[cisco-voip] SIP Trunk providers

[Matt Slaga (US)]

If you want more than one SIP line, you will need to find a SIP trunk
provider.  If you try to use one of these BYOD like viatalk or others,
you will quickly find that each line requires its own credentials.
Problem is, CME only allows one set of credentials total.  You need to
find a SIP trunk provider like broadvox or Time Warner (or others), and
definitely check to see which has the shortest RTT.  If you are lucky,
your ISP might provide you with SIP service.  That is usually best
because they can provide some type of SLA where no remote one would.

The other issue you may run into is codec, DTMF relay and a few others.
Some providers only want you to run G711, which over the internet is
generally not a good idea.  Some only provide in-band DTMF which means
your CME will need extra DSPs to remove and place out of band (which is
only what Cisco supports at the moment, at least when I last configured
this several months ago).

Hope this helps!

Matt

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert Bell
Sent: Monday, December 17, 2007 7:40 AM
To: 'cisco.voip'; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SIP Trunk providers

Well I've setup a broadvoice.com account.  They were one of the only
providers when I did my research that had a BYOD (Bring Your Own Device)
plan.  Some others you could make your gear work with, but they charged
you
the same.  With BroadVoice you can get a $7-8 plan (subject to taxes).  

I'll be interest to hear if anyone else has other info now.  BV had a
lot of
bad press right before I joined the because of a major outage, but I'm
just
playing with VoIP at the moment to stay up on technology and I've found
them
reliable.  My problem stems mostly from my provider (who will remain
nameless, but I have to wonder if they couldn't do more in the QoS arena
for
me - but they have their own VoIP provider).  Okay, I've said too much
on
that.

And I've never setup SRST, but I was under the impression that was
failover
for CCM to CCME when the Primary or subscriber CCM are unavailable to a
remote site so local comms continues.  I think you'll have to get very
crafty with your dial-peers and give them the SIP dial-peer a higher
priority then the POTS one.  Hey, but I'm a newbie also, so if you have
an
idea or find one here, I'm always interested.

B

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of cisco.voip
Sent: Monday, December 17, 2007 1:00 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SIP Trunk providers

I am tring to setup a small office with a cisco UC500 for their phone 
system.
I want to connect the calls over the internet connection via a SIP
provider 
and use the FXO port for SRST backup.
 Does anyone have a recommendation for a SIP provider to terminate
internet 
calls to PSTN for a reasonable price.
Also, how do I manage the pone number for dialing access, if the local
pstn 
line is number 301.123.4567, how do I get that number to be serviced
by the sip provider.

Thanks


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