November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem with outbound call to second office
by Charles Bujold 02 Nov '11

02 Nov '11

Strange situation. Have 2 offices each with phones connected to the Freeswitch server in Office 1.. Office 1 Freeswitch Server PFsense Internet Office 2 Telephone 2 Telephone 1 à 192.168.0.15 à 192.162.0.1 à 156.233.20.37 ß Telco Router ß 192.168.1.123 192.168.0.10 152.43.37.129 If the telephone in Office 2, call the telephone in office 1 everything works perfectly. If Office 1 telephone, calls Office 2 Telephone, Telephone 2 rings they can pick up while Freeswitch still gives to Telephone 1 a ring tone even after they pick up and eventually send telephone 1 to voicemail. I placed wireshark on the Freeswitch Server (Ubuntu 10.4 ) and I see the following: Call Originates from 192.168.0.10 to 192.168.0.15 with a SIP packet to 152.43.37.129 I can see that the Call RTP/UDP is to be port 25542 received from Telephone 2 I see that Freeswitch sends a packet to 192.168.0.10 (Telephone 1) I see that Telephone 2 at 152.43.37.129 is asking to talk on port 25542 And then I see that the Freeswitch server send an ICMP packet to 152.43.37.129 stating that the port 25542 is unreachable. Obviously they are talking, just not communicating! The PFsense firewall has all ports open and UDP enable. I even placed a rule that all traffic from 152.43.37.129 be forwarded to the Freeswitch server, and still no voice can be heard. What I dont get, is where does the problem exist, since Office 2 can call us and everything works. This suggest that there is no firewall issue SIP and RTP/UDP is working. Yet when we call, we see that the port has been negotiated which means the SIP port is working and then we get the error that the port is unreachable. My question: - Is the problem in the Ubuntu server ( no firewall) or is it elsewhere? They seem to get in without a problem, why cant we get out? Just to clarify Telephone 1 can make any calls to any other Telephone number (local and long distance) and everything works perfectly. Is the problem in the Telco router? Unable to put a tracer on the Telco router but inside Office 2 wireshark states that it is not receiving anything from Office 1 which corroborates what we see in the Freeswitch server wireshark trace. When I place a wireshark trace at the PFsense server I see the exact same thing as what I see on the Freeswitch server wireshark trace. I can see the Port is unreachable error go to the Office 2 telephone and in Office 2 I see receiving the ICMP message that the port is unreachable. My second question: - Why would the port be unreachable and is it on the Ubuntu server? PFsense server? or Telco router? Third question: - Is there a place in Freeswitch I can check to verify that the RTP ports are in the appropriate range? Any suggestion on how to troubleshoot or resolve this issue would be greatly appreciated. CJB

1 0

No sound when 2 client in the same NAT
by Yihui Li 02 Nov '11

02 Nov '11

Hi, I run FS on a VPS with public IP, and 2 x-lite client on 2 of my home pc which are in the same NAT. They works fine if each of them calls FS sample ivr or bridged to other external phone. But when they call each other, the call is completely silent. The sip trace is listed below. Can anyone please help. Thanks. recv 767 bytes from udp/[90.192.85.6]:6216 at 15:08:19.500457: ------------------------------------------------------------------------ INVITE sip:1015@178.79.188.153 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1017@90.192.85.6:6216> To: "1015"<sip:1015@178.79.188.153> From: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 235 v=0 o=- 8 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 43926 RTP/AVP 0 8 3 101 a=alt:1 1 : tAU3AHbD K56reERa 90.192.85.6 43928 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 379 bytes to udp/[90.192.85.6]:6216 at 15:08:19.501004: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport=6216 From: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d To: "1015"<sip:1015@178.79.188.153> Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Content-Length: 0 ------------------------------------------------------------------------ send 869 bytes to udp/[90.192.85.6]:6216 at 15:08:19.502160: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport=6216 From: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d To: "1015" <sip:1015@178.79.188.153>;tag=K2eK1a4UZ0vvp Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 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 Proxy-Authenticate: Digest realm="178.79.188.153", nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 326 bytes from udp/[90.192.85.6]:6216 at 15:08:19.541053: ------------------------------------------------------------------------ ACK sip:1015@178.79.188.153 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport To: "1015" <sip:1015@178.79.188.153>;tag=K2eK1a4UZ0vvp From: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 1031 bytes from udp/[90.192.85.6]:6216 at 15:08:19.552533: ------------------------------------------------------------------------ INVITE sip:1015@178.79.188.153 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1017@90.192.85.6:6216> To: "1015"<sip:1015@178.79.188.153> From: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="1017",realm="178.79.188.153 ",nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0",uri="sip:1015@178.79.188.153 ",response="603f04a293a7535b109d9fae13085db8",cnonce="fc27cb770d3b4701e15f7aadc4690a64",nc=00000001,qop=auth,algorithm=MD5 User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 235 v=0 o=- 8 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 43926 RTP/AVP 0 8 3 101 a=alt:1 1 : tAU3AHbD K56reERa 90.192.85.6 43928 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 379 bytes to udp/[90.192.85.6]:6216 at 15:08:19.552996: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d To: "1015"<sip:1015@178.79.188.153> Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:08:19.545486 [NOTICE] switch_channel.c:915 New Channel sofia/internal/1017(a)178.79.188.153 [d46f4a05-6572-48e9-bea9-828e70addb74] 2011-11-01 15:08:19.545486 [INFO] mod_dialplan_xml.c:336 Processing 1017 <1017>->1015 in context default 2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *1 execute_extension::dx XML features 2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1017.2011-11-01-15-08-19.wav 2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *3 execute_extension::cf XML features 2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *4 execute_extension::att_xfer XML features 2011-11-01 15:08:19.545486 [NOTICE] switch_channel.c:915 New Channel sofia/internal/sip:1015@90.192.85.6:63216[b6fc17cc-1585-4e43-803d-f431c90fc196] send 1309 bytes to udp/[90.192.85.6]:63216 at 15:08:19.565528: ------------------------------------------------------------------------ INVITE sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKFyaDU717BB1XS Max-Forwards: 69 From: "Extension 1017" <sip:1017@178.79.188.153>;tag=Nm144052Sj91D To: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311> Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Contact: <sip:mod_sofia@178.79.188.153:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 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: 317 X-FS-Support: update_display Remote-Party-ID: "Extension 1017" <sip:1017@178.79.188.153 >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1320132861 1320132862 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27238 RTP/AVP 0 98 99 9 8 3 101 13 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 438 bytes from udp/[90.192.85.6]:63216 at 15:08:19.707081: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311> To: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311>;tag=054e4538 From: "Extension 1017"<sip:1017@178.79.188.153>;tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:08:19.705449 [NOTICE] sofia.c:5375 Ring-Ready sofia/internal/ sip:1015@90.192.85.6:63216! 2011-11-01 15:08:19.705449 [INFO] switch_ivr_originate.c:1115 Sending early media 2011-11-01 15:08:19.705449 [NOTICE] mod_sofia.c:2475 Pre-Answer sofia/internal/1017(a)178.79.188.153! send 1188 bytes to udp/[90.192.85.6]:6216 at 15:08:19.713171: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d To: "1015" <sip:1015@178.79.188.153>;tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Contact: <sip:1015@178.79.188.153:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 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: 251 Remote-Party-ID: "1015" <sip:1015@178.79.188.153 >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1320132951 1320132952 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27148 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ recv 733 bytes from udp/[90.192.85.6]:63216 at 15:08:23.699243: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311> To: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311>;tag=054e4538 From: "Extension 1017"<sip:1017@178.79.188.153>;tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 1 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 44478 RTP/AVP 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 428 bytes to udp/[90.192.85.6]:63216 at 15:08:23.700969: ------------------------------------------------------------------------ ACK sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKg735v2jB9KQgN Max-Forwards: 70 From: "Extension 1017" <sip:1017@178.79.188.153>;tag=Nm144052Sj91D To: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311>;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 ACK Contact: <sip:mod_sofia@178.79.188.153:5060> Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:08:23.685455 [NOTICE] sofia.c:5983 Channel [sofia/internal/ sip:1015@90.192.85.6:63216] has been answered send 1158 bytes to udp/[90.192.85.6]:6216 at 15:08:23.706877: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d To: "1015" <sip:1015@178.79.188.153>;tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Contact: <sip:1015@178.79.188.153:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 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: 251 Remote-Party-ID: "Outbound Call" <sip:1015@178.79.188.153 >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1320132951 1320132952 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27148 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2011-11-01 15:08:23.705453 [NOTICE] switch_ivr_originate.c:3194 Channel [sofia/internal/1017(a)178.79.188.153] has been answered send 1158 bytes to udp/[90.192.85.6]:6216 at 15:08:24.207213: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d To: "1015" <sip:1015@178.79.188.153>;tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Contact: <sip:1015@178.79.188.153:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 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: 251 Remote-Party-ID: "Outbound Call" <sip:1015@178.79.188.153 >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1320132951 1320132952 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27148 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ recv 733 bytes from udp/[90.192.85.6]:63216 at 15:08:24.380014: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311> To: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311>;tag=054e4538 From: "Extension 1017"<sip:1017@178.79.188.153>;tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 1 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 44478 RTP/AVP 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 428 bytes to udp/[90.192.85.6]:63216 at 15:08:24.380297: ------------------------------------------------------------------------ ACK sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKg735v2jB9KQgN Max-Forwards: 70 From: "Extension 1017" <sip:1017@178.79.188.153>;tag=Nm144052Sj91D To: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311>;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 ACK Contact: <sip:mod_sofia@178.79.188.153:5060> Content-Length: 0 ------------------------------------------------------------------------ recv 733 bytes from udp/[90.192.85.6]:63216 at 15:08:25.063077: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311> To: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311>;tag=054e4538 From: "Extension 1017"<sip:1017@178.79.188.153>;tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 1 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 44478 RTP/AVP 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 428 bytes to udp/[90.192.85.6]:63216 at 15:08:25.063384: ------------------------------------------------------------------------ ACK sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKg735v2jB9KQgN Max-Forwards: 70 From: "Extension 1017" <sip:1017@178.79.188.153>;tag=Nm144052Sj91D To: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311>;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 ACK Contact: <sip:mod_sofia@178.79.188.153:5060> Content-Length: 0 ------------------------------------------------------------------------ recv 711 bytes from udp/[90.192.85.6]:6216 at 15:08:25.157427: ------------------------------------------------------------------------ ACK sip:1015@178.79.188.153:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-52077a3eda00e179-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1017@90.192.85.6:6216> To: "1015"<sip:1015@178.79.188.153>;tag=mB8B35mZv9jFj From: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 ACK Proxy-Authorization: Digest username="1017",realm="178.79.188.153 ",nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0",uri="sip:1015@178.79.188.153 ",response="603f04a293a7535b109d9fae13085db8",cnonce="fc27cb770d3b4701e15f7aadc4690a64",nc=00000001,qop=auth,algorithm=MD5 User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ recv 711 bytes from udp/[90.192.85.6]:6216 at 15:08:25.585421: ------------------------------------------------------------------------ ACK sip:1015@178.79.188.153:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-52077a3eda00e179-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1017@90.192.85.6:6216> To: "1015"<sip:1015@178.79.188.153>;tag=mB8B35mZv9jFj From: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 ACK Proxy-Authorization: Digest username="1017",realm="178.79.188.153 ",nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0",uri="sip:1015@178.79.188.153 ",response="603f04a293a7535b109d9fae13085db8",cnonce="fc27cb770d3b4701e15f7aadc4690a64",nc=00000001,qop=auth,algorithm=MD5 User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ recv 533 bytes from udp/[90.192.85.6]:63216 at 15:09:24.414921: ------------------------------------------------------------------------ BYE sip:mod_sofia@178.79.188.153:5060 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:63216 ;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311> To: "Extension 1017"<sip:1017@178.79.188.153>;tag=Nm144052Sj91D From: <sip:1015@90.192.85.6:63216 ;rinstance=4c1fe319ab93f311>;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: eyeBeam release 1102q stamp 51814 Reason: SIP;description="User Hung Up" Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:09:24.425460 [NOTICE] sofia.c:573 Hangup sofia/internal/ sip:1015@90.192.85.6:63216 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] send 576 bytes to udp/[90.192.85.6]:63216 at 15:09:24.426280: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:63216 ;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport=63216 From: <sip:1015@90.192.85.6:63216 ;rinstance=4c1fe319ab93f311>;tag=054e4538 To: "Extension 1017"<sip:1017@178.79.188.153>;tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:09:24.445471 [NOTICE] switch_ivr_bridge.c:1355 Hangup sofia/internal/1017(a)178.79.188.153 [CS_EXECUTE] [NORMAL_CLEARING] 2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1395 Session 2 (sofia/internal/sip:1015@90.192.85.6:63216) Ended 2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/sip:1015@90.192.85.6:63216 [CS_DESTROY] send 623 bytes to udp/[90.192.85.6]:6216 at 15:09:24.448813: ------------------------------------------------------------------------ BYE sip:1017@90.192.85.6:6216 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKHgXyyX3e6vD3g Max-Forwards: 70 From: "1015" <sip:1015@178.79.188.153>;tag=mB8B35mZv9jFj To: "1017" <sip:1017@178.79.188.153>;tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE Contact: <sip:1015@178.79.188.153:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1395 Session 1 (sofia/internal/1017(a)178.79.188.153) Ended 2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/1017(a)178.79.188.153 [CS_DESTROY] recv 533 bytes from udp/[90.192.85.6]:63216 at 15:09:25.132183: ------------------------------------------------------------------------ BYE sip:mod_sofia@178.79.188.153:5060 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:63216 ;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1015@90.192.85.6:63216;rinstance=4c1fe319ab93f311> To: "Extension 1017"<sip:1017@178.79.188.153>;tag=Nm144052Sj91D From: <sip:1015@90.192.85.6:63216 ;rinstance=4c1fe319ab93f311>;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: eyeBeam release 1102q stamp 51814 Reason: SIP;description="User Hung Up" Content-Length: 0 ------------------------------------------------------------------------ send 576 bytes to udp/[90.192.85.6]:63216 at 15:09:25.132451: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:63216 ;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport=63216 From: <sip:1015@90.192.85.6:63216 ;rinstance=4c1fe319ab93f311>;tag=054e4538 To: "Extension 1017"<sip:1017@178.79.188.153>;tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 623 bytes to udp/[90.192.85.6]:6216 at 15:09:25.449077: ------------------------------------------------------------------------ BYE sip:1017@90.192.85.6:6216 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKHgXyyX3e6vD3g Max-Forwards: 70 From: "1015" <sip:1015@178.79.188.153>;tag=mB8B35mZv9jFj To: "1017" <sip:1017@178.79.188.153>;tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE Contact: <sip:1015@178.79.188.153:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 376 bytes from udp/[90.192.85.6]:6216 at 15:09:25.483104: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKHgXyyX3e6vD3g Contact: <sip:1017@90.192.85.6:6216> To: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d From: "1015"<sip:1015@178.79.188.153>;tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ recv 376 bytes from udp/[90.192.85.6]:6216 at 15:09:26.386027: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKHgXyyX3e6vD3g Contact: <sip:1017@90.192.85.6:6216> To: "1017"<sip:1017@178.79.188.153>;tag=ab30a26d From: "1015"<sip:1015@178.79.188.153>;tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------

2 1

no Ringing-Signal on incomming calls via mod_freetdm/sangoma_isdn
by Helmut Kuper 02 Nov '11

02 Nov '11

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I switched from old openzap PRI-Stack to freetdm with sangoma_isdn because of some odd problems with our EWSD switch. Setup was quite easy. Only one thing is strange. When I call into FS from pstn to a sip phone I get no ringing resp ALERTING signal to my pstn phone. Executing ring_ready in dialplan just befor bridging works. Unfortuantely it shows the same behaviour as using "instant_ringback=true", e.g. First ringing then busy ... I had that problem also using old openzap. So today I looked around in FS code and found that when mod_sofia sends a SWITCH_MESSAGE_INDICATE_RINGING on receiving a sip 180 reply back to the caller's channel it works. Is there any reason why sofia doesn't do this by default? Oh, I use FS git-1d3417a 2011-06-07 17-35-49 -0400. best regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk6xpEoACgkQ4tZeNddg3dxGjwCePxDkj4fVdprToKPov5+OeM/3 YakAnitvJZod29hSW4J9T/Sn2Vuob0rU =4KAd -----END PGP SIGNATURE-----

1 0

ekiga, celt codec, Freeswitch and audio problems
by Helmut Kuper 02 Nov '11

02 Nov '11

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried to setup a phone call using celt codec from ekiga to freeswitch to ekiga, where fs doesn't bypass media. ekiga: 3.2.7 (codec list is CELT@48000, CELT@32000, G722, PCMA) Freeswitch: git-a14b20a 2011-09-24 09-58-04 -0500 celt: 0.7.1 (Unfortunately this is what ekiga is using) When FS bypasses the media everything works fine. When ekiga is calling into FS and FS plays a mp3 file to it, it sounds fantastic on ekiga side. When FS doesn't bypasses media the spoken words are distorted. When I use G722 with ekiga and FS in between, there is clear audio. I played around with rtp timer and rewrite_timestamps, but no success. Is it because of celt 0.7.1? Any ideas where to look further? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk6xdFwACgkQ4tZeNddg3dwVfQCdGIX+BSksJp9gfW5C5QWqrS/i zBAAoLm5i8n7wGaEkIFkOKolUPgM3zYE =8WAc -----END PGP SIGNATURE-----

