[Freeswitch-users] Reorder tone immediately on Cisco 7960 Line 1 upon off-hook

[A E [Gmail]]

Hi Guys,

I'm pretty sure this is a Cisco firmware problem (v 8.9) problem but just
wanted to check if anyone else has seen this.

With Line 1 on the Cisco 7960 registered to FS, when the line buttons is
pressed or the "Speaker" button is pressed, it immediately gets a re-order
tone instead of a dial-tone. Watching the console on FS, I see an Invite
going to "%14" which obv doesn't exist.

   INVITE sip:%14 at 192.168.3.80 SIP/2.0
   Via: SIP/2.0/UDP 69.201.146.xxx:5062;branch=z9hG4bK121399ed
   From: "user1" <sip:user1 at 192.168.3.80
>;tag=003094c3e08b02705d7d5000-560be9da
   To: <sip:%14 at 192.168.3.80>
   Call-ID: 003094c3-e08b000a-43fa29c5-5d1840e9 at 69.201.146.xxx
   Max-Forwards: 70
   Date: Thu, 01 Sep 2011 16:39:30 GMT
   CSeq: 101 INVITE
   User-Agent: Cisco-CP7960G/8.0
   Contact: <sip:user1 at 69.201.146.xxx:5062;transport=udp>
   Expires: 180
   Accept: application/sdp
   Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
   Supported: replaces,join,norefersub
   Content-Length: 278
   Content-Type: application/sdp
   Content-Disposition: session;handling=optional

   v=0
   o=Cisco-SIPUA 12375 0 IN IP4 69.201.146.xxx
   s=SIP Call
   t=0 0
   m=audio 19982 RTP/AVP 0 8 18 101
   c=IN IP4 69.201.146.xxx
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=sendrecv

When this same account is configured on any other line, pressing the line
button brings up a dial-tone.

Anyone have any idea why it does that or how to make it not do it?

Cheers
aeg
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