asterisk users - Feb 2005 - VONAGE <----> ASTERISK SIP TERMINATION?????

Has anyone out there successfully set up their * box to terminate their
VONAGE calls? 

 

I (and I am sure lots of others) would love to hear how you did it.

 

I'd like to be able to get rid of the extra hardware I have hanging around
here and use the ASTERISK machine to handle the SIP termination instead of
needing to have a Linksys modem (w/phone) and an additional X100P card.

 

Thanks.

Wishing for a solution.

(lucasNOSPAM@NOSPAMeyeonsystems.com)

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050218/4dacad20/attachment.htm

Yes, it's doable, had this running for several months here.  However,
you'll need to get a softphone for $10/month from them, and they'll
provide the sip-credentials on their website.  It's a lousy solution if
you really just want one number, because then you'll have to pay
$15/month for their basic "hardline" service, plus an extra $10 for
the
softline.  It may make sense if you need several numbers in rate centers
where the usual suspects don't have numbers -- as it did for me.  Once a
less expensive provider popped up, I traded the $25 Vonage mess for a $5
unlimited DID.

http://lists.digium.com/pipermail/asterisk-users/2004-June/052678.html

This still works, just add "insecure=very"

-----Original Message-----
From: Lucas Wrenn [mailto:lucas@eyeonsystems.com] 
Sent: Friday, February 18, 2005 9:25 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] VONAGE <----> ASTERISK SIP TERMINATION?????


Has anyone out there successfully set up their * box to terminate their
VONAGE calls? 
 
I (and I am sure lots of others) would love to hear how you did it.
 
I'd like to be able to get rid of the extra hardware I have hanging
around here and use the ASTERISK machine to handle the SIP termination
instead of needing to have a Linksys modem (w/phone) and an additional
X100P card.
 
Thanks.
Wishing for a solution.
(lucasNOSPAM@NOSPAMeyeonsystems.com)

Make sure you have the proper licenses to use the codecs:

g729
http://www.digium.com/index.php?menu=asterisk_g729

g723
http://www.dspg.com/technology/LicensePricing.html


On Fri, 18 Feb 2005 18:17:58 -0800, Nitesh Divecha
<nitesh@vipernetworks.com> wrote:
> Hello All,
> 
> Any one has success with codec g723 & g729?
> I am having extremely hard time to setup this codec.
> The only codec worked is g711a/u.
> 
> If I set g723 & g729 as first and second choice codec in my sip.conf,
VM and
> MeetMe stop working.
> 
> Sip.conf
> 
> [general]
> port = 5060           ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
> disallow=all
> ;allow=g273
> ;allow=g729
> allow=ulaw
> allow=alaw
> 
> #include sip_nat.conf
> #include sip_additional.conf
> 
> I am using Snom 220/200 and all are set to use g729.
> 
> Thank you,
> 
> Nitesh
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>