Hi ,<br><br>I am attaching the sip debug messages enabled on asterisk
server .the present status is that when i call from Mirial softphone
,the call gets established and gets disconnected after a while .i can't
see video on mcuWeb page .i think i have done all things correctly but i
am stuck with this thing .please help me out of this problem .<br><br>INVITE <a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=<div id=":188">z9hG4bK504e1a07<br>Max-Forwards: 70<br>From: "205" <<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=as03ab7ca9<br>
To: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>><br>
Contact: <<a href="http://sip:205@192.168.115.53:5060/" target="_blank">sip:205@192.168.115.53:5060</a>><br>Call-ID: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 1.8.6.0<br>Date: Tue, 11 Oct 2011 06:54:33 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>
Content-Length: 363<br><br>v=0<br>o=root 1739466419 1739466419 IN IP4 192.168.115.53<br>s=Asterisk PBX 1.8.6.0<br>c=IN IP4 192.168.115.53<br>b=CT:384<br>t=0 0<br>m=audio 14856 RTP/AVP 0 8 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br>m=video 15692 RTP/AVP 99<br>a=rtpmap:99 H264/90000<br>a=sendrecv<br><br>---<br><br><--- SIP read from UDP:<a href="http://192.168.115.29:5060/" target="_blank">192.168.115.29:5060</a> ---><br>
SIP/2.0 100 Trying<br>Content-Length: 0<br>To: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>><br>Cseq: 102 INVITE<br>Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07<br>
Server: Glassfish_SIP_1.0.0<br>
Call-Id: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>From: "205" <<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=as03ab7ca9<br>
<br><-------------><br>--- (8 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://192.168.115.29:5060/" target="_blank">192.168.115.29:5060</a> ---><br>SIP/2.0 180 Ringing<br>Content-Length: 0<br>
To: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>>;tag=gtmjcomk-b<br>
Contact: <sip:192.168.115.29:5070;fid=server_1><br>Cseq: 102 INVITE<br>Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07<br>From: "205"<<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=as03ab7ca9<br>
Call-Id: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>Server: Glassfish_SIP_1.0.0<br><br><-------------><br>--- (9 headers 0 lines) ---<br>
<br><--- Transmitting (no NAT) to <a href="http://192.168.115.40:5060/" target="_blank">192.168.115.40:5060</a> ---><br>SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 192.168.115.40:5060;branch=z9hG4bK-d7358366c5-DL;received=192.168.115.40<br>
From: "205" <<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=DLb62dc55e0d;epid=021776A8<br>To: <<a href="mailto:sip%3A300@192.168.115.53" target="_blank">sip:300@192.168.115.53</a>;user=phone>;tag=as6ba67fa9<br>
Call-ID: DLd873cc4431-1078986205@fstl-312<br>CSeq: 1 INVITE<br>Server: Asterisk PBX 1.8.6.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Contact: <<a href="http://sip:300@192.168.115.53:5060/" target="_blank">sip:300@192.168.115.53:5060</a>><br>
Content-Length: 0<br><br><br><------------><br><br><--- SIP read from UDP:<a href="http://192.168.115.29:5060/" target="_blank">192.168.115.29:5060</a> ---><br>SIP/2.0 200 Ok<br>Content-Length: 245<br>To: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>>;tag=gtmjcomk-b<br>
Contact: <sip:192.168.115.29:5070;fid=server_1><br>Cseq: 102 INVITE<br>Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07<br>Content-Type: application/sdp<br>From: "205"<<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=as03ab7ca9<br>
Call-Id: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>Server: Glassfish_SIP_1.0.0<br><br>v=0<br>o=- 0 0 IN IP4 192.168.115.29<br>
s=MediaMixerSession<br>
c=IN IP4 192.168.115.29<br>t=0 0<br>m=audio 60014 RTP/AVP 0 8<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>m=video 35814 RTP/AVP 99<br>a=rtpmap:99 H264/90000<br>a=fmtp:99 profile-level-id=428021<br><-------------><br>
