SOLVED AXVOICE no incoming, outgoing working properly

[markiper]

Hello Forum,

I am appealing to all you guys out there, as I think I have hit a wall with AXVOICE in regards to the reason of a problem not being able to receive any incoming calls. My AXVOICE configuration is just as the one that Ward mentions on his PiaF 1.3 tutorial, except for the fact that my numbers for some reason are on a differente server (magnum.axvoice.com:9060):

disallow=all
allow=ulaw
authname=yourusername
canreinvite=no
context=all-incoming
defaultip=MAGNUM.AXVOICE.COM
dtmfmode=inband
fromdomain=MAGNUM.AXVOICE.COM
fromuser=yourusername
host=MAGNUM.AXVOICE.COM
insecure=very
nat=yes
PORT=9060
secret=yourpassword
type=friend
user=phone
username=yourusername

This same configuration has been posted on the trixbox forum as being confirmed to be the correct one by AXVOICE (http://trixbox.org/forums/trixbox-forums/sip-and-iax-trunks-and-providers/axvoice-cant-call-out), but for some odd reason I cannot receive any calls and I keep on getting a message that says: "I am sorry, the number you have dialed is no longer is service or it has been disconnect....". Upon calling the support line for AXVOICE (888-277-8647) they cannot even find my accounts on their system . I have tried calling a customer service number provide by the customer support people (212-933-9460) but no one picks up the call. I have had AXVOCE for over a year at this point, and I cannot understand what is going on, other than me changing my server as the older one just died on me. The only other opion that I see is if the MAC ADDRESS on the account plays a role on the authentication process or the validation of the account, which is the only thing that has changed from the old to the new server, and I cannot modify it by login into the AXVOICE portal. Any ideas from all of you out there will be greatly appreciated.

Regards,
markiper

 

[wardmundy]

Send an email to [email protected] and include a copy of your posting above. You might also want to try opening UDP ports from 10000 to 62000 and see if that helps.

 

[markiper]

Thanks Ward,

I have tried opening the ports you mentioned, and no luck. I even tried putting my PiaF box on a public IP without any ports being blocked and still no luck. I will write to them as you mention and hope to get this problem solved.

 

[markiper]

even more confused.....

I have contacted AXVOICE via e-mail, and the answer was not to re-assuring. They ask me to revert to the working configuration in order to perform further testing.... The working configuration was the same as described above but a different server with PiaF version 1.1 on it, now I have a new server with PiaF version 1.3 on it. Digging a little deeper I have been able to see that if I remove the line that reads PORT=9060 from the configuration of the trunk, and i include the port number on the host line (hots=margnum.axvoice.com:9060), I get the INCOMING CALLS working properly, but I kill the OUTGOING CALLS. Know I am really confused about this behavior and what is causing such an strange issue. If anyone in the forum can explain me the proper usage of the expression PORT= on the trunk definition, I will be really thankful for it.

Regards,
markiper

 

[cliff]

Try naming the trunk magnum.axvoice.com.

 

[markiper]

Thanks for the suggestion, Ciff, but it did not work either. I will like to hear some success stories from other people using AXVOICE with PiaF version 1.3 on the MAGNUM.AXVOICE.COM server, as apparently the old server SIP.AXVOICE.COM is working even for Uncle Ward.

Regards,
markiper

 

[cliff]

http://pbxinaflash.com/community/threads/free-accounts-and-minutes-with-axvoice.73/?t=73

http://www.dslreports.com/forum/r19053057-Axvoicecom-beta-service-Invite

 

[rjefferis]

I am using AxVoice, it has been working perfectly for a few months. Took me a while to find working settings.

Here are my configs.

Outgoing
Trunk Name
Axvoice

Peer Details
username=yourusername
type=peer
secret=yourpassword
qualify=yes
port=9060
nat=yes
insecure=port,invite
host=magnum.axvoice.com
dtmfmode=rfc2833
disallow=all
canreinvite=no
allow=ulaw


Incoming
Trunk Name
magnum.axvoice.com

User Details
canreinvite=no
context=from-pstn
host=magnum.axvoice.com
insecure=very
nat=yes
secret=yourpassword
type=peer
username=yourusername


Register String
yourusername:[email protected]:9060


As i said, i just played around for ages until it worked. These have been working great for ages.

Hope they work for you.

ps. the username is the username, not your AxVoice ext no.

 

[markiper]

It worked, thanks rjefferis. The only thing that I had to add is the line FROMUSER=USERNAME in the outgoing settings in order to be able to properly route from multiple AXVOICE accounts that I have registered in my system. This is still very strange, as the configuration I posted was wroking properly until i upraded my server to PiaF version 1.3. The only other consideration I have to chek will be the IPTABLES to make sure that port 9060 is allowed to pass traffic into de PiaF server. Thanks for the inforamtion, now that I have a working configuration I can start looking into what the problems could be with the old one I had.

