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... [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@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.
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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