Re: [asterisk-users] anveo, a different kind of trunk provider?

2017-01-02 Thread John Kiniston 
On Mon, Jan 2, 2017 at 5:26 AM, Thufir Hawat  wrote:

>
> But, their registration string with Asterisk is:
>
>  Locate [general] secion and add the following
> register => ACCOUNT_NUMBER:sip_passw...@sip.anveo.com:5010
>
>
> Wouldn't this send every outbound call through that Anveo account?
>
>
>
Registration is for Inbound calls to your PBX, Not outbound calls from your
PBX.

Registration lets the remote end know how to contact you when calls are
placed to you and your IP is dynamic.
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[asterisk-users] anveo, a different kind of trunk provider?

2017-01-02 Thread Thufir Hawat 


Anveo has their config as so:

[anveo]
type=friend
host=sip.anveo.com
port=5010
username= ACCOUNT_NUMBER
secret= SIP_PASSWORD
insecure=port,invite
disallow=all
allow=ulaw
context=from-anveo

http://www.anveo.com/faq.asp?code=sip_asterisk



but this seems slightly odd.  I have an account with them where my hard 
phone, an SPA 942 IP phone, connects directly to them.  I just entered the 
SIP details.  Presumably they're running Asterisk and have it configured 
for my SIP account.


But, their registration string with Asterisk is:

 Locate [general] secion and add the following
register => ACCOUNT_NUMBER:sip_passw...@sip.anveo.com:5010


Wouldn't this send every outbound call through that Anveo account?

Let's say that I add a more hard or softphones, but configure them to 
connect to my Asterisk server running on AWS.  When Anveo dials out to the 
POTS everything shows as coming from a single number, the account number?





thanks,

Thufir

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Re: [asterisk-users] Asterisk hep.conf

2017-01-02 Thread Olivier 
Hello,

Reading this thread, may I ask if you could get this to work ?

Regards

2016-06-29 6:29 GMT+02:00 Annus Fictus :

> hello,
>
> I'm trying to use Asterisk 13.9.1 with Homer SIP Capture Server.
>
> My hep.conf Asterisk configuration is:
>
> [general]
> enabled = yes
> capture_address=107.170.151.154:9060
> ;capture_password = foo
> capture_id = 2464
>
> SIP Signaling work correctly but no RTCP STATS arrive to Homer Server. On
> the Asterisk Console, many messages like this:
>
> NOTICE[3739] res_hep.c: Unable to send packet: Address Family mismatch
> between source/destination
>
> Any hint?
>
> Regards
>
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