--- (10 headers 11 lines) ---<br>Found RTP audio format 0<br>Found RTP audio format 8<br>Found audio description format PCMU for ID 0<br>Found audio description format PCMA for ID 8<br>Found RTP video format 99<br>Found video description format H264 for ID 99<br>
Capabilities: us - 0x30000c (ulaw|alaw|h263p|h264), peer - audio=0xc
(ulaw|alaw)/video=0x200000 (h264)/text=0x0 (nothing), combined -
0x20000c (ulaw|alaw|h264)<br>Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)<br>
Peer audio RTP is at port <a href="http://192.168.115.29:60014/" target="_blank">192.168.115.29:60014</a><br>Peer video RTP is at port <a href="http://192.168.115.29:35814/" target="_blank">192.168.115.29:35814</a><br>list_route: hop: <sip:192.168.115.29:5070;fid=server_1><br>
set_destination: Parsing <sip:192.168.115.29:5070;fid=server_1> for address/port to send to<br>set_destination: set destination to <a href="http://192.168.115.29:5070/" target="_blank">192.168.115.29:5070</a><br>Transmitting (no NAT) to <a href="http://192.168.115.29:5070/" target="_blank">192.168.115.29:5070</a>:<br>
ACK sip:192.168.115.29:5070;fid=server_1 SIP/2.0<br>Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK07561f41<br>Max-Forwards: 70<br>From: "205" <<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=as03ab7ca9<br>
To: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>>;tag=gtmjcomk-b<br>Contact: <<a href="http://sip:205@192.168.115.53:5060/" target="_blank">sip:205@192.168.115.53:5060</a>><br>
Call-ID: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>
CSeq: 102 ACK<br>User-Agent: Asterisk PBX 1.8.6.0<br>Content-Length: 0<br><br><br>---<br>Audio is at 5060<br>Video is at <a href="http://192.168.115.53:5060/" target="_blank">192.168.115.53:5060</a><br>Adding codec 0x4 (ulaw) to SDP<br>
Adding codec 0x8 (alaw) to SDP<br>
Adding video codec 0x200000 (h264) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br><br><--- Reliably Transmitting (no NAT) to <a href="http://192.168.115.40:5060/" target="_blank">192.168.115.40:5060</a> ---><br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 192.168.115.40:5060;branch=z9hG4bK-d7358366c5-DL;received=192.168.115.40<br>From: "205" <<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=DLb62dc55e0d;epid=021776A8<br>
To: <<a href="mailto:sip%3A300@192.168.115.53" target="_blank">sip:300@192.168.115.53</a>;user=phone>;tag=as6ba67fa9<br>Call-ID: DLd873cc4431-1078986205@fstl-312<br>CSeq: 1 INVITE<br>Server: Asterisk PBX 1.8.6.0<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>Contact: <<a href="http://sip:300@192.168.115.53:5060/" target="_blank">sip:300@192.168.115.53:5060</a>><br>Content-Type: application/sdp<br>Content-Length: 361<br><br>v=0<br>o=root 797835425 797835425 IN IP4 192.168.115.53<br>
s=Asterisk PBX 1.8.6.0<br>c=IN IP4 192.168.115.53<br>b=CT:384<br>t=0 0<br>m=audio 10346 RTP/AVP 0 8 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>
a=ptime:20<br>a=sendrecv<br>m=video 12196 RTP/AVP 99<br>a=rtpmap:99 H264/90000<br>a=sendrecv<br><br><------------><br> -- Locally bridging SIP/205-00000002 and SIP/mcuWeb-00000003<br><br><--- SIP read from UDP:<a href="http://192.168.115.40:5060/" target="_blank">192.168.115.40:5060</a> ---><br>
ACK <a href="http://sip:300@192.168.115.53:5060/" target="_blank">sip:300@192.168.115.53:5060</a> SIP/2.0<br>CSeq: 1 ACK<br>Via: SIP/2.0/UDP 192.168.115.40:5060;branch=z9hG4bK-818b523671-DL<br>To: <<a href="mailto:sip%3A300@192.168.115.53" target="_blank">sip:300@192.168.115.53</a>;user=phone>;tag=as6ba67fa9<br>
From: "205" <<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=DLb62dc55e0d;epid=021776A8<br>Call-ID: DLd873cc4431-1078986205@fstl-312<br>Max-Forwards: 70<br>Contact: "205" <<a href="http://sip:205@192.168.115.40:5060/" target="_blank">sip:205@192.168.115.40:5060</a>><br>
