Thanks for Response Michael, This is the capture of wireshark INVITE sip:3333@192.168.4.18:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.244:5065;branch=z9hG4bK9268238112493813801;rport From: 30000 <sip:30000@192.168.4.18:5060>;tag=1653316119 To: "3333" <sip:3333@192.168.4.18:5060> Call-ID: 188371401128256-63249236018@192.168.4.244 CSeq: 1 INVITE Contact: <sip:30000@192.168.4.244:5065> Max-Forwards: 70 Supported: replaces, join, path User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 185 v=0 o=30000 24630637 15149730 IN IP4 192.168.4.244 s=A conversation c=IN IP4 192.168.4.244 t=0 0 m=audio 10006 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.4.244:5065;branch=z9hG4bK9268238112493813801;rport=5065 From: 30000 <sip:30000@192.168.4.18:5060>;tag=1653316119 To: "3333" <sip:3333@192.168.4.18:5060>;tag=6te4H15jaF6XF Call-ID: 188371401128256-63249236018@192.168.4.244 CSeq: 1 INVITE Contact: <sip:3333@192.168.4.18:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-23:25M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 191 Remote-Party-ID: "3333" <sip:3333@192.168.4.18>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1337124076 1337124077 IN IP4 192.168.4.18 s=FreeSWITCH c=IN IP4 192.168.4.18 t=0 0 m=audio 23862 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 One more important thing, I noticed. I dowloaded this code from git dec 2012, in this code stop_dtmf is not working as it used to work in previous code, i mean stop_dtmf stop detecting sip_info along with dtmf.(it was downloaded approx 1 year 6 month before). ----- Original Message ----- From: "Michael Collins" <msc@freeswitch.org> To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Sent: Thursday, February 16, 2012 3:27:40 AM Subject: Re: [Freeswitch-users] stop sip_info On Tue, Feb 14, 2012 at 10:27 PM, Piyush Sharma < piyush.sharma@coraltele.com > wrote: Thanks for your response Brian. I am sending you the capture of wireshark. hope it will help to understand the problem, and giving me some solution. Notice he said, "Just the SDP will be fine," but you sent a 283K pcap file! :D If you can just extract out the SDP and paste it into this thread that might suffice. If Brian has only a cell phone or iPad then he's probably not going to be using Wireshark. Don't forget that he's back in the hospital . -MC