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VoIPDiscount with Asterisk - Success!

 
 
Sparks
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Posts: n/a
 
      10-26-2005
>
> Perhaps you would be kind enough to post the relevent section of your
> sip_additional.conf file?
>
> Mine's:
>
> [locsipdisc]
> username=MyUsername
> type=peer
> secret=MyPassword
> qualify=yes
> nat=yes
> insecure=very
> host=sip.sipdiscount.com
> fromuser=MyUsername
> fromdomain=sipdiscount.com
> disallow=all
> authuser=MyUsername
>
> Cheers.
>


I see your problem

You have disallowed all codecs (disallow=all) but not allowed any!

Mine is...

[sipdiscount]
username=MyUsername
type=peer
secret=MyPassword
qualify=yes
nat=yes
insecure=very
host=sip.sipdiscount.com
fromuser=MyUsername
fromdomain=sipdiscount.com
disallow=all
authuser=MyUsername
allow=ulaw
allow=alaw


Sparks...


 
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Jono
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Posts: n/a
 
      10-26-2005


Sparks wrote:
||| Perhaps you would be kind enough to post the relevent section of
||| your sip_additional.conf file?
|||
||| Mine's:
|||
||| [locsipdisc]
||| username=MyUsername
||| type=peer
||| secret=MyPassword
||| qualify=yes
||| nat=yes
||| insecure=very
||| host=sip.sipdiscount.com
||| fromuser=MyUsername
||| fromdomain=sipdiscount.com
||| disallow=all
||| authuser=MyUsername
|||
||| Cheers.
|||
||
|| I see your problem
||
|| You have disallowed all codecs (disallow=all) but not allowed any!
||
|| Mine is...
||
|| [sipdiscount]
|| username=MyUsername
|| type=peer
|| secret=MyPassword
|| qualify=yes
|| nat=yes
|| insecure=very
|| host=sip.sipdiscount.com
|| fromuser=MyUsername
|| fromdomain=sipdiscount.com
|| disallow=all
|| authuser=MyUsername
|| allow=ulaw
|| allow=alaw
||
||
|| Sparks...

I'll give it a try.

Thanks.


 
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Jono
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Posts: n/a
 
      10-26-2005


Sparks wrote:
||| Perhaps you would be kind enough to post the relevent section of
||| your sip_additional.conf file?
|||
||| Mine's:
|||
||| [locsipdisc]
||| username=MyUsername
||| type=peer
||| secret=MyPassword
||| qualify=yes
||| nat=yes
||| insecure=very
||| host=sip.sipdiscount.com
||| fromuser=MyUsername
||| fromdomain=sipdiscount.com
||| disallow=all
||| authuser=MyUsername
|||
||| Cheers.
|||
||
|| I see your problem
||
|| You have disallowed all codecs (disallow=all) but not allowed any!
||
|| Mine is...
||
|| [sipdiscount]
|| username=MyUsername
|| type=peer
|| secret=MyPassword
|| qualify=yes
|| nat=yes
|| insecure=very
|| host=sip.sipdiscount.com
|| fromuser=MyUsername
|| fromdomain=sipdiscount.com
|| disallow=all
|| authuser=MyUsername
|| allow=ulaw
|| allow=alaw
||
||
|| Sparks...

That made a massive difference - I can now ring out, get the 0 Euro Cents
message....But NO audio

Grrrr

BTW, why is disallow=all in there anyway?


 
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Sparks
Guest
Posts: n/a
 
      10-26-2005

> That made a massive difference - I can now ring out, get the 0 Euro Cents
> message....But NO audio


.....bugger!

not sure what to suggest, other than checking your routers settings!


>
> Grrrr
>
> BTW, why is disallow=all in there anyway?


It is so you can then specify the codecs you want to use with that trunk -
if for example your default was changed to GSM, then this trunk will stay
with whatever you have specified.


 
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Jono
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Posts: n/a
 
      10-26-2005


Sparks wrote:
||| That made a massive difference - I can now ring out, get the 0 Euro
||| Cents message....But NO audio
||
|| ....bugger!
||
|| not sure what to suggest, other than checking your routers settings!
||
||
|||
||| Grrrr
|||
||| BTW, why is disallow=all in there anyway?
||
|| It is so you can then specify the codecs you want to use with that
|| trunk - if for example your default was changed to GSM, then this
|| trunk will stay with whatever you have specified.

Thanks for your help getting this far.

I might be investing in a new router soon, though I can't see anything out
of sorts there.


 
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Jono
Guest
Posts: n/a
 
      10-26-2005


Sparks wrote:
||| That made a massive difference - I can now ring out, get the 0 Euro
||| Cents message....But NO audio
||
|| ....bugger!
||
|| not sure what to suggest, other than checking your routers settings!
||
||
|||
||| Grrrr
|||
||| BTW, why is disallow=all in there anyway?
||
|| It is so you can then specify the codecs you want to use with that
|| trunk - if for example your default was changed to GSM, then this
|| trunk will stay with whatever you have specified.

Well bugger me. I've finally got it working.

I had to create an IAX2 trunk with the following settings:
(note the use of Voipbuster's servers.)

allow=ulaw&alaw
authuser=MySipDiscountUsername
disallow=all
fromdomain=voipbuster.com
fromuser=MySipDiscountUsername
host=iax.voipbuster.com
insecure=very
nat=yes
qualify=yes
secret=MySipDiscountPassword
type=peer
username=MySipDiscountUsername


 
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paul123
Guest
Posts: n/a
 
      10-27-2005
Hi Sparks and all,

I have a teething problem with Sipdiscount on Asterisk@home. I've used
your settings in Asterisk@home (beta 2.1), but found I wasn't getting
the account as "registered" when looking at the Asterisk info page...

However, as I'd had a similar problem with voipbuster, I changed the
lines:
>>fromdomain=sipdiscount.com
>>host=sip.sipdiscount.com

to
>>fromdomain=83.138.185.232
>>host=213.61.187.146

and it registers and works!....well, almost.

I can make calls but I get a one way audio problem - I can't hear the
person I'm calling, though they can hear me.

Any ideas?

Paul

 
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sameerms18 sameerms18 is offline
Junior Member
Join Date: Nov 2011
Posts: 1
 
      11-18-2011
Hi,

i have configured my voipdiscount account in elastix. i can call my local sip extensions. i am in saudi arabia now. i want to make outside calls through my voip provider trunk. what i need to do? i am new in elastix. i dont know how to configure dialing pattern for voipdiscount to make call to outside. please help me
 
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