[asterisk-users] SIPBroker vs SIPgate

Adrian Marsh Adrian.Marsh at ubiquisys.com
Wed Sep 5 10:21:56 CDT 2007


I had to turn Sipbroker off at one point, as I found that some Conf.
Calls on a 3rd party system didn't like the DTMF being passed (users
unable to enter conferences).  I traced all the failures to calls
passing out via SIPbroker, disabled it so the calls went via PSTN and
all was well..

Now I'm trying to re-work the call logic to include it again.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of SIP
Sent: 04 September 2007 18:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIPBroker vs SIPgate

Seriously, from our experience, SIPBroker IS the best way to interact 
with all the open networks. For any closed networks, you might create 
special rules for interaction, but that would rely on setting up a deal 
with the respective destination network to actually ALLOW your calls.

There are some pay per play networks that do peering automagically (such

as XConnect), but it's a cost per connected call (granted, a tiny one, 
but still a cost), and it won't guarantee you any better connectivity to

a closed network than, say, SIPBroker.

N.


Adrian Marsh wrote:
> Yeah,
>
> I can see that now after testing it all - but this is what raised my
> question..  What IS the best mechanism for all the VoIP
servers/networks
> to interact ? Setting up individual agreements for each "network" is
so
> 1980's, and in this modern world there must be a better solution..
>
> A.
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of SIP
> Sent: 04 September 2007 15:14
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SIPBroker vs SIPgate
>
> Adrian Marsh wrote:
>   
>> All,
>>
>> I've been experimenting with shortcodes for SIPgate etc.  Passing
>>     
> calls
>   
>> to SIPbroker seems a good way to go, but the message I've had back
>>     
> from
>   
>> SIPgate is "we don't support SIPBroker"...
>>
>> So whats the easiest way to support SIP <> SIP network calling?
>>
>> At the moment, I've setup some local shortcodes (eg dial **777. to
>>     
> goto
>   
>> sipgate.co.uk) based on what Gradwell have publically posted, but I
>> can't even get SIPgate to work with this either !!  (Can't pass these
>> directly to Gradwell as their SIP trunks don't support it..)
>>
>> A.
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>>     
> SIP <-> SIP calling across networks really only works if the receiving

> network allows incoming calls from non-local networks.  SIPgate does 
> not, so unless you're registered on the SIPgate network, calling
another
>
> SIPgate user from your SIPgate number, it won't accept the call.
>
> N.
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>   


_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



More information about the asterisk-users mailing list