2 2

double dtmf
by Yuriy Nasida 02 Nov '11

02 Nov '11

Hello community, I have some issue with delivery of dtmf.The call flow is:one sip provider - FS - second sip provider - endpoint which expects correct dtmf On FS I have simple custom lua application which implement logic of calling cards system. I have not issues when I call to lua application. I can enter destination number (by using dtmf apparently) and all works fine. But when I try to send dtmf second time for endpoint which expects correct dtmf I have issue with double dtmf. It is noted that when I call from sip device I have not ANY issues. But I have issue when I call from any cell phone or landline phone. Any ideas are welcomed. Thanks.

3 4

LCR and Continue On Fail
by Antonio Teixeira 02 Nov '11

02 Nov '11

Hello Guys. Logs & Dialplan http://pastebin.freeswitch.org/17613 I'm currently trying to connect LCR with the Fall back provided with Continue On Fail. I could be misunderstading the use of continue on fail i read several posts on the mailing and in theory this should work :) Anyway my objective is that If failure _cause == UNALLOCATED_NUMBER continue using the lcr reach else: hangup the call. So this is me or could be a bug ?? Regards & Thanks for the help :) Antonio Teixeira

2 2

Public IP in Contact header
by Hynek Cihlar 02 Nov '11

02 Nov '11

Can anybody please point me to settings or a freeswitch state that makes sofia put public IP into the Contact header? Currently I have SIP clients behind NAT registering to freeswitch. Freeswitch registers these clients with their respective private IPs. Bellow is the profile status. ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 178.238.X.Y,178.238.X.Y Dialplan XML Context default Challenge Realm auto_from RTP-IP 178.238.X.Y Ext-RTP-IP 178.238.X.Y SIP-IP 178.238.X.Y Ext-SIP-IP 178.238.X.Y URL sip:mod_sofia@178.238.X.Y:5060 BIND-URL sip:mod_sofia@178.238.X.Y:5060;maddr=178.238.X.Y HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA,PCMU,GSM,G7221@32000h,G7221@16000h,G722 CODECS OUT PCMA,PCMU,GSM,G7221@32000h,G7221@16000h,G722 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: 3454998XYZ@192_168_59_102 User: 1@ribtjuhd Contact: "383372XYZ" <sip:1@192.168.59.111:5060> Agent: A580 IP/022270000000 Status: Registered(UDP)(unknown) EXP(2011-11-02 13:30:43) EXPSECS(100) Host: eva IP: 81.90.X.Y Port: 22055 Auth-User: 1 Auth-Realm: ribtjuhd MWI-Account: 1@ribtjuhd Thanks! Hynek

3 3

FreeSWITCH Conference Call Today - New Dialplan Feature, SIP 101
by Michael Collins 02 Nov '11

02 Nov '11

Hello all! In our conference call today we are going to discuss a cool new dialplan feature (better logical OR processing) and then Ken Rice and I will be discussing some basic SIP knowledge to help get everyone up to speed when helping newbies. Talk to you all soon! -Michael

1 0

bridging calls from different sofia profiles
by Julien Chavanton 01 Nov '11

01 Nov '11

Hi, I have audio problems when bridging calls from different sofia profiles. I use 2 profiles in order to connect softphone through a VPN to avoid NAT problem One profile is sending/listening on a private IP (for softphone) and the other profile is using a public IP (for external PSTN gateways) I have tested with X-lite and Ekiga and I face different one way audio with both: With X-lite the audio received from the private IP is not forwarded While with Ekiga the audio from the public IP profile is not forwarded I have traced SIP and RTP and the signaling looks good. RTP is received by FS on the correct socket but not forwarded. I understand I do not provide enough information to diagnostic, but maybe I could get input on how to diagnostic RTP socket etc. in this scenario ?

2 1

Log level of log file by fs_cli
by fieldpeak 01 Nov '11

01 Nov '11

Hi friends, i know we can set the loglevel inside switch.conf.xml to change the log level of log files, however, i would like to change the log level of log files thru fs_cli on the fly(no need restart FS), is there any commands to support it, thanks. -- Regards, Charles

4 5