DID not correctly provisioned? Hmm................ interesting. I seem to be having the same issue with them.
Unfortunately, most every other provider, for my area code, 405, says they require using their equipment and charges a fairly significant setup fee. Too much for a proof of concept. Otherwise I would gladly switch. At this point I probably will after the proof of concept. Their support is proving too weak.
Oh, and the 14 digit number is the usual 10 digit number + a 4 digit extension that you are prompted to enter after dialing the 10, 7 if you are local, digit number.
Anyway I have been playing with it some more this weekend. I never could get it to work with SIP. Upon further research I found that they are using IAX. Hence, the use of port 5036. Not only that they are using the old IAX, version 1 again explaining the port number. After modifying the make file in the /usr/src/asterisk/channels directory to allow version 1 and applying the change I am able to get the register line to work.
-- Registered to '198.175.8.53', who sees us as 68.97.xxx.xxx:5036
An IAX1 show registry confirms this.
Host Username Perceived Refresh State
198.175.8.53:5036 405227xxxx 68.97.xxx.xxx:5036 60 Registered
Still not perfect though. With debug on I am being inundated with these warnings (mostly just an annoyance I'm sure)
Jan 8 21:22:26 WARNING[2024]: chan_iax.c:3334 iax_ack_registry: Unknown variable 'mwi' with value '0'
Jan 8 21:22:26 WARNING[2024]: chan_iax.c:3334 iax_ack_registry: Unknown variable 'vmsg' with value '0'
Jan 8 21:22:26 WARNING[2024]: chan_iax.c:3334 iax_ack_registry: Unknown variable 'plan' with value 'vmpaid'
However, I am still unable to dial out. I get this message back.
Jan 8 21:31:07 WARNING[2024]: chan_iax.c:3964 socket_read: Call rejected by 198.175.8.53: No authority found
In looking at an ethereal trace of the call conversation it doesn't appear to be related to authentication. It didn't appear to make it that far.
I did a compare of a successful call using the glophone client and a failed call from asterisk and it is interesting to note that the glophone client uses these options.
exten=1405323xxxx;
callerid=700xxxxxxx;
dnid=1405323xxxx;
username=405227xxxxxxxx;
formats=2;
version=1;
Whereas, Asterisk uses these
exten=1405323xxxx;
callerid=700xxxxxxx;
username=405227xxxxxxxx;
formats=2;
version=1;
language=en;
context=myphone.voiceglo.coô; <--- this is how it looks....even in both Asterisk console mode and ethereal
capability=64558;
adsicpe=0
As can be seen Asterisk has 4 extra fields and, probably most impotantly, is missing the the 'dnid' field, which matches the 'exten' field.
I did find this patch which may resolve the 'dnid' - 'exten' issue. http://asterisk.gnuinter.net/patches/asterisk-dnid.patch. Unfortunately, I am unsure how to apply the patch. Any pointers about how to apply the patch would be greatly appreciated. Whether or not it would work.
Does anyone know if this can be corrected?
Thanks
JV
----- Original Message -----
Date: Thu, 6 Jan 2005 08:35:50 -0700 (MST)
From: Greg Hill <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Glophone/Voiceglo and Asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: TEXT/PLAIN; charset=US-ASCII
On Thu, 6 Jan 2005, John Voss wrote:
> Has anyone managed to get Asterisk to work with Glophone/Voiceglo
> since this posting.
>
> http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html
>
> I've tried copying the config in this listing with no success.
>
> One thing that I have noticed is that all the listings that I have
> found mention the use
> of 10 digit numbers. They now give you 14 digit numbers which
> shouldn't matter. However,
> it does make me wonder if anything else has changed.
>
> Any help anyone can supply will be greatly appreciated.
14 digit numbers..? I could imagine 13, with 011 prepended to the
numbers.. hmm.
The config in the post you reference looks similar to the one that I used,
which is (approximately):
[voiceglo]
type=peer
username=801203xxxx
secret=NEERHFDxxxx
;nat=yes
host=myphone.voiceglo.com
disallow=all
;disallow=g729
allow=ulaw
;allow=alaw
;allow=gsm
;allow=g729
canreinvite=no
;qualify=400
restrictid=no
fromdomain=myphone.voiceglo.com
dtmfmode=inband
I did have it working at one point, however, I didn't (still don't) have
the g729 codec for my asterisk. I could only place calls through Voiceglo
by using their bundled SJ Labs software (which did include g729) and
setting it to register through my *. This way * never needed to listen to
the RTP stream anyway and could just pass it through. At the time, g729
was the only codec you could use. And they also used inband DTMF -- a very
bad combination.
I cancelled my service after they failed to correct (or even recognize) a
significant problem: the DID they assigned me was provisioned incorrectly
(routing config problem, evidently) and could not be reached from at least
one local (to me) ILEC exchange. In fact, they didn't even recognize my
(repeated) requests to cancel the account. Funny thing was, when I asked
my credit card company to chargeback Voiceglo, I got a call within just a
few days from a Voiceglo rep, who acted surprised to have received a
chargeback and wanted to know why I hadn't contacted them first to see if
we couldn't resolve any problems. I nearly hung up on her.
Maybe they've done some hiring and firing since then and run a better shop
now. This stuff is all I know about them, and it's nearly a year out of
date. Anyway, if there is anybody else who can provide service in the area
you need, I think I might recommend going that route instead.
Greg
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