<div dir="ltr"><div><div><div>We have been looking at that all day, but cant figure out the issue. Funny thing is that its happening only when GS Originate the call. May be we are over looking something. Here are two call example. IPs are modified for security. <br><br>FreeSWITCH (Version 1.6.6 64bit) is ready<br>freeswitch@internal> sofia_contact <a href="mailto:1276@domain.company.com">1276@domain.company.com</a><br>sofia/internal/sip:1276@192.168.1.168:12113;fs_nat=yes;fs_path=sip%3A1276%4068.5.194.163%3A12113<br>freeswitch@internal> sofia_contact <a href="mailto:142@domain.company.com">142@domain.company.com</a><br>sofia/internal/sip:142@172.16.42.13:11852;fs_nat=yes;fs_path=sip%3A142%4074.67.200.39%3A33812<br><br><br></div>142 calls 1276 (1276 does not hear anything) (Seems freeswitch not handling nat properly) On TCP dump, I can see free switching receiving the RTP packet, but trying to deliver to local IP for 1276. <br></div>1276 calls 142 (All good both parties can hear)<br></div>142 call PSTN number (All good)<br><div><div><div>1272 call PSTN number (All good) <br><br></div><div>Same configuration, same dialplan works just fine with 1.6.2 and 1.4.18. 1.6.2 had intermittent DTMF issue, we upgraded to 1.6.5, found this one way audio, upgraded to 1.6.6. <br></div><div>We have narrowed it down to Grand Stream Softphone and Grad Stream IP phones. <br></div><div>Here is trace for Both calls. <br></div><div><br></div><div>######################################<br>U <a href="http://74.67.200.39:33812">74.67.200.39:33812</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>INVITE <a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a> SIP/2.0.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport.<br>From: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>;tag=479799221.<br>To: <<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>>.<br>Call-ID: <a href="mailto:152039027-11852-15@BHC.BG.EC.BD">152039027-11852-15@BHC.BG.EC.BD</a>.<br>CSeq: 140 INVITE.<br>Contact: "142" <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>.<br>Max-Forwards: 70.<br>User-Agent: Grandstream Wave/IOS 1.0.19.<br>Privacy: none.<br>P-Preferred-Identity: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>.<br>Supported: replaces, path, timer, eventlist.<br>Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.<br>Content-Type: application/sdp.<br>Accept: application/sdp, application/dtmf-relay.<br>Content-Length: 235.<br>.<br>v=0.<br>o=142 8000 8000 IN IP4 172.16.42.13.<br>s=SIP Call.<br>c=IN IP4 172.16.42.13.<br>t=0 0.<br>m=audio 50476 RTP/AVP 0 8 101.<br>a=sendrecv.<br>a=rtpmap:0 PCMU/8000.<br>a=ptime:20.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-15.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://74.67.200.39:33812">74.67.200.39:33812</a><br>SIP/2.0 100 Trying.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport=33812;received=74.67.200.39.<br>From: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>;tag=479799221.<br>To: <<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>>.<br>Call-ID: <a href="mailto:152039027-11852-15@BHC.BG.EC.BD">152039027-11852-15@BHC.BG.EC.BD</a>.<br>CSeq: 140 INVITE.<br>User-Agent: VOIPGATEWAY.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://74.67.200.39:33812">74.67.200.39:33812</a><br>SIP/2.0 407 Proxy Authentication Required.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport=33812;received=74.67.200.39.<br>From: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>;tag=479799221.<br>To: <<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>>;tag=Kt2jU0QN57N8F.<br>Call-ID: <a href="mailto:152039027-11852-15@BHC.BG.EC.BD">152039027-11852-15@BHC.BG.EC.BD</a>.<br>CSeq: 140 INVITE.<br>User-Agent: VOIPGATEWAY.<br>Accept: application/sdp.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.<br>Proxy-Authenticate: Digest realm="<a href="http://domain.company.com">domain.company.com</a>", nonce="5eb04922-17e3-426e-9ddf-4333692102b5", algorithm=MD5, qop="auth".<br>Content-Length: 0.<br>.<br><br>##<br>U <a href="http://74.67.200.39:33812">74.67.200.39:33812</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>ACK <a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a> SIP/2.0.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport.