Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas
Hi
That does the job (dialout only)
sip.conf
[sipdiscount]
type=peer
host=sip1.sipdiscount.com
dtmfmode=inband
disallow=all
allow=gsm
allow=ulaw
allow=alaw
fromdomain=stun.sipdiscount.com
username=YOURLOGIN
fromuser=YOURLOGIN
secret=YOURPASSWORD
canredirect=no
nat=yes
qualify=yes
###
extensions.conf
exten => _00X.,1,Verbose
exten => _00X.,2,Dial(SIP/sipdiscount/${EXTEN})
exten => _00X.,3,Hangup
###
On Wed, Feb 08, 2006 at 08:20:05PM +0100, Alejandro Vargas
wrote:> Sipdiscount has replaced their asterisk servers for another thing.
> Then, no more iax. Ok, but I can't make calls using sip also... I'm
> getting a "forbidden" error when using sip1.sipdiscount.com.
Anybody
> got it working?
> --
> Alejandro Vargas
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> This message has been scanned and dangerous content has been removed by our
> MailScanner. This includes all executables. If the transfer of executables
> is desired please consider to send them as a zip-file, which is allowed to
pass
> the checks, but which will be scanned for viruses.
>
> Please be sure to keep your local Antivirus up-to-date, as this message is
no
> guarantee that all viruses have been removed.
>
>
On 08/02/06, Alejandro Vargas <alejandro.anv@gmail.com> wrote:> Sipdiscount has replaced their asterisk servers for another thing. > Then, no more iax. Ok, but I can't make calls using sip also... I'm > getting a "forbidden" error when using sip1.sipdiscount.com. Anybody > got it working?A pretty simple setup works for me: sip.conf: [sipdiscount] type=peer host=sip1.sipdiscount.com username=xxxxxxx secret=yyyyyyy canreinvite=no dtmfmode=info extensions.conf: [sipdiscount-out] exten => _6.,1,Dial(SIP/${EXTEN:1}@sipdiscount) exten => _6.,2,Hangup (I use a prefix of '6' to reach sipdiscount) Peter -- Peter Bowyer Email: peter@bowyer.org Tel: +44 1296 768003 VoIP: sip:peter@bowyer.org VoIP: *5048707000@sipbroker.com FWD: **275*5048707000 VoipTalk: **473*5048707000
2006/2/17, Peter Bowyer <peeebeee@gmail.com>:> A pretty simple setup works for me:The problem may be the username/password. But the page says this: SIP Discount offers the possibility to test our service right away, for free! No need to sign up: just enter the account details below in your favorite softphone or ATA and start calling! You can call all destinations marked with * in our rate list . (Trial calls are limited to a maximum duration of 1 minute). To enjoy unlimited calls, simply sign up for SIP Discount. User Name: test Password: test Domain/Realm: sipdiscount.com SIP Proxy/registrar: sip1.sipdiscount.com SIP Outbound Proxy (optional): sip1.sipdiscount.com STUN server (optional): stun.sipdiscount.com The only problem I say is asterisk is not sending stun.sipdiscount.com or sipdiscount.com as domain. It is sending sip1.sipdiscount.com. -- Alejandro Vargas
Hi I would sugest, that you just register without balancing your account. Than use the supplied username/password and it will work. I doubt that the test/test works. Greets Adibar On Fri, Feb 17, 2006 at 12:31:17PM +0100, Alejandro Vargas wrote:> 2006/2/17, Peter Bowyer <peeebeee@gmail.com>: > > A pretty simple setup works for me: > > The problem may be the username/password. But the page says this: > > SIP Discount offers the possibility to test our service right away, > for free! No need to sign up: just enter the account details below in > your favorite softphone or ATA and start calling! You can call all > destinations marked with * in our rate list . (Trial calls are limited > to a maximum duration of 1 minute). To enjoy unlimited calls, simply > sign up for SIP Discount. > > User Name: test > Password: test > Domain/Realm: sipdiscount.com > SIP Proxy/registrar: sip1.sipdiscount.com > SIP Outbound Proxy (optional): sip1.sipdiscount.com > STUN server (optional): stun.sipdiscount.com > > The only problem I say is asterisk is not sending stun.sipdiscount.com > or sipdiscount.com as domain. It is sending sip1.sipdiscount.com. > > -- > Alejandro Vargas > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned and dangerous content has been removed by our > MailScanner. This includes all executables. If the transfer of executables > is desired please consider to send them as a zip-file, which is allowed to pass > the checks, but which will be scanned for viruses. > > Please be sure to keep your local Antivirus up-to-date, as this message is no > guarantee that all viruses have been removed. > >--
2006/2/17, adibar <asterisk@linux.ch>:> I would sugest, that you just register without balancing your > account. Than use the supplied username/password and it will > work. I doubt that the test/test works.Thanks. This worked. I already had a sipdiscoutn account without credit, but It never worked before (always needed to use test). -- Alejandro Vargas
On Fri, 2006-02-17 at 14:32 +0100, Alejandro Vargas wrote:> 2006/2/17, adibar <asterisk@linux.ch>: > > > I would sugest, that you just register without balancing your > > account. Than use the supplied username/password and it will > > work. I doubt that the test/test works. > > Thanks. This worked. I already had a sipdiscoutn account without > credit, but It never worked before (always needed to use test).they may have recently disabled the test account given that if everyone is using it abuse would be high. While a free account does little to stop abuse, it does add a very small hurdle to it, which can slow people down and potentially add for slightly better tracking of problem users. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060217/eee397ec/attachment.pgp
I use the following in my Asterisk 1.2.4: [out-sipdiscount] type=peer secret=mypass username=myuser fromuser=myuser host=sip1.sipdiscount.com call-limit=1 disallow=all allow=ulaw allow=g726 allow=g729 allow=g723.1 Using this configuration, I am able to place calls using a Dial statement such as: Dial(SIP/00442071234567@out-sipdiscount) HTH, Roshan http://roshan.info On Fri, Feb 17, 2006 at 12:31:17PM +0100, Alejandro Vargas scribbled:> 2006/2/17, Peter Bowyer <peeebeee@gmail.com>: > > A pretty simple setup works for me: > > The problem may be the username/password. But the page says this: > > SIP Discount offers the possibility to test our service right away, > for free! No need to sign up: just enter the account details below in > your favorite softphone or ATA and start calling! You can call all > destinations marked with * in our rate list . (Trial calls are limited > to a maximum duration of 1 minute). To enjoy unlimited calls, simply > sign up for SIP Discount. > > User Name: test > Password: test > Domain/Realm: sipdiscount.com > SIP Proxy/registrar: sip1.sipdiscount.com > SIP Outbound Proxy (optional): sip1.sipdiscount.com > STUN server (optional): stun.sipdiscount.com > > The only problem I say is asterisk is not sending stun.sipdiscount.com > or sipdiscount.com as domain. It is sending sip1.sipdiscount.com. > > -- > Alejandro Vargas > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >