That SIP request was what was received on the FS server.
I just made two calls. One from the polylcom phone which worked and
one through asterisk which failed. I'm comparing the two SIP INVITEs
to see what the difference is. (Looking at the Invites that FS
RECEIVED.)
Much is the same. here is what I found different.
To: On the successful call has "user=phone" appended. Nothing on the
failed call
Call-ID: On the failed call is "abcdefetc:domain" On the successful
call is "abcdetc:phone_ip" <--- Might be significant.
Remote-Party-ID: Exists for failed call. Does not exist for
successful call
Proxy-Authorization: The uri for successful call has port 5060. The
uri for failed call has no port <--- Might be significant.
Still digging through debug stuff but wanted to send this to the list
incase someone had any ideas.
Thanks!!
-Noah
On Sep 22, 2008, at 4:36 PM, Jai Rangi wrote:
Is your asterisk server behind the firewall or NAT.
Does your FS respond to the invite from asterisk,
Did you run sip trace on both asterisk and FS. I mean it will be
useful to know if the below sip trace is from asterisk or FS. If
from asterisk, then need to make sure if FS got the request and how
it replied to that request.
Hope this will help you in debugging the problem.
Jai
www.didforsale.com
*Buy SIP DIDs all Over US at low cost, unlimited minutes
http://www.didforsale.com";
On Mon, Sep 22, 2008 at 4:01 PM, Noah Silverman
<[EMAIL PROTECTED]> wrote:
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:[EMAIL PROTECTED] SIP/2.0.
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK03cd6fd2;rport.
From: "3235551212" <sip:[EMAIL PROTECTED]>;tag=as146a87d7.
To: <sip:[EMAIL PROTECTED]>.
Contact: <sip:[EMAIL PROTECTED]>.
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "3235551212" <sip:
[EMAIL PROTECTED]>;privacy=off;screen=no.
Proxy-Authorization: Digest username="3235551212",
realm="111.111.111.111", algorithm=MD5, uri="sip:[EMAIL PROTECTED]
", 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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