[Freeswitch-users] Asterisk registration with FS

[Noah Silverman]

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 at 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 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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>
>
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