[Freeswitch-users] stop sip_info

[Piyush Sharma]

Thanks for Response Michael,
This is the capture of wireshark

INVITE sip:3333 at 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 at 192.168.4.18:5060>;tag=1653316119
To: "3333" <sip:3333 at 192.168.4.18:5060>
Call-ID: 188371401128256-63249236018 at 192.168.4.244
CSeq: 1 INVITE
Contact: <sip:30000 at 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 at 192.168.4.18:5060>;tag=1653316119
To: "3333" <sip:3333 at 192.168.4.18:5060>;tag=6te4H15jaF6XF
Call-ID: 188371401128256-63249236018 at 192.168.4.244
CSeq: 1 INVITE
Contact: <sip:3333 at 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 at 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 at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at 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 at 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