<br>From: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>;tag=479799221.<br>To: <<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>>;tag=Kt2jU0QN57N8F.<br>Call-ID: <a href="mailto:152039027-11852-15@BHC.BG.EC.BD">152039027-11852-15@BHC.BG.EC.BD</a>.<br>CSeq: 140 ACK.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://74.67.200.39:33812">74.67.200.39:33812</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>INVITE <a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a> SIP/2.0.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK322043518;rport.<br>From: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>;tag=479799221.<br>To: <<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>>.<br>Call-ID: <a href="mailto:152039027-11852-15@BHC.BG.EC.BD">152039027-11852-15@BHC.BG.EC.BD</a>.<br>CSeq: 141 INVITE.<br>Contact: "142" <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>.<br>Proxy-Authorization: Digest username="xxxxx", realm="<a href="http://domain.company.com">domain.company.com</a>", nonce="5eb04922-17e3-426e-9ddf-4333692102b5", uri="<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>", response="d309cb76b83042023f6794835ad60a89", algorithm=MD5, cnonce="15121380", qop=auth, nc=00000003.<br>Max-Forwards: 70.<br>User-Agent: Grandstream Wave/IOS 1.0.19.<br>Privacy: none.<br>P-Preferred-Identity: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>.<br>Supported: replaces, path, timer, eventlist.<br>Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.<br>Content-Type: application/sdp.<br>Accept: application/sdp, application/dtmf-relay.<br>Content-Length: 235.<br>.<br>v=0.<br>o=142 8000 8000 IN IP4 172.16.42.13.<br>s=SIP Call.<br>c=IN IP4 172.16.42.13.<br>t=0 0.<br>m=audio 50476 RTP/AVP 0 8 101.<br>a=sendrecv.<br>a=rtpmap:0 PCMU/8000.<br>a=ptime:20.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-15.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://74.67.200.39:33812">74.67.200.39:33812</a><br>SIP/2.0 100 Trying.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39.<br>From: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>;tag=479799221.<br>To: <<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>>.<br>Call-ID: <a href="mailto:152039027-11852-15@BHC.BG.EC.BD">152039027-11852-15@BHC.BG.EC.BD</a>.<br>CSeq: 141 INVITE.<br>User-Agent: VOIPGATEWAY.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://68.5.194.163:12113">68.5.194.163:12113</a><br>INVITE <a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a> SIP/2.0.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKX5H9FDZX1eyNN.<br>Route: <<a href="http://sip:1276@68.5.194.163:12113">sip:1276@68.5.194.163:12113</a>>.<br>Max-Forwards: 68.<br>From: "Softphone" <<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;tag=NcN4ypSvZS2DQ.<br>To: <<a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a>>.<br>Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243919 INVITE.<br>Contact: <<a href="http://sip:142@222.222.222.222:5060">sip:142@222.222.222.222:5060</a>>.<br>User-Agent: VOIPGATEWAY.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 270.<br>X-FS-Support: update_display,send_info.<br>.<br>v=0.<br>o=FreeSWITCH 1455142324 1455142325 IN IP4 222.222.222.222.<br>s=FreeSWITCH.<br>c=IN IP4 222.222.222.222.<br>t=0 0.<br>m=audio 17642 RTP/AVP 0 8 101 13.<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=rtpmap:13 CN/8000.<br>a=ptime:20.<br><br>#<br>#<br>U <a href="http://68.5.194.163:12113">68.5.194.163:12113</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>SIP/2.0 100 Trying.<br>From: Softphone<<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;tag=NcN4ypSvZS2DQ.<br>To: <a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a>.<br>Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243919 INVITE.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN.<br>Content-Length: 0.<br>.<br><br>###<br>U <a href="http://68.5.194.163:12113">68.5.194.163:12113</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>SIP/2.0 180 Ringing.<br>From: Softphone<<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;tag=NcN4ypSvZS2DQ.<br>To: sip:1276@192.168.1.168:12113;tag=ReKf2-xIWCt.<br>Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243919 INVITE.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN.