Comments on: Internet Telephony Shootout II: Finding the Best International VoIP Provider for Asterisk

By: AM

AM — Thu, 15 Mar 2007 23:11:24 +0000

Axvoice is just a reseller – their backend is RNKTelecom – a very solid provider. So when you e-mail support, you are actually e-mailing RNK. Now try to get a hold of Axvoice for billing or problems with an account – you’ll find that they’re probably just a guy sitting in a bedroom running the site. There are other resellers of RNK – Voxby.com is one. I’m with Axvoice but am trying to get out.

[WM: Let’s be fair. There are at least a couple of guys.]

By: Kristopher

Kristopher — Wed, 14 Feb 2007 22:06:35 +0000

Hooray! Fixed it! Turns out I had the context wrong in my peer details. Corrected that and all was well. Thanks NV and U. Ward!

[WM: Just like size… context matters.]

By: Kristopher

Kristopher — Wed, 14 Feb 2007 21:29:41 +0000

I’m having what appears to be the same problem as Fabrizio. I’m using AxVoice and I can place calls but can’t receive. When I call myself from an external line such as a cell phone, I’m getting the "The number you have dialed is not in service." This is clearly coming from my PBX and not AxVoice, based on these lines I noticed watching the CLI:

— Executing Playback("SIP/ksandrick-08857508″, "ss-noservice") in new stack
— Playing ‘ss-noservice’ (language ‘en’)
— Executing PlayTones("SIP/ksandrick-08857508″, "congestion") in new stack
— Executing Congestion("SIP/ksandrick-08857508″, "5″) in new stack

I can’t seem to find any resources, anywhere that indicate what the incoming settings should be, other than blank, which doesn’t seem to be working. Any thoughts/insight/assistance would be greatly appreciated.

Very respectfully,
Kris

[WM: I have repeatedly contacted AxVoice about these connection problems. They swear the problems are generated at our end by connection at odd UDP ports instead of 5060. If you want it to work, you’ll need to call them and politely ask what port they are showing your connection on. It’ll probably be something up in the 30,000 to 50,000 UDP range. Then open that port on your firewall and point it to the internal address of your Asterisk server. An Asterisk UDP expert can probably explain why this is happening, but all I know is how to fix it. Sorry.]

By: Minime

Minime — Sun, 15 Oct 2006 21:08:30 +0000

I think asterisk is great. I use NV trixbox because it is the best VMware precompiled machine out there. I have 4 incoming lines working and created a 00X. outgoing route and added the same on every trunk.

When I dial any outgoing international number (for ex: 0032484425525) the lady tells me the number (this example: 32484425526) is not in the directory. Goodbye!

I am missing something here but have been trying for over a month now but can not seem to find it. Any ideas? I am desperate for a solution.

By: Keith

Keith — Fri, 25 Aug 2006 19:35:10 +0000

the last comment *asterisk* reporting that, because the ‘from-sip-external’ only has it. You need to comment that portion out and add in where you want the calls to go.

Also, AXVoice does not pass DID or CID info. So if you have a couple #s that you want to point to different location (ie, one # to directly dial you, one # to goto Digital Receptionist) that will not work.
I’m in talks with this about this necessity, and hopefully they fix it soon.

By: Fabrizio

Fabrizio — Fri, 21 Apr 2006 01:40:41 +0000

I’m wondering why there is no incoming settings required? I am having problems getting calls with axvoice.. When monitoring the system from CLI.. I get the following;

— Executing Set("SIP/mezzan-c767″, "TIMEOUT(absolute)=15″) in new stack

— Executing Set("SIP/myusername-f168″, "TIMEOUT(absolute)=15″) in new stack
— Channel will hangup at 2006-04-21 01:38:52 UTC.
— Executing Answer("SIP/myusername-f168″, "") in new stack
— Executing Wait("SIP/myusername-f168″, "2″) in new stack
— Executing Playback("SIP/myusername-f168″, "ss-noservice") in new stack
— Playing ‘ss-noservice’ (language ‘en’)
— Executing Congestion("SIP/myusername-f168″, "") in new stack
== Spawn extension (from-sip-external, s, 5) exited non-zero on ‘SIP/myusername-f168’

When I dial in to the PBX i get a message "The number you have dialed is not in service"

Not sure what I did wrong since I followed instructions here and triple checked..

[WM: Contact the provider. If the number is not in service, there’s a problem at their end.]

