[Freeswitch-users] Asterisk registration with FS

Jai Rangi jprangi at gmail.com
Mon Sep 22 16:36:45 PDT 2008


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%3A13237773456 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<sip%3A3235551212 at 111.111.111.111>
> >;tag=as146a87d7.
> To: <sip:13237773456 at 111.111.111.111 <sip%3A13237773456 at 111.111.111.111>>.
> Contact: <sip:3235551212 at 222.222.222.222<sip%3A3235551212 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 <sip%3A13237773456 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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