<br>Contact: ATAPHONE<<a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a>>.<br>User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://74.67.200.39:33812">74.67.200.39:33812</a><br>SIP/2.0 183 Session Progress.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39.<br>From: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>;tag=479799221.<br>To: <<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>>;tag=m3UBXU8r2gcUB.<br>Call-ID: <a href="mailto:152039027-11852-15@BHC.BG.EC.BD">152039027-11852-15@BHC.BG.EC.BD</a>.<br>CSeq: 141 INVITE.<br>Contact: <sip:1276@222.222.222.222:5060;transport=udp>.<br>User-Agent: VOIPGATEWAY.<br>Accept: application/sdp.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 222.<br>P-Asserted-Identity: "1276" <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>.<br>.<br>v=0.<br>o=FreeSWITCH 1455132056 1455132057 IN IP4 222.222.222.222.<br>s=FreeSWITCH.<br>c=IN IP4 222.222.222.222.<br>t=0 0.<br>m=audio 27910 RTP/AVP 0 101.<br>a=rtpmap:0 PCMU/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>##############<br>##<br>#######<br>U <a href="http://68.5.194.163:12113">68.5.194.163:12113</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>SIP/2.0 200 OK.<br>From: Softphone<<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;tag=NcN4ypSvZS2DQ.<br>To: sip:1276@192.168.1.168:12113;tag=ReKf2-xIWCt.<br>Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243919 INVITE.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN.<br>Contact: ATAPHONE<<a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a>>.<br>User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.<br>Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER.<br>Supported: timer,replaces.<br>Content-Type: application/sdp.<br>Content-Length: 207.<br>.<br>v=0.<br>o=1276 87748 1 IN IP4 192.168.1.168.<br>s=-.<br>c=IN IP4 192.168.1.168.<br>t=0 0.<br>m=audio 8000 RTP/AVP 0 96.<br>a=rtpmap:0 PCMU/8000.<br>a=rtpmap:96 telephone-event/8000.<br>a=ptime:20.<br>a=rtpmap:96 telephone-event/8000.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://68.5.194.163:12113">68.5.194.163:12113</a><br>ACK <a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a> SIP/2.0.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKZQ4tK304U0aUc.<br>Max-Forwards: 70.<br>From: "Softphone" <<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;tag=NcN4ypSvZS2DQ.<br>To: <<a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a>>;tag=ReKf2-xIWCt.<br>Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243919 ACK.<br>Contact: <<a href="http://sip:142@222.222.222.222:5060">sip:142@222.222.222.222:5060</a>>.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://74.67.200.39:33812">74.67.200.39:33812</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39.<br>From: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>;tag=479799221.<br>To: <<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>>;tag=m3UBXU8r2gcUB.<br>Call-ID: <a href="mailto:152039027-11852-15@BHC.BG.EC.BD">152039027-11852-15@BHC.BG.EC.BD</a>.<br>CSeq: 141 INVITE.<br>Contact: <sip:1276@222.222.222.222:5060;transport=udp>.<br>User-Agent: VOIPGATEWAY.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 222.<br>P-Asserted-Identity: "Outbound Call" <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>.<br>.<br>v=0.<br>o=FreeSWITCH 1455132056 1455132057 IN IP4 222.222.222.222.<br>s=FreeSWITCH.<br>c=IN IP4 222.222.222.222.<br>t=0 0.<br>m=audio 27910 RTP/AVP 0 101.<br>a=rtpmap:0 PCMU/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>##<br>U <a href="http://74.67.200.39:33812">74.67.200.39:33812</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>ACK sip:1276@222.222.222.222:5060;transport=udp SIP/2.0.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1657841580;rport.<br>From: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>;tag=479799221.<br>To: <<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>>;tag=m3UBXU8r2gcUB.<br>Call-ID: <a href="mailto:152039027-11852-15@BHC.BG.EC.BD">152039027-11852-15@BHC.BG.EC.BD</a>.<br>CSeq: 141 ACK.<br>Contact: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>.