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
<noah@allresearch.com> 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...
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@111.111.111.111 SIP/2.0.
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK03cd6fd2;rport.
From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as146a87d7.
To: <sip:13237773456@111.111.111.111>.
Contact: <sip:3235551212@222.222.222.222>.
Call-ID: 4382446d3a46269d7f469e93147b46eb@111.111.111.111
CSeq: 103 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "3235551212" <sip:
3235551212@111.111.111.111>;privacy=off;screen=no.
Proxy-Authorization: Digest username="3235551212",
realm="111.111.111.111", algorithm=MD5, uri="sip:13237773456@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.
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