[Asterisk-biz] Forklift a 2000 phone PBX - 5000 calls up

[steve]

Hi:

Their outgoing termination is different than their incoming DID's.   

Steve
-----Original Message-----
From: asterisk-biz-bounces at lists.digium.com
[mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Preston Garrison
Sent: Monday, March 28, 2005 2:03 PM
To: yair at hakak.com; asterisk-biz at lists.digium.com
Subject: Re: [Asterisk-biz] Forklift a 2000 phone PBX - 5000 calls up

Actually since they use companies like Level3 who hands off SIP, they 
don't need anything more then a media proxy.  With the exception of 
needing to do things like voicemail, which SER actually has modules 
available to handle that.

Preston Garrison
direct: 877-748-4142
fax: 310-774-3901
cell: 623-748-4140

-----Original Message-----
From: Yair Hakak <yhakak at gmail.com>
To: Commercial and Business-Oriented Asterisk Discussion 
<asterisk-biz at lists.digium.com>
Sent: Mon, 28 Mar 2005 21:26:33 +0200
Subject: Re: [Asterisk-biz] Forklift a 2000 phone PBX - 5000 calls up

I get the feeling that the vast majority of vonage calls are
vonage->PSTN (which requires the SIP proxy to hand off to a gateway),
not vonage<->vonage. Couple that with the fact that a large portion of
vonage users are behind NATs (and STUN isn't exactly 100%) , and i
start thinking that there is some vonage component (SER, asterisk,
jasomi, whatever) in the media path in the majority of cases.

In case of vonage<->vonage you're absoutely right (except for NAT).

does this sound logical?

-yair


On Mon, 28 Mar 2005 11:09:18 -0800, steve <steve at 17q.com> wrote:
> Hi:
>
> That is somewhat true but it is not the same as SER.  Example:   With 
SER
> there is no codex translation.  Vonage is using SER with over 500,000 
users.
>
> Steve
>
> -----Original Message-----
> From: asterisk-biz-bounces at lists.digium.com
> [mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of steve 
szmidt
> Sent: Monday, March 28, 2005 9:34 AM
> To: Commercial and Business-Oriented Asterisk Discussion
> Subject: Re: [Asterisk-biz] Forklift a 2000 phone PBX - 5000 calls up
>
> On Monday 28 March 2005 12:12, alex at pilosoft.com wrote:
> > On Mon, 28 Mar 2005, Dan Iordanescu wrote:
> > > If you define the sip users in sip.conf with: canreinvite=yes 
then *
> > > behaves like SER; it's not in the middle anymore. Asterisk-users 
list
> > > has a lot more on this stuff.
> >
> > Stick to selling epygi, Dan, that doesn't require thinking.
>
> That's uncalled for Alex.
>
> If someone does not understand then try to educate or at least offer
> constructive critisism.
>
> --
>
> Steve Szmidt
>
> "They that would give up essential liberty for temporary safety
> deserve neither liberty nor safety."
>                                Benjamin Franklin
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