<br>Proxy-Authorization: Digest username="xxxx", realm="<a href="http://domain.company.com">domain.company.com</a>", nonce="5eb04922-17e3-426e-9ddf-4333692102b5", uri="<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>", response="d309cb76b83042023f6794835ad60a89", algorithm=MD5, cnonce="15121380", qop=auth, nc=00000003.<br>Max-Forwards: 70.<br>Supported: replaces, path, timer, eventlist.<br>User-Agent: Grandstream Wave/IOS 1.0.19.<br>Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.<br>Content-Length: 0.<br>.<br><br>###############################################################<br>####<br>#<br><br>#<br>#########<br>################################################################################################<br>U <a href="http://74.67.200.39:33812">74.67.200.39:33812</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>BYE sip:1276@222.222.222.222:5060;transport=udp SIP/2.0.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK326133779;rport.<br>From: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>;tag=479799221.<br>To: <<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>>;tag=m3UBXU8r2gcUB.<br>Call-ID: <a href="mailto:152039027-11852-15@BHC.BG.EC.BD">152039027-11852-15@BHC.BG.EC.BD</a>.<br>CSeq: 142 BYE.<br>Contact: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>.<br>Max-Forwards: 70.<br>Supported: replaces, path, timer, eventlist.<br>User-Agent: Grandstream Wave/IOS 1.0.19.<br>Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://74.67.200.39:33812">74.67.200.39:33812</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK326133779;rport=33812;received=74.67.200.39.<br>From: "142" <<a href="http://sip:142@domain.company.com:5060">sip:142@domain.company.com:5060</a>>;tag=479799221.<br>To: <<a href="http://sip:1276@domain.company.com:5060">sip:1276@domain.company.com:5060</a>>;tag=m3UBXU8r2gcUB.<br>Call-ID: <a href="mailto:152039027-11852-15@BHC.BG.EC.BD">152039027-11852-15@BHC.BG.EC.BD</a>.<br>CSeq: 142 BYE.<br>User-Agent: VOIPGATEWAY.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Content-Length: 0.<br>.<br><br>###<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://68.5.194.163:12113">68.5.194.163:12113</a><br>BYE <a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a> SIP/2.0.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK2jg5rmKFKUDKF.<br>Max-Forwards: 70.<br>From: "Softphone" <<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;tag=NcN4ypSvZS2DQ.<br>To: <<a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a>>;tag=ReKf2-xIWCt.<br>Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243920 BYE.<br>User-Agent: VOIPGATEWAY.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Reason: Q.850;cause=16;text="NORMAL_CLEARING".<br>Content-Length: 0.<br>.<br><br>###<br>U <a href="http://68.5.194.163:12113">68.5.194.163:12113</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>SIP/2.0 200 OK.<br>From: Softphone<<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;tag=NcN4ypSvZS2DQ.<br>To: sip:1276@192.168.1.168:12113;tag=ReKf2-xIWCt.<br>Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243920 BYE.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK2jg5rmKFKUDKF.<br>User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.<br>Content-Length: 0.<br>.<br><br>#######################################################<br>#########<br>#<br><br>Successful Call with 2 way audio :1276 calls 142 <br><br>#<br>#<br>#<br>##################################<br>U <a href="http://68.5.194.163:12113">68.5.194.163:12113</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>INVITE <a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a> SIP/2.0.<br>From: ATAPHONE<<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=VxLf2-dmKe50.<br>To: <a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>.<br>Call-ID: <a href="mailto:d44p80-lYfd5f2@domain.company.com">d44p80-lYfd5f2@domain.company.com</a>.<br>CSeq: 176 INVITE.<br>Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH.<br>Contact: ATAPHONE<<a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a>>.<br>Max-Forwards: 70.<br>User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.<br>Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER.<br>Supported: timer,replaces.<br>Content-Type: application/sdp.<br>Content-Length: 257.<br>.<br>v=0.<br>o=1276 87384 1 IN IP4 192.168.1.168.<br>s=-.<br>c=IN IP4 192.168.1.168.