 

[wardmundy]

With every new version of Asterisk come new surprises.

 

[markiper]

The wonderful world of VoIP. Thanks Ward, at least I have the trunks working properly for the time being. By the way, you mentioned on your last post about a Voip provider that you guys will be working closely with......I hope to hear some news in regards to this pretty soon.

 

[wardmundy]

Send me a note and we'll get you set up to test it.

 

[markiper]

thanks Ward

I did send you a note, hopefully I get to hear from you soon.

 

[anjeleno]

It's been nearly 5 years since there has been any activity on this thread... I've been Googling like crazy and everyone seems to have the same problem with inbound calls on AxVoice. Trunk registers perfectly. Outbound calls work like magic. But I can never get inbound calls to properly route to extensions or ring groups. I only ever get the default IVR (7777). I've tried different combinations of virtually every configuration posted, to no avail.

Does anyone have a definitive config that works with AxVoice?

I just switched from my own custom Virtualbox install to an Amazon EC2 PiaF Green AMI. DNS records pointing to the elastic IP, so I've got custom sipping configured. Everything except inbound routes with AxVoice is running flawlessly!!!

Thanks for the heavy-lifting, Ward!

 

[wardmundy]

Ditch the port 9060 entry. We use the same server without this entry, and it works just fine in both directions.

 

[anjeleno]

Thanks for the followup Ward. I removed the 9060 entry. Still unable to route incoming calls. My current PEER Details are as follows:

allow=ulaw
authname=USERNAME
canreinvite=no
disallow=all
dtmfmode=rfc2833
fromuser=USERNAME
host=magnum.axvoice.com
insecure=very
nat=yes
port=5060 (HAVE TRIED WITH AND WITHOUT THIS LINE)
qualify=yes
secret=SECRET
sendrpid=yes
type=peer
username=USERNAME

Currently, incoming settings are BLANK. Have tried with and without incoming settings and various user context.

Current registration string:
USERNAME:[email protected]
(HAVE TRIED WITH AND WITHOUT SUFFIXING PORT/DID)

The error I'm seeing most often in the Asterisk CLI is:

NOTICE[1624][C-0000003c]: chan_sip.c:25184 handle_request_invite: Call from 'xxxxxx' (69.90.174.98:5060) to extension 'MY DID' rejected because extension not found in context '*VARIOUS_CONTEXT*'.

However, I only get that error preventing AxVoice from reaching PiaF, when I'm playing with custom contexts. Otherwise, AxVoice goes directly to IVR when you dial the DID, but then there is no activity in the CLI... :-/ Well, actually, there's plenty of activity in the CLI, but it looks like some scammer in Germany looking for exploits. fortunately, everything is locked down (except for sip and rtp ports, so I can reach it from the real world), but I use very secure passwords.

Is there a way to manipulate context to force AxVoice to route calls properly? I have played with different contexts without any luck.

All help is appreciated!!!

 

[anjeleno]

After MAJOR frustration and LOADS of caffeine and Googling, I came across this article which shed some additional light on possible issues. A lot of useful info, but I got as far as the 4th or 5th paragraph and discovered the solution to my problem. The leading "1" in my DID was excluded from both the registration string, and outbound caller ID, and from the inbound call route DID number field. I never would've even given it a 2nd thought... Hope this helps someone else!

For anyone else looking for a working AxVoice config that fixes incoming call routing, here's exactly what worked for me 12/16/2013:

(EXCEPT FOR THE SPECIFIED INFO, ALL OTHER VALUES ARE DEFAULT)

SIP TRUNK GENERAL SETTINGS:
Trunk Name: ax-trunk
Outbound Caller ID: 12223334567
Maximum Channels: 2

OUTGOING SETTINGS:
Trunk Name: magnum.axvoice.com (suggested by a few other posts)

PEER DETAILS:
username=USERNAME*
type=peer
secret=PASSWORD*
qualify=yes
port=5060
nat=yes
insecure=port,invite
host=magnum.axvoice.com
dtmfmode=rfc2833
disallow=all
canreinvite=no
allow=ulaw
context=from-trunk

*These are the settings found in the AxVoice.com account portal, under Account/SIP Setting)

INCOMING SETTINGS: LEFT BLANK

REGISTER STRING:
USERNAME:[email protected]/12223334567

INCOMING ROUTE:
Description: from_axvoice
DID Number: 12223334567
Enable Superfecta CID: YES
Destination: extension <7000>

OUTBOUND ROUTE:
Route Name: to_axvoice
Route CID: (LEFT BLANK)
Route Position: I made mine the first route (this is up to you)

Standard Dial Patterns

Trunk Sequence for Matched Routes:
0 = ax_trunk

Carry on...

 

[wardmundy]

Right you are. AxVoice changed from 10-digit to 11-digit numbers about 2 years ago. The Register String is the one that was killing your inbound calls. Thanks.