[Freeswitch-users] Asterisk registration with FS

[Noah Silverman]

Tried that and still doesn't work.

I've attached the SIP INVITE so that maybe you'll see something that  
gives you a clue.

Also, I don't know if it matters, but the FS server is actually in an  
off site data center.  I'm connecting to it remotely from my office.   
Works fine for a single polycom phone.  (That, and the sound quality  
is AMAZING! )

Here is what I have in Asterisk now...
[Freeswitch]
host=111.111.111.111
username=3235551212
secret=password
fromdomain=111.111.111.111
port=5060
type=peer
trustrpid=yes
sendrpid=yes
context=from-trunk
canreinvite=no
disallow=all
allow=ulaw



Here's the SIP INVITE.  (IP's changed to protect the innocent.)
111.111.111.111 is the address of my FS server
222.222.222.222 is the address of my asterisk server
3235551212 is my username/did/account in FS

U 222.222.222.222:1024 -> 111.111.111.111:5060
INVITE sip:13237773456 at 111.111.111.111 SIP/2.0.
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK03cd6fd2;rport.
From: "3235551212" <sip:3235551212 at 111.111.111.111>;tag=as146a87d7.
To: <sip:13237773456 at 111.111.111.111>.
Contact: <sip:3235551212 at 222.222.222.222>.
Call-ID: 4382446d3a46269d7f469e93147b46eb at 111.111.111.111
CSeq: 103 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "3235551212" <sip: 
3235551212 at 111.111.111.111>;privacy=off;screen=no.
Proxy-Authorization: Digest username="3235551212",  
realm="111.111.111.111", algorithm=MD5, uri="sip:13237773456 at 111.111.111.111 
", nonce="c52ba984-f888-dd11-80f4-00188b37805b",  
response="01635afacb7eeebc6fc0888991c0411d", qop=auth,  
cnonce="4e065f27", nc=00000001.
Date: Mon, 22 Sep 2008 22:54:14 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 234.





On Sep 22, 2008, at 2:35 PM, Brian West wrote:

> you'll need to set from-domain in the sip.conf on asterisk ;)
>
> /b
>
> On Sep 22, 2008, at 2:30 PM, Noah Silverman wrote:
>
>> Below is the config in my sip.conf for asterisk.  (IP and DID changed
>> for privacy)
>>
>> [Freeswitch]
>> host=111.111.111.111
>> username=3235551212
>> secret=password
>> port=5060
>> type=peer
>> trustrpid=yes
>> sendrpid=yes
>> context=from-trunk
>> canreinvite=no
>> disallow=all
>> allow=ulaw
>>
>
>
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