<br>t=0 0.<br>m=audio 8002 RTP/AVP 0 18 8 96.<br>a=rtpmap:0 PCMU/8000.<br>a=rtpmap:18 G729/8000.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:96 telephone-event/8000.<br>a=ptime:20.<br>a=rtpmap:96 telephone-event/8000.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://68.5.194.163:12113">68.5.194.163:12113</a><br>SIP/2.0 100 Trying.<br>Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH;received=68.5.194.163.<br>From: ATAPHONE <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=VxLf2-dmKe50.<br>To: <<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>.<br>Call-ID: <a href="mailto:d44p80-lYfd5f2@domain.company.com">d44p80-lYfd5f2@domain.company.com</a>.<br>CSeq: 176 INVITE.<br>User-Agent: VOIPGATEWAY.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://68.5.194.163:12113">68.5.194.163:12113</a><br>SIP/2.0 407 Proxy Authentication Required.<br>Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH;received=68.5.194.163.<br>From: ATAPHONE <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=VxLf2-dmKe50.<br>To: <<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;tag=SjctQ6jcUQD5a.<br>Call-ID: <a href="mailto:d44p80-lYfd5f2@domain.company.com">d44p80-lYfd5f2@domain.company.com</a>.<br>CSeq: 176 INVITE.<br>User-Agent: VOIPGATEWAY.<br>Accept: application/sdp.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.<br>Proxy-Authenticate: Digest realm="<a href="http://domain.company.com">domain.company.com</a>", nonce="1cf240ca-c5d1-4ac0-8417-becd41784ffb", algorithm=MD5, qop="auth".<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://68.5.194.163:12113">68.5.194.163:12113</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>ACK <a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a> SIP/2.0.<br>From: ATAPHONE<<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=VxLf2-dmKe50.<br>To: <a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>;tag=SjctQ6jcUQD5a.<br>Call-ID: <a href="mailto:d44p80-lYfd5f2@domain.company.com">d44p80-lYfd5f2@domain.company.com</a>.<br>CSeq: 176 ACK.<br>Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://68.5.194.163:12113">68.5.194.163:12113</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>INVITE <a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a> SIP/2.0.<br>From: ATAPHONE<<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=VxLf2-dmKe50.<br>To: <a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>.<br>Call-ID: <a href="mailto:d44p80-lYfd5f2@domain.company.com">d44p80-lYfd5f2@domain.company.com</a>.<br>CSeq: 177 INVITE.<br>Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-l5MoOcq0.<br>Contact: ATAPHONE<<a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a>>.<br>Max-Forwards: 70.<br>User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.<br>Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER.<br>Supported: timer,replaces.<br>Proxy-Authorization: Digest username="yyyy",realm="<a href="http://domain.company.com">domain.company.com</a>",uri="<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>",response="a87ca168b1869994a6b4782df7bffe99",algorithm=MD5,nonce="1cf240ca-c5d1-4ac0-8417-becd41784ffb",qop=auth,cnonce="00017214",nc=00000001.<br>Content-Type: application/sdp.<br>Content-Length: 257.<br>.<br>v=0.<br>o=1276 87384 1 IN IP4 192.168.1.168.<br>s=-.<br>c=IN IP4 192.168.1.168.<br>t=0 0.<br>m=audio 8002 RTP/AVP 0 18 8 96.<br>a=rtpmap:0 PCMU/8000.<br>a=rtpmap:18 G729/8000.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:96 telephone-event/8000.<br>a=ptime:20.<br>a=rtpmap:96 telephone-event/8000.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://68.5.194.163:12113">68.5.194.163:12113</a><br>SIP/2.0 100 Trying.<br>Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163.<br>From: ATAPHONE <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=VxLf2-dmKe50.<br>To: <<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>.<br>Call-ID: <a href="mailto:d44p80-lYfd5f2@domain.company.com">d44p80-lYfd5f2@domain.company.com</a>.<br>CSeq: 177 INVITE.<br>User-Agent: VOIPGATEWAY.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://74.67.200.39:33812">74.67.200.39:33812</a><br>INVITE <a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a> SIP/2.0.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK6pN8Z0pX7y6XD.