1-416-555-1234 will be coverted into 0014165551234. 4165551234 will be converted into 0014165551234. 5551234 will be coverted into 0014165551234 and all international dialing will drop the 011 and add 00.
Add SIP Trunk:
Maximum Channles: 1
Dial Rules:
1+NXXNXXXXXX
1425+NXXXXXX <NOTE: Change 425 to your local area code>
011|.
1425+NXXXXXX <NOTE: Change 425 to your local area code>
011|.
Outbound Dial Prefix: 00
Trunk Name: SIPDiscountFree
Peer Details:
allow=gsm&ulaw
auth=md5
disallow=all
host=sip.sipdiscount.com
qualify=yes
secret=<PASSWORD>
type=peer
username=<USERNAME>
auth=md5
disallow=all
host=sip.sipdiscount.com
qualify=yes
secret=<PASSWORD>
type=peer
username=<USERNAME>
Clear all from User Context and User Details
<Registration is required>
Register String: <USERNAME>:<PASSWORD>@sip.sipdiscount.com
Setup Outbound Routing:
Add Route: NorthAmerica
Dial Patterns:
911
1800NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX
1NXXNXXXXXX
NXXNXXXXXX
NXXXXXX
Trunk Sequence: Add SipDiscountFree
Add Route: SipDiscountFree FIXME <Remove Pay Mobile Numbers>
Dial Patterns:
01130.
01131.
01132.
01133.
01134.
011351.
011352.
011354.
011358.
01139.
01141.
011423.
01143.
01144.
01145.
01146.
01147.
01149.
01131.
01132.
01133.
01134.
011351.
011352.
011354.
011358.
01139.
01141.
011423.
01143.
01144.
01145.
01146.
01147.
01149.
Trunk Sequence: Add SipDiscountFree
Isolation of chargeable routes from free routes
Please get the latest version of the LCDial.sh AGI script or the LCDC least cost dialplan compiler.
Comments
333registering
<br>
All I had to change was the name of the 'section' from sipdiscount to sip.sipdiscount.com to match the hostname of the server.<br>
Here it is in its entirity (works for me - incoming and outgoing) :-<br>
333No incoming audio
This turned out to be out of date dynamic DNS :/ I use dyndns.org and it had failed to update after the last IP address change. After updating my dyndns account with my latest IP details the audio sprang back into action.
333Sipdiscount, Freecall, Voipstunt?
80.239.235.200 is sip.voiparound.com and sip1.sipdiscount.com
80.239.235.201 is sip.voipstunt.com and sip.sipdiscount.com
which is why i've had to create 3 different user ids for this free calling stuff!
anyone getting one server to work but not the other? i get problems with "ALL CIRCUITS ARE BUSY" most of the time lately and odnt know why. it used to work fine
333Unable to reach servers
Have had this problem for a while :(
/TKJ
C:\Documents and Settings\Administrator>tracert sip1.sipdiscount.com
Sporer rute til sip1.sipdiscount.com 194.120.0.203
over et maksimum af 30 hop:
1 77 ms 8 ms 8 ms 10.80.0.1
2 99 ms 6 ms 7 ms 10.250.0.161
3 97 ms 37 ms 105 ms ge-0-1-0-125.680m.boanxj1.ip.tele.dk
97'>62.242.37.
97
4 81 ms 8 ms 7 ms ge1-2-50.1000m.boanxg4.ip.tele.dk
5 102 ms 8 ms 15 ms pos4-0.2488m.boanxg2.ip.tele.dk
6 11 ms 11 ms 12 ms pos3-0.2488m.kd4nxg2.ip.tele.dk
7 25 ms 74 ms 25 ms pos5-0.2488m.asd9nxg1.ip.tele.dk
8 27 ms 27 ms 78 ms asd-sara-ias-ur10.nl.kpn.net
9 * * * Anmodning fik timeout.
10 * * * Anmodning fik timeout.
11 28 ms 31 ms 27 ms asd2-rou-1012.NL.eurorings.net
12 39 ms 30 ms 29 ms ffm-s1-rou-1021.DE.eurorings.net
4'>134.222.231.13
4
13 34 ms 41 ms 28 ms ffm-s1-rou-1001.DE.eurorings.net
'>134.222.227.50
14 * * * Anmodning fik timeout.
15 * * * Anmodning fik timeout.
16 * * * Anmodning fik timeout.
17 * * * Anmodning fik timeout.
18 * * * Anmodning fik timeout.
19 * * * Anmodning fik timeout.
20 * * * Anmodning fik timeout.
21 * * * Anmodning fik timeout.
22 * * * Anmodning fik timeout.
23 * * * Anmodning fik timeout.
24 * * * Anmodning fik timeout.
25 * * * Anmodning fik timeout.
26 * * * Anmodning fik timeout.
27 * * * Anmodning fik timeout.
28 * * * Anmodning fik timeout.
29 * * * Anmodning fik timeout.
30 * * * Anmodning fik timeout.
Sporing fuldført.
333Incoming calls - no codecs
Capabilities: us - 0x18 (alaw|g726), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Mar 5 16:37:35 NOTICE28017: chan_sip.c:3312 process_sdp: No compatible codecs!
It seems they send INVITES with no codec capabilities!
333
Sorry being quite busy for a couple of days I could not check back the page.
Here's my config
1) have
register=USERNAME:PASSWORD:USERNAME@sip1.sipdiscount.com
in the beginning of your sip_additional.conf
2) also I have defined
Intl_out
username=XXXXXXXXXX (your sipdisount username)
type=peer
secret=XXXXXXXXXXXXXX (your sip discount password)
qualify=yes
insecure=very
host=sip1.sipdiscount.com
fromuser=hceylan
fromdomain=sipdiscount.com
dtmfmode=inband
domain=sipdiscount.com
authuser=XXXXXXXXXXX (your sip user again)
in the same file....
3) in extensions_additional.conf, set this
OUT_2 = SIP/Intl_out
4) in the same file
outrt-002-Intl
include => outrt-002-Intl-custom
exten => _900.,1,Macro(dialout-trunk,2,${EXTEN:1},)
exten => _900.,2,Macro(outisbusy) ; No available circuits
5) In the same file under outbound-allroutes, add this
include => outrt-002-Intl
This should do it. Note that my config does NOT disallow non-free calls.
So I just be carefully where I call.
Soon I plan to implement also preventing non-free calls in the configuration.
Hope this works for you as it worked for me...
Ceylan
333
33331 second problem
If you can share how you fixed the problem, that would be great.
Thanks
John
33331 second problem solved
I solved the problem with 31 sec hangups by tweaking couple of params in *. If you or anybody else is still having the problem, I can share details here...
Ceylan
333Re: IAX
IAX support is being discontinued, but AFAIK is still available if you use the old server, sip.sipdiscount.com . The New server sip1.sipdiscount.com is SIP-only, not based on Asterisk, and actually provided by TVIConnect.