Discussion: SipDiscount

I finally managed to get my asterisk to register to their server and now I can recieve incoming calls - yippee!<br>
<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>

[sip.sipdiscount.com]
type=friend
host=sip.sipdiscount.com
progressinband=yes
fromuser=USERNAME
authuser=USERNAME
username=USERNAME
secret=PASSWORD
qualify=yes
disallow=all
allow=g729
allow=gsm
dtmfmode=rfc2833
canreinvite=no
nat=yes
context=incoming-sipdiscount
fromdomain=sip.sipdiscount.com
insecure=port,invite

I recently had a problem with incoming and outgoing calls having no incoming audio - the called, or calling party could here me but I could not hear the other party.

This turned out to be out of date dynamic DNS :/ I use dyndns.org and it had failed to update after the last IP address change.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.

i'm sure we've all noticed the same websites with different names but they're sip servers are all the same too
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

Am I the only one unable to reach their 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 [62.242.37.
97]
  4    81 ms     8 ms     7 ms  ge1-2-50.1000m.boanxg4.ip.tele.dk [83.88.9.193]

  5   102 ms     8 ms    15 ms  pos4-0.2488m.boanxg2.ip.tele.dk [83.88.12.41]
  6    11 ms    11 ms    12 ms  pos3-0.2488m.kd4nxg2.ip.tele.dk [83.88.22.162]
  7    25 ms    74 ms    25 ms  pos5-0.2488m.asd9nxg1.ip.tele.dk [83.88.3.58]
  8    27 ms    27 ms    78 ms  asd-sara-ias-ur10.nl.kpn.net [195.69.144.144]
  9     *        *        *     Anmodning fik timeout.
 10     *        *        *     Anmodning fik timeout.
 11    28 ms    31 ms    27 ms  asd2-rou-1012.NL.eurorings.net [134.222.231.66]

 12    39 ms    30 ms    29 ms  ffm-s1-rou-1021.DE.eurorings.net [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.

Incoming calls no longer work for me on sipdicount. I get the following debug information:

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 NOTICE[28017]: chan_sip.c:3312 process_sdp: No compatible codecs!

It seems they send INVITES with no codec capabilities!

Hello John,

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

Ceylan,
If you can share how you fixed the problem, that would be great.

Thanks
John

Hello Kumar,

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

> Seems IAX is no longer supported. Or I'm wrong?

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.

I have the same experience. I am a registered user and having around 10 euros credit. All my calls, doens't matter whether free or non-free, get disconnect after 31 secs. I have sent a mail number of times to sipdiscount customer services, but never heard of anything. Now I am stuck with my money.

Regards,
Kumar