<br>Route: <<a href="http://sip:142@74.67.200.39:33812">sip:142@74.67.200.39:33812</a>>.<br>Max-Forwards: 68.<br>From: "ATA" <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=U4yBUvmKN9Saj.<br>To: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>.<br>Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243932 INVITE.<br>Contact: <<a href="http://sip:1276@222.222.222.222:5060">sip:1276@222.222.222.222:5060</a>>.<br>User-Agent: VOIPGATEWAY.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 270.<br>X-FS-Support: update_display,send_info.<br>.<br>v=0.<br>o=FreeSWITCH 1455141647 1455141648 IN IP4 222.222.222.222.<br>s=FreeSWITCH.<br>c=IN IP4 222.222.222.222.<br>t=0 0.<br>m=audio 18346 RTP/AVP 0 8 101 13.<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=rtpmap:13 CN/8000.<br>a=ptime:20.<br><br>#<br><br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://74.67.200.39:33812">74.67.200.39:33812</a><br>INVITE <a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a> SIP/2.0.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK6pN8Z0pX7y6XD.<br>Route: <<a href="http://sip:142@74.67.200.39:33812">sip:142@74.67.200.39:33812</a>>.<br>Max-Forwards: 68.<br>From: "ATA" <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=U4yBUvmKN9Saj.<br>To: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>.<br>Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243932 INVITE.<br>Contact: <<a href="http://sip:1276@222.222.222.222:5060">sip:1276@222.222.222.222:5060</a>>.<br>User-Agent: VOIPGATEWAY.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 270.<br>X-FS-Support: update_display,send_info.<br>.<br>v=0.<br>o=FreeSWITCH 1455141647 1455141648 IN IP4 222.222.222.222.<br>s=FreeSWITCH.<br>c=IN IP4 222.222.222.222.<br>t=0 0.<br>m=audio 18346 RTP/AVP 0 8 101 13.<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=rtpmap:13 CN/8000.<br>a=ptime:20.<br><br>##<br>U <a href="http://74.67.200.39:33812">74.67.200.39:33812</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>SIP/2.0 100 Trying.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD.<br>From: "ATA" <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=U4yBUvmKN9Saj.<br>To: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>.<br>Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243932 INVITE.<br>Supported: replaces, path, eventlist.<br>User-Agent: Grandstream Wave/IOS 1.0.19.<br>Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://74.67.200.39:33812">74.67.200.39:33812</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>SIP/2.0 180 Ringing.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD.<br>From: "ATA" <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=U4yBUvmKN9Saj.<br>To: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>;tag=1713531477.<br>Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243932 INVITE.<br>Contact: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>.<br>Supported: replaces, path, timer, eventlist.<br>User-Agent: Grandstream Wave/IOS 1.0.19.<br>Allow-Events: talk, hold.<br>Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://68.5.194.163:12113">68.5.194.163:12113</a><br>SIP/2.0 183 Session Progress.<br>Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163.<br>From: ATAPHONE <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=VxLf2-dmKe50.<br>To: <<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;tag=tU5jS13Fr03Qp.<br>Call-ID: <a href="mailto:d44p80-lYfd5f2@domain.company.com">d44p80-lYfd5f2@domain.company.com</a>.<br>CSeq: 177 INVITE.<br>Contact: <sip:142@222.222.222.222:5060;transport=udp>.<br>User-Agent: VOIPGATEWAY.<br>Accept: application/sdp.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 219.<br>Remote-Party-ID: "142" <<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1455140898 1455140899 IN IP4 222.222.222.222.<br>s=FreeSWITCH.<br>c=IN IP4 222.222.222.222.<br>t=0 0.<br>m=audio 19096 RTP/AVP 0 96.<br>a=rtpmap:0 PCMU/8000.<br>a=rtpmap:96 telephone-event/8000.<br>a=fmtp:96 0-16.<br>a=ptime:20.<br><br>##################################################################<br>U <a href="http://74.67.200.39:33812">74.67.200.39:33812</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD.<br>From: "ATA" <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=U4yBUvmKN9Saj.