Content-Length: 0<br><br><-------------><br>--- (9 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://192.168.115.29:5060/" target="_blank">192.168.115.29:5060</a> ---><br>SIP/2.0 200 Ok<br>Content-Length: 245<br>
To: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>>;tag=gtmjcomk-b<br>Contact: <sip:192.168.115.29:5070;fid=server_1><br>Cseq: 102 INVITE<br>Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07<br>
Content-Type: application/sdp<br>From: "205"<<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=as03ab7ca9<br>Call-Id: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>
Server: Glassfish_SIP_1.0.0<br><br>v=0<br>o=- 0 0 IN IP4 192.168.115.29<br>s=MediaMixerSession<br>c=IN IP4 192.168.115.29<br>t=0 0<br>m=audio 60014 RTP/AVP 0 8<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>m=video 35814 RTP/AVP 99<br>
a=rtpmap:99 H264/90000<br>a=fmtp:99 profile-level-id=428021<br><-------------><br>--- (10 headers 11 lines) ---<br>set_destination: Parsing <sip:192.168.115.29:5070;fid=server_1> for address/port to send to<br>
set_destination: set destination to <a href="http://192.168.115.29:5070/" target="_blank">192.168.115.29:5070</a><br>Transmitting (no NAT) to <a href="http://192.168.115.29:5070/" target="_blank">192.168.115.29:5070</a>:<br>
ACK sip:192.168.115.29:5070;fid=server_1 SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK4559c232<br>Max-Forwards: 70<br>From: "205" <<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=as03ab7ca9<br>To: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>>;tag=gtmjcomk-b<br>
Contact: <<a href="http://sip:205@192.168.115.53:5060/" target="_blank">sip:205@192.168.115.53:5060</a>><br>Call-ID: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>
CSeq: 102 ACK<br>User-Agent: Asterisk PBX 1.8.6.0<br>Content-Length: 0<br><br><br><--- SIP read from UDP:<a href="http://192.168.115.29:5060/" target="_blank">192.168.115.29:5060</a> ---><br>BYE <a href="http://sip:205@192.168.115.53:5060/" target="_blank">sip:205@192.168.115.53:5060</a> SIP/2.0<br>
Max-Forwards: 69<br>Content-Length: 0<br>To: "205"<<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;<div id=":188">tag=as03ab7ca9<br>Contact: <sip:192.168.115.29:5070;fid=server_1><br>
Cseq: 2 BYE<br>
Via: SIP/2.0/UDP 192.168.115.29:5070;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01<br>From: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>>;tag=gtmjcomk-b<br>Call-Id: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>
<br><-------------><br>--- (9 headers 0 lines) ---<br>Sending to <a href="http://192.168.115.29:5070/" target="_blank">192.168.115.29:5070</a> (no NAT)<br>Scheduling destruction of SIP dialog '<a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a>' in 32000 ms (Method: BYE)<br>
<br><--- Transmitting (no NAT) to <a href="http://192.168.115.29:5070/" target="_blank">192.168.115.29:5070</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.115.29:5070;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01;received=192.168.115.29<br>
From: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>>;tag=gtmjcomk-b<br>To: "205"<<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=as03ab7ca9<br>
Call-ID: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>CSeq: 2 BYE<br>Server: Asterisk PBX 1.8.6.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>Content-Length: 0<br><br><br><------------><br>Scheduling destruction of SIP dialog 'DLd873cc4431-1078986205@fstl-312' in 32000 ms (Method: OPTIONS)<br>set_destination: Parsing <<a href="http://sip:205@192.168.115.40:5060/" target="_blank">sip:205@192.168.115.40:5060</a>> for address/port to send to<br>
set_destination: set destination to <a href="http://192.168.115.40:5060/" target="_blank">192.168.115.40:5060</a><br>Reliably Transmitting (no NAT) to <a href="http://192.168.115.40:5060/" target="_blank">192.168.115.40:5060</a>:<br>