By: Marcello

Marcello — Wed, 29 Mar 2006 02:42:12 +0000

I have configured my asterisk box for axvoice using exactly the instructions provided… Could not figure out why asterisk is not registering to axvoice. Ports are opened on the firewall and other Voip providers are working just fine in my * box. Any ideas? I turned the qualify=yes and get unreachable on sip show peers. On the full log, I get 216.143.130.36 timeout (sip.axvoice.com)…

[WM: Sounds like an incorrect username or password. If not, it may be something at Axvoice’s end. Give them a call. They have great support.]

By: Jason

Jason — Mon, 23 Jan 2006 22:07:14 +0000

Since Axvoice uses asterisk as well, feature codes overlap. While it’s nice to assume we’ll never need to use the features on the Axvoice side with out own asterisk installation, fact is outages happen, carrier or otherwise. In order to be able to dial asterisk feature codes from hand sets I’ve been trying to find some sort of digit translation… atleast that’s what it’s called in the avaya world. Say **98 is dialed, I want *98 outpulsed on the Axvoice trunk. I’ve googled asterisk digit translation and came up empty, perhaps I don’t know the correct Asterisk term for what I want to do? All I really want is **. = *. on x trunk. Any ideas?

By: Dave

Dave — Sat, 14 Jan 2006 18:56:55 +0000

Ward,

I’ve signed up to use Axvoice service and have followed your instructions to configure AAH for axvice. Incoming calls are working OK but I’m having a problem with outgoing calls. When I try to make an outgoing call, I get the following message in the Asterisk CLI:

–SIP/axvoive-35ab is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)

From googling, I’ve come across some references to wrong CODEC being used, but am unsure if this is the problem.

Any help would be appreciated.

Thanks,
Dave

[WM: Call AxVoice. They have very good tech support.]

By: Mike Haslam

Mike Haslam — Thu, 29 Dec 2005 00:57:15 +0000

Tried this configuration but incoming calls are not working. Fromt he cli in Asterisk with verbose on 10 it shows:

Verbosity was 3 and is now 10
— Executing AbsoluteTimeout("SIP/stayhosted-a275″, "15″) in new stack
— Set Absolute Timeout to 15
— Executing Congestion("SIP/stayhosted-a275″, "") in new stack
== Spawn extension (from-sip-external, s, 2) exited non-zero on ‘SIP/stayhosted-a275’
— Executing AbsoluteTimeout("SIP/stayhosted-a275″, "15″) in new stack
— Set Absolute Timeout to 15
— Executing Congestion("SIP/stayhosted-a275″, "") in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on ‘SIP/stayhosted-a275’

Any help greatly appreciated.

[WM: Take a look at the Managing Incoming Calls section of our HOW-TO article on Asterisk@Home 2.2.]

By: p2pvoice

p2pvoice — Thu, 10 Nov 2005 05:26:01 +0000

Hello WM:
I am enjoying – and learning and using – your ongoing A@H series. Just to bring this to your attention, I signed up for a 30-day trial with axvoice and then decided to cancel it. Here is what they responded with:

Hi,

We do not support faxes. If you have any quality issues we might be able to
assist you if you can call our Technical Support at 888-277-8647. We can
cancel your account but the money is not refundable because the 30-Day money
back Guarantee does’nt apply to BYOD option. Minimum term for this option is
one-month.

Support
Axvoice Inc.

Needless to say, I am disappointed with their response.
Thanks

By: Blair

Blair — Sun, 23 Oct 2005 02:50:02 +0000

Thank you for the link to the Asterisk pdf.
Keep up the great work.

By: dstroot

dstroot — Mon, 17 Oct 2005 18:10:29 +0000

Ward - I have followed your stuff from the beginning and have a full-blown implementation with a SPA-3000, IAXy, etc. (I was actually slightly ahead of you in trying "stuff" but you have taught me a great deal. PLEASE keep it up! The one thing you mentioned a while back is automatic call blocking and unblocking. I'd really like to give this a try. I found some instructions from another blogger and plan to give it go but was wondering if you have a clearly written article coming up in the Ward Mundy style (i.e. easy to implement) ;) [WM: Hopefully next week we'll get to call blocking or blacklisting. If you want to get a head start (again), here's the guy that put it down where the goats could get it.]

By: Mike

Mike — Mon, 17 Oct 2005 13:34:18 +0000

What?!!? …We don’t do books… Ward, all I can say is THANKS for your BOOK of tutorials on REAL world examples… Where the REAL lessons take place!!! 🙂