<br>To: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>;tag=1713531477.<br>Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243932 INVITE.<br>Contact: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>.<br>Supported: replaces, path, timer, eventlist.<br>User-Agent: Grandstream Wave/IOS 1.0.19.<br>Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.<br>Content-Type: application/sdp.<br>Content-Length: 235.<br>.<br>v=0.<br>o=142 8000 8000 IN IP4 172.16.42.13.<br>s=SIP Call.<br>c=IN IP4 172.16.42.13.<br>t=0 0.<br>m=audio 26390 RTP/AVP 0 8 101.<br>a=sendrecv.<br>a=rtpmap:0 PCMU/8000.<br>a=ptime:20.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-15.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://74.67.200.39:33812">74.67.200.39:33812</a><br>ACK <a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a> SIP/2.0.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKB4K487aFSBpUQ.<br>Max-Forwards: 70.<br>From: "ATA" <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=U4yBUvmKN9Saj.<br>To: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>;tag=1713531477.<br>Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243932 ACK.<br>Contact: <<a href="http://sip:1276@222.222.222.222:5060">sip:1276@222.222.222.222:5060</a>>.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://68.5.194.163:12113">68.5.194.163:12113</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163.<br>From: ATAPHONE <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=VxLf2-dmKe50.<br>To: <<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;tag=tU5jS13Fr03Qp.<br>Call-ID: <a href="mailto:d44p80-lYfd5f2@domain.company.com">d44p80-lYfd5f2@domain.company.com</a>.<br>CSeq: 177 INVITE.<br>Contact: <sip:142@222.222.222.222:5060;transport=udp>.<br>User-Agent: VOIPGATEWAY.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 219.<br>Remote-Party-ID: "Outbound Call" <<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1455140898 1455140899 IN IP4 222.222.222.222.<br>s=FreeSWITCH.<br>c=IN IP4 222.222.222.222.<br>t=0 0.<br>m=audio 19096 RTP/AVP 0 96.<br>a=rtpmap:0 PCMU/8000.<br>a=rtpmap:96 telephone-event/8000.<br>a=fmtp:96 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://68.5.194.163:12113">68.5.194.163:12113</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>ACK <a href="http://sip:142@222.222.222.222:5060">sip:142@222.222.222.222:5060</a> SIP/2.0.<br>From: ATAPHONE<<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=VxLf2-dmKe50.<br>To: <a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>;tag=tU5jS13Fr03Qp.<br>Call-ID: <a href="mailto:d44p80-lYfd5f2@domain.company.com">d44p80-lYfd5f2@domain.company.com</a>.<br>CSeq: 177 ACK.<br>Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-VwM5aez*.<br>Contact: ATAPHONE<<a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a>>.<br>Max-Forwards: 70.<br>User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.<br>Content-Length: 0.<br>.<br><br>###########################################################################################<br>##############################<br>##############################################################################################################################################################<br>U <a href="http://74.67.200.39:33812">74.67.200.39:33812</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>BYE <a href="http://sip:1276@222.222.222.222:5060">sip:1276@222.222.222.222:5060</a> SIP/2.0.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK364522226;rport.<br>From: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>;tag=1713531477.<br>To: "ATA" <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=U4yBUvmKN9Saj.<br>Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243933 BYE.<br>Contact: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>.<br>Max-Forwards: 70.<br>Supported: replaces, path, timer, eventlist.<br>User-Agent: Grandstream Wave/IOS 1.0.19.<br>Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://74.67.200.39:33812">74.67.200.39:33812</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK364522226;rport=33812;received=74.67.200.39.<br>From: <<a href="http://sip:142@172.16.42.13:11852">sip:142@172.16.42.13:11852</a>>;tag=1713531477.