please ignore, sorry for noise
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I'm trying to make some recommendations to a company as to what ITSP to
use. I've almost always been locked into Verizon. I wanted to see if
Voip.ms was the overall favorite, or if it is something else. The only
other non Verizon install I did was about 1.5 years ago. I used
bandwidth.com then
i like voip.ms but they are a tier 2 or tier 3 player. if you're serious
about sip look to a large player like l3, att, global crossing, etc. if
you're less serious but it's still important check out your local clec's.
in either case i'd recommend an ingate for compatibility as the major
I've used Voip.ms at about a dozen locations now, but none with more than 8
handsets and three concurrent calls. All out of the Atlanta server.
Zero complaints, and their failover settings have worked for several
locations, just as programmed.
On Tue, Sep 20, 2011 at 1:55 PM, Matthew Kitchin
Thanks. Are there any large players that are known to work directly with
Sipx? I'd prefer not to go with an external SBC if possible, because I
have no experience in it. The circuit would be TW Telecom and their VoIP
is only certified on Cisco.
On 9/20/2011 12:59 PM, Michael Picher wrote:
i
level3 is not certifiable with sipx. they require ip and authentication for
outbound calls.
bandwidth.com works.
there is a list in templates. beyond that it is an ask andsee thing.
On Sep 20, 2011 2:05 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
Thanks. Are there any
I would agree with Mike's assessment here. I've used VOIP.ms for almost
three years now. They are okay, and certainly have improved over the years.
They are much more reliable than they were several years ago. But, I don't
sell them as a sip trunk solution to my customers.
Having said that,
Thanks. Would you consider bandwidth.com a top tier provider? I believe
you quit recommending them at some point.
On 9/20/2011 1:10 PM, Tony Graziano wrote:
level3 is not certifiable with sipx. they require ip and
authentication for outbound calls.
bandwidth.com http://bandwidth.com works.
We use voip.ms for all our long distance calling. We are in a situation where
we have to pay for all long distance calls thru our PRI (around $.065).
As a test, I ran all of our calls thru voip.ms just to see how well it handled
the volume - I had a few issues with DTMF - but call quality and
Thanks. Is broadvox legacy network something that is going away at some
point?
I have had 0 voice quality issues using Verizon VoIP and G729 at my
current company. I would never use them again though because of their
customer service issues. I don't even know if they offer their service
if you
we also are a voxitas reseller (now Appia services). they're trucking also
works with sipx and is also t 38 compatible if required.
On Sep 20, 2011 2:27 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
Thanks. Is broadvox legacy network something that is going away at some
We use Voxitas/Appia
They have been very good over the last 3-4 years.
On Tue, 2011-09-20 at 12:55 -0500, Matthew Kitchin (public/usenet)
wrote:
I'm trying to make some recommendations to a company as to what ITSP to
use. I've almost always been locked into Verizon. I wanted to see if
I haven't heard anything about it going away, and they have a ton of
customers on it now, and more going on. There are compatibility issues with
their Fusion network, and several products are only certified with their
Legacy network, so I think it is hear to stay until they resolve the issues
on
Thanks all. I will get quotes from Broadvox and Appia.
On 9/20/2011 1:31 PM, Matt White wrote:
We use Voxitas/Appia
They have been very good over the last 3-4 years.
On Tue, 2011-09-20 at 12:55 -0500, Matthew Kitchin (public/usenet)
wrote:
I'm trying to make some recommendations to a
MWI was previously working great on my system, but starting (I think) with an
update to 4.4 a week ago, I have stopped seeing MWI, at least for some users.
As shown in the proxy log snip below, when this user 101 tries to subscribe,
the request is denied, but then it is retried and it succeeds.
please see: http://track.sipfoundry.org/browse/XX-9855
On Tue, Sep 20, 2011 at 3:02 PM, Jeff Gilmore j...@thegilmores.net wrote:
MWI was previously working great on my system, but starting (I think) with an
update to 4.4 a week ago, I have stopped seeing MWI, at least for some users.
As
On Tue, Sep 20, 2011 at 10:02 PM, Jeff Gilmore j...@thegilmores.net wrote:
MWI was previously working great on my system, but starting (I think) with an
update to 4.4 a week ago, I have stopped seeing MWI, at least for some users.
Would be great if you could tell us what version you were on
I was on 4.2.1 and am now on sipXecs (4.4.0- 2011-09-02EDT15:37:13
domU-12-31-39-15-62-46)
Thanks,
Jeff
On Sep 20, 2011, at 3:16 PM, George Niculae wrote:
On Tue, Sep 20, 2011 at 10:02 PM, Jeff Gilmore j...@thegilmores.net wrote:
MWI was previously working great on my system, but starting (I
I am seeing some interesting stuff in the sip status log relating to expired
nonces:
2011-09-20T18:14:38.106745Z:309204:INCOMING:INFO:choicevoip.ev.ithaca.ny.us:SipClientTcp-18:41FBC940:SipStatus:Read
SIP message:\nLocal Host:192.168.48.2 Port: 5060\nRemote
On Tue, Sep 20, 2011 at 10:31 PM, Jeff Gilmore j...@thegilmores.net wrote:
I am seeing some interesting stuff in the sip status log relating to expired
nonces:
Yes, I bet is due to REST voicemail authentication required introduced
in 4.4, will check and come back
Thanks,
George
On Tue, Sep 20, 2011 at 10:36 PM, George Niculae geo...@ezuce.com wrote:
On Tue, Sep 20, 2011 at 10:31 PM, Jeff Gilmore j...@thegilmores.net wrote:
I am seeing some interesting stuff in the sip status log relating to expired
nonces:
Yes, I bet is due to REST voicemail authentication required
FYI, I updated the phones to 3.2.4B and the problem was resolved. Yea!
Jeff
On Sep 15, 2011, at 4:02 PM, Jeff Gilmore wrote:
I am sending this a second time, because I don't see it appearing on list or
forum.
I have started having a problem with a Polycom IP320. The problem is that,
on
On Tue, Sep 20, 2011 at 11:01 PM, George Niculae geo...@ezuce.com wrote:
On Tue, Sep 20, 2011 at 10:36 PM, George Niculae geo...@ezuce.com wrote:
On Tue, Sep 20, 2011 at 10:31 PM, Jeff Gilmore j...@thegilmores.net wrote:
I am seeing some interesting stuff in the sip status log relating to
Bump :)
On Mon, Sep 19, 2011 at 10:57 AM, Sven Evensen sven.even...@onrelay.comwrote:
Dont remember this being discussed before so...
Say sipx is set up with two sip trunks, one in CA and one in NY, each with
a seperate DID range. Outgoing calls must have
a CLI in the DID range.
Say we
I think it is a matter of:
1. Whether or not the ITSP will allow it. Some carrier's will not let
you display any number other then the ones you have with them.
2. Whether the transform extension function will do all you are trying
to do with it.
3. I would imagine an extensive dialing plan in an
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Organization: SipXecs Forum
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Hello,
I have an extension that forwards to a PSTN over a Vitelity
sip trunk when busy. If I
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