Free Accounts and Minutes with AxVoice

[wardmundy]

*** SORRY. THIS OFFER HAS EXPIRED! ***

This just in from our friends at AxVoice. We use 'em, and we like 'em...

You might also be aware of our new platform that we have been testing for past 4 months. We can offer some free beta test accounts for your users. The account will include a free DID in your choice of area codes in the U.S. and Canada, free incoming/outgoing from/to the US and Canada for at least a month or as long as our beta testing continues. It can then be converted into paid account without any upfront costs.

Use the following credentials.

Beta Signup Link - www.axvoice.com/testaccount/
SIP Server - magnum.axvoice.com:9060
Port - 9060
User Management Portal - https://secure.axvoice.com/user/
Voicemail Access Number - 9000
Support and Feedback - [email protected]

*** For those that hate wading through an entire message thread to get the answer, here it is. ***

Once you get your account information, here are the settings that appear to work as of November 20:

TRUNK NAME: axvoice

PEER DETAILS:
allow=ulaw
authname=*******
canreinvite=no
context=from-trunk-axvoice
defaultip=magnum.axvoice.com
disallow=all
dtmfmode=rfc2833
fromdomain=magnum.axvoice.com
fromuser=******
host=magnum.axvoice.com
insecure=very
nat=yes
port=9060
secret=******
type=friend
user=phone
username=******

Register String:
******:*******@magnum.axvoice.com:9060

Finally, you need to either disable your PBX in a Flash firewall (help-pbx) or insert the following command in the /etc/sysconfig/iptables. Then reboot your server.

-A INPUT -p udp -m udp --dport 9060 -j ACCEPT

 

[BillG]

Anyone have this working yet? I'm using the info above as well but just see "Auth. Sent" in the CLI - doesn't look like it's going to register.

 

[Titanous]

I got exactly the same thing. Could someone post a working config?

 

[BillG]

Whew - so it's not just me! I've tried every combination of settings that I can think of and just can't get this to work on either my new PBXiaF or my older Trixbox.

 

[wardmundy]

Using the sample in the Nerd Vittles article, be sure you adjust the fromdomain, defaultip, and host AND the registration string to magnum.axvoice.com and also add a line that says:

port = 9060

 

[BillG]

I've changed everything to "magnum...", added the line port=9060. I also had to add :9060 to the end of the registration string to get it to try to register on that port. I also tried adding :9060 to every line where the domain is mentioned. Still, though, the closest I can get is "Request Sent" now, no matter what I do. Any other suggestions?

 

[stefchik]

I have set up the trunk as recommended with "magnum" and port 9060, all worked fine and registered on the first try, however outgoing doesn't work and it gives "no such host:axvoice" in the log!
any suggestions anyone?
Thanks

 

[gbh2o]

Mine works for me

Anyone have this working yet? I'm using the info above as well but just see "Auth. Sent" in the CLI - doesn't look like it's going to register.

When they're working [it is a beta, after all], mine works:

Trunk name AxvoiceOut

allow=ulaw,g729
canreinvite=no
context=ax-in
disallow=all
dtmfmode=rfc2833
fromuser=myusername
host=magnum.axvoice.com
insecure=very
nat=no
port=9060
qualify=yes
secret=password
type=peer
username=myusername

User context ax-in

context=from-trunk
secret=password
type=peer
username=my username

registration string = myusernameassword:[email protected]:9060/1mynumber

Then I manually botch the extensions_custom.conf with:

[ax-in]
exten => _1mynumber,1,SetCallerID("AX:${CALLERIDNAME}" <7${CALLERIDNUM}>)
exten => _1mynumber,n,NoOp(Incoming call on AxvoiceIn 1mynumber)
;exten => _1mynumber,n,Background(pls-wait-connect-call)
exten => _1mynumber,n,Dial(local/607@from-internal&sip/mycellnumber@favoritetrunk,60,m)
exten => _1mynumber,n,Voicemail(607@default)
exten => _1mynumber,n,Hangup

 

[BillG]

<shrug> - I set mine up exactly as you have yours, gbh2o, and I get as far as "Request Sent" when I do sip show registry. <shrug> - it's not worth the effort - I haven't had this much trouble with any other provider (except Stanaphone, which I STILL can't get to work.)

 

[wardmundy]

Is UDP 9060 redirected on your router to your PBX in a Flash box??

 

[BillG]

Oh - interesting point. 9060 is now directed to PBXiaF but still no dice. "Request Sent"

 

[axvoice]

Here is a working copy of sip and extension files. I hope this helps.

SIP.conf

[general]
bindport=9060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
context=incoming ; Set the context name. The context which contains your local user's extensions in extensions.conf
register=> username[email protected]:9060

[outbound-trunk]
username=username
host=magnum.axvoice.com
port=9060 ; You have to specifiy a different port if the default
type=peer
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=g729
allow=alaw
allow=ulaw
allow=gsm

;Your local channels
[line1]
userid=line1
type=friend
secret=1234
qualify=4000
host=dynamic
qualify=4000
context=incoming
call-limit=2
dtmfmode=rfc2833

[line2]
userid=line2
type=friend
secret=1234
qualify=4000
host=dynamic
context=incoming
call-limit=2
dtmfmode=rfc2833


EXTENSIONS.CONF
[incoming]
; Incoming calls land here. The following 2 extensions match with dnid sent from axvoice server.
; We pass dnid to your asterisk so that you can perform dnid based routing
exten=> 12129331234,1,Dial(SIP/line1)
exten=> 12129331234,2,Hangup

exten=> 16469201234,1,Dial(SIP/line2)
exten=> 16469201234,2,Hangup

exten=> _X.,1,Dial(SIP/${EXTEN}@outbound-trunk) ; This will call any outbound number
exten=> _X.,2,Hangup

 

[BillG]

No dice - still "Request Sent" Is this a problem on my end or Axvoice?

 

[merlyn]

No dice - still "Request Sent" Is this a problem on my end or Axvoice?

I have a odd question for you BillG ... can you goto the axvoice website and login successfuly?
It sounds like it is sending the request and never getting back to you since it cannot find your account info (just a quess)

I currently cannot even login to the axvoice website. I activated my account using the email that they sent. They send back another email with my account name and password that i keep trying to enter and get nothing but the error ... We are sorry, but we are unable to authenticate that username/password combination. Please try again.


I have not gotten far enough along in the PBX install to know if Axvoice is working or not. But if my account got screwed up when i created it maybee yours did too.

I have not gotten a chance to contact tech support yet to find out why.

Merlyn

 

[axvoice]

Most probably it's something on your end. Try registering your adapter or softphone directly. Don't forget to user :9060
PM your username so I can look into this further.

 

[BillG]

I'll give a softphone a try - good idea. I have half a dozen other providers registered with no problems though.

 

[wardmundy]

Be sure there's not a typo in your username or password.

 

[BillG]

Merlyn, make sure you use the site that Ward mentions above to log in - the regular site won't let you.

I can successfully make a call using Eyebeam softphone directly (but not through Asterisk), but I still can't receive a call ("number is not in service" - thanks, Allison.)

 

[wardmundy]

Try ticking the Allow Anonymous SIP Calls option in the FreePBX general setup screen.

 

[BillG]

Nope - still just "Request Sent"