<br>To: "ATA" <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=U4yBUvmKN9Saj.<br>Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.<br>CSeq: 87243933 BYE.<br>User-Agent: VOIPGATEWAY.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://222.222.222.222:5060">222.222.222.222:5060</a> -> <a href="http://68.5.194.163:12113">68.5.194.163:12113</a><br>BYE <a href="http://sip:1276@192.168.1.168:12113">sip:1276@192.168.1.168:12113</a> SIP/2.0.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK89gKvrKN5DvSQ.<br>Max-Forwards: 70.<br>From: <<a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>>;tag=tU5jS13Fr03Qp.<br>To: ATAPHONE <<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=VxLf2-dmKe50.<br>Call-ID: <a href="mailto:d44p80-lYfd5f2@domain.company.com">d44p80-lYfd5f2@domain.company.com</a>.<br>CSeq: 87243943 BYE.<br>User-Agent: VOIPGATEWAY.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>Supported: path, replaces.<br>Reason: Q.850;cause=16;text="NORMAL_CLEARING".<br>Content-Length: 0.<br><br>##<br>U <a href="http://68.5.194.163:12113">68.5.194.163:12113</a> -> <a href="http://222.222.222.222:5060">222.222.222.222:5060</a><br>SIP/2.0 200 OK.<br>From: <a href="mailto:sip%3A142@domain.company.com">sip:142@domain.company.com</a>;tag=tU5jS13Fr03Qp.<br>To: ATAPHONE<<a href="mailto:sip%3A1276@domain.company.com">sip:1276@domain.company.com</a>>;tag=VxLf2-dmKe50.<br>Call-ID: <a href="mailto:d44p80-lYfd5f2@domain.company.com">d44p80-lYfd5f2@domain.company.com</a>.<br>CSeq: 87243943 BYE.<br>Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK89gKvrKN5DvSQ.<br>User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.<br>Content-Length: 0.<br><br><br><br><br></div></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Feb 10, 2016 at 5:54 PM, Ítalo Rossi <span dir="ltr"><<a href="mailto:italo@freeswitch.org" target="_blank">italo@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">You need to look at the sip signaling to see what's going on</div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="h5">On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <span dir="ltr"><<a href="mailto:shafeeq.v@gmail.com" target="_blank">shafeeq.v@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div dir="ltr">Hello All<br><br><div style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.8px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255)">We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server.<span> </span><br><br>Here is scenario:<span> </span><br><span dir="ltr" style="text-align:left">Grandstream call any extensions (one way audio)<span> </span><br>Any extension call Grandstream ( Audio works just fine)</span><span style="color:rgb(80,0,80)"><br><br>We have tried multiple softphones and the result is same.<span> </span><br><br></span>Everything was working fine with<span> </span><span style="border-bottom:1px dashed rgb(204,204,204)"><span>1.4.18</span></span><span> </span>and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch.<span> </span><br><br></div><div style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.8px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255)">Any help or hint will be much appreciated.<span> </span><br><br></div><div style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.8px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255)">Thank you,<span> </span><br></div><br></div>
<br></div></div>_________________________________________________________________________<br>
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UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><span class="HOEnZb"><font color="#888888"><br></font></span></blockquote></div><span class="HOEnZb"><font color="#888888"><br><br clear="all"><div><br></div>-- <br><div><div dir="ltr">Ítalo Rossi<div><a href="mailto:italo@freeswitch.org" target="_blank">italo@freeswitch.org</a></div></div></div>
</font></span></div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" rel="noreferrer" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
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<a href="http://confluence.freeswitch.org" rel="noreferrer" target="_blank">http://confluence.freeswitch.org</a><br>
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