<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>
<P>___________________________________________________________<BR>Sign-up for Ads Free at Mail.com<BR><A href="http://mail01.mail.com/scripts/payment/adtracking.cgi?bannercode=adsfreejump01" target=_blank>http://www.mail.com/?sr=signup</A></P><BR>
--
<p>___________________________________________________________<br>Sign-up for Ads Free at Mail.com<br>
<a href="http://mail01.mail.com/scripts/payment/adtracking.cgi?bannercode=adsfreejump01" target="_blank">http://www.mail.com/?sr=signup</a></p>