BYE <a href="http://sip:205@192.168.115.40:5060/" target="_blank">sip:205@192.168.115.40:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK22de4c9c<br>Max-Forwards: 70<br>From: <<a href="mailto:sip%3A300@192.168.115.53" target="_blank">sip:300@192.168.115.53</a>;user=phone>;tag=as6ba67fa9<br>To: "205" <<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=DLb62dc55e0d;epid=021776A8<br>
Call-ID: DLd873cc4431-1078986205@fstl-312<br>CSeq: 102 BYE<br>User-Agent: Asterisk PBX 1.8.6.0<br>X-Asterisk-HangupCause: Normal Clearing<br>X-Asterisk-HangupCauseCode: 16<br>Content-Length: 0<br><br><br>---<br><br><--- SIP read from UDP:<a href="http://192.168.115.40:5060/" target="_blank">192.168.115.40:5060</a> ---><br>
SIP/2.0 200 OK<br>From: <<a href="mailto:sip%3A300@192.168.115.53" target="_blank">sip:300@192.168.115.53</a>;user=phone>;tag=as6ba67fa9;epid=021776A8<br>Call-ID: DLd873cc4431-1078986205@fstl-312<br>CSeq: 102 BYE<br>
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK22de4c9c;received=192.168.115.53<br>
To: "205" <<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=DLb62dc55e0d;epid=021776A8<br>Allow: INVITE,CANCEL,ACK,OPTIONS,INFO,SUBSCRIBE,NOTIFY,BYE,MESSAGE,UPDATE,REFER<br>
Contact: "205" <<a href="http://sip:205@192.168.115.40:5060/" target="_blank">sip:205@192.168.115.40:5060</a>><br>
User-Agent: Dylogic Mirial 7.0.54<br>Content-Length: 0<br><br><-------------><br>--- (10 headers 0 lines) ---<br>Really destroying SIP dialog 'DLd873cc4431-1078986205@fstl-312' Method: OPTIONS<br><br><--- SIP read from UDP:<a href="http://192.168.115.29:5060/" target="_blank">192.168.115.29:5060</a> ---><br>
BYE <a href="http://sip:205@192.168.115.53:5060/" target="_blank">sip:205@192.168.115.53:5060</a> SIP/2.0<br>Max-Forwards: 69<br>Content-Length: 0<br>Contact: <sip:192.168.115.29:5070;fid=server_1><br>To: "205"<<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=as03ab7ca9<br>
Cseq: 2 BYE<br>Via: SIP/2.0/UDP 192.168.115.29:5070;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01<br>Call-Id: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>
From: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>>;tag=gtmjcomk-b<br><br><-------------><br>--- (9 headers 0 lines) ---<br>Sending to <a href="http://192.168.115.29:5070/" target="_blank">192.168.115.29:5070</a> (no NAT)<br>
<br><--- Transmitting (no NAT) to <a href="http://192.168.115.29:5070/" target="_blank">192.168.115.29:5070</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.115.29:5070;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01;received=192.168.115.29<br>
From: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>>;tag=gtmjcomk-b<br>To: "205"<<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=as03ab7ca9<br>
Call-ID: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>CSeq: 2 BYE<br>Server: Asterisk PBX 1.8.6.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>Content-Length: 0<br><br><br><------------><br><br><--- SIP read from UDP:<a href="http://192.168.115.29:5060/" target="_blank">192.168.115.29:5060</a> ---><br>BYE <a href="http://sip:205@192.168.115.53:5060/" target="_blank">sip:205@192.168.115.53:5060</a> SIP/2.0<br>
Max-Forwards: 69<br>Content-Length: 0<br>Contact: <sip:192.168.115.29:5070;fid=server_1><br>To: "205"<<a href="mailto:sip%3A205@192.168.115.53" target="_blank">sip:205@192.168.115.53</a>>;tag=as03ab7ca9<br>
Cseq: 2 BYE<br>
Via: SIP/2.0/UDP 192.168.115.29:5070;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01<br>Call-Id: <a href="http://687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060/" target="_blank">687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060</a><br>
From: <<a href="http://sip:300@192.168.115.29:5060/" target="_blank">sip:300@192.168.115.29:5060</a>>;tag=gtmjcomk-b</div><br><br><br></div><br>