<P>DID not correctly provisioned? Hmm................ interesting. I seem
to be having the same issue with them.</P>
<P>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.</P>
<P>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.</P>
<P>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.</P>
<P> -- Registered to '198.175.8.53',
who sees us as 68.97.xxx.xxx:5036<BR></P>
<P>An IAX1 show registry confirms this.</P>
<P>
Host
Username
Perceived
Refresh
State<BR>
198.175.8.53:5036 405227xxxx
68.97.xxx.xxx:5036
60
Registered</P>
<P>Still not perfect though. With debug on I am being inundated with these
warnings (mostly just an annoyance I'm sure)</P>
<P> Jan 8 21:22:26 WARNING[2024]:
chan_iax.c:3334 iax_ack_registry: Unknown variable 'mwi' with value
'0'<BR> Jan 8 21:22:26
WARNING[2024]: chan_iax.c:3334 iax_ack_registry: Unknown variable 'vmsg'
with value '0'<BR> Jan 8
21:22:26 WARNING[2024]: chan_iax.c:3334 iax_ack_registry: Unknown variable
'plan' with value 'vmpaid'<BR><BR>However, I am
still unable to dial out. I get this message back. </P>
<P> Jan 8 21:31:07 WARNING[2024]:
chan_iax.c:3964 socket_read: Call rejected by 198.175.8.53: No authority
found</P>
<P>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.</P>
<P>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.</P>
<P>
exten=1405323xxxx;<BR>
callerid=700xxxxxxx;<BR>
dnid=1405323xxxx;<BR>
username=405227xxxxxxxx;<BR>
formats=2;<BR> version=1;</P>
<P>Whereas, Asterisk uses these</P>
<P>
exten=1405323xxxx;<BR>
callerid=700xxxxxxx;<BR>
username=405227xxxxxxxx;<BR>
formats=2;<BR> version=1;</P>
<P>
language=en;<BR>
context=myphone.voiceglo.coô; <--- this is how it looks....even
in both Asterisk console mode and
ethereal<BR>
capability=64558;<BR> adsicpe=0</P>
<P>As can be seen Asterisk has 4 extra fields and, probably most
impotantly, is missing the the 'dnid' field, which matches the
'exten' field.</P>
<P>I did find this patch which may resolve the 'dnid' -
'exten' issue. <A
href="http://asterisk.gnuinter.net/patches/asterisk-dnid.patch">http://asterisk.gnuinter.net/patches/asterisk-dnid.patch</A>.
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.</P>
<P>Does anyone know if this can be corrected? </P>
<P>Thanks</P>
<P>JV</P>
<P>----- Original Message -----<BR>Date: Thu, 6 Jan 2005 08:35:50
-0700 (MST)<BR>From: Greg Hill <<A
href="http://mail02.mail.com/scripts/mail/compose.mail?compose=1&.ob=0d4c6eb457d304ffd815005ad7d3a340c309aeb5&composeto=gregh-asterisk%40hillnet.us">gregh-asterisk@hillnet.us</A>><BR>Subject:
Re: [Asterisk-Users] Glophone/Voiceglo and Asterisk<BR>To: Asterisk Users
Mailing List - Non-Commercial Discussion<BR><<A
href="http://mail02.mail.com/scripts/mail/compose.mail?compose=1&.ob=0d4c6eb457d304ffd815005ad7d3a340c309aeb5&composeto=asterisk-users%40lists.digium.com">asterisk-users@lists.digium.com</A>><BR>Message-ID:
<<A
href="http://mail02.mail.com/scripts/mail/compose.mail?compose=1&.ob=0d4c6eb457d304ffd815005ad7d3a340c309aeb5&composeto=Pine.LNX.4.44.0501060825350.32471-100000%40hillnet.us">Pine.LNX.4.44.0501060825350.32471-100000@hillnet.us</A>><BR>Content-Type:
TEXT/PLAIN; charset=US-ASCII<BR><BR>On Thu, 6 Jan 2005, John Voss
wrote:<BR><BR>> Has anyone managed to get Asterisk to work with
Glophone/Voiceglo <BR>> since this posting.<BR>><BR>>
<A
href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html"
target=_blank><FONT
color=blue>http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</FONT></A><BR>><BR>>
I've tried copying the config in this listing with no
success.<BR>><BR>> One thing that I have noticed is that all
the listings that I have <BR>> found mention the use<BR>> of
10 digit numbers. They now give you 14 digit numbers which
<BR>> shouldn't matter. However,<BR>> it does make me
wonder if anything else has changed.<BR>><BR>> Any help anyone
can supply will be greatly appreciated.<BR><BR>14 digit numbers..? I
could imagine 13, with 011 prepended to the<BR>numbers..
hmm.<BR><BR>The config in the post you reference looks similar to
the one that I used,<BR>which is
(approximately):<BR>[voiceglo]<BR>type=peer<BR>username=801203xxxx<BR>secret=NEERHFDxxxx<BR>;nat=yes<BR>host=myphone.voiceglo.com<BR>disallow=all<BR>;disallow=g729<BR>allow=ulaw<BR>;allow=alaw<BR>;allow=gsm<BR>;allow=g729<BR>canreinvite=no<BR>;qualify=400<BR>restrictid=no<BR>fromdomain=myphone.voiceglo.com<BR>dtmfmode=inband<BR><BR>I
did have it working at one point, however, I didn't (still don't)
have<BR>the g729 codec for my asterisk. I could only place calls through
Voiceglo<BR>by using their bundled SJ Labs software (which did include
g729) and<BR>setting it to register through my *. This way * never needed
to listen to<BR>the RTP stream anyway and could just pass it through. At
the time, g729<BR>was the only codec you could use. And they also used
inband DTMF -- a very<BR>bad combination.<BR><BR>I cancelled
my service after they failed to correct (or even recognize)
a<BR>significant problem: the DID they assigned me was provisioned
incorrectly<BR>(routing config problem, evidently) and could not be
reached from at least<BR>one local (to me) ILEC exchange. In fact, they
didn't even recognize my<BR>(repeated) requests to cancel the account.
Funny thing was, when I asked<BR>my credit card company to chargeback
Voiceglo, I got a call within just a<BR>few days from a Voiceglo rep, who
acted surprised to have received a<BR>chargeback and wanted to know why I
hadn't contacted them first to see if<BR>we couldn't resolve any
problems. I nearly hung up on her.<BR><BR>Maybe they've done
some hiring and firing since then and run a better shop<BR>now. This stuff
is all I know about them, and it's nearly a year out of<BR>date.
Anyway, if there is anybody else who can provide service in the
area<BR>you need, I think I might recommend going that route
instead.<BR><BR>Greg<BR><BR><BR><BR>--
</P>
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