407 Proxy Authentication
I've tried updating our ACL conf, but that doesn't seem to help as I still get the 407 error (line 50 from the Gist below). Am I correct that the authentication is failing when Verizon attempts to contact our switch? Or is it an authentication failure when I hit Verizon? Here's the trace of the call attempt (I've replaced an IP address and phone number, so I do realize that they're not correct) - http://gist.github.com/416604 send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037: ------------------------------------------------------------------------ INVITE sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377@63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 INVITE Contact: <sip:gw+verizon@0.0.0.0:5080;transport=udp;gw=verizon> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377@63.79.178.192 <sip%3A%2B19727289377@63.79.178.192>> v=0 o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0 s=FreeSWITCH c=IN IP4 0.0.0.0 t=0 0 m=audio 21600 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2010-05-27 21:00:10.128406 [DEBUG] sofia.c:4168 Channel sofia/external/17895551234 entering state [calling][0] recv 296 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192551: ------------------------------------------------------------------------ SIP/2.0 100 Trying v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0 f: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377@63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD t: <sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>> i: 348b8efb-e497-122d-278a-000bdb94aab9 CSeq: 131385870 INVITE l: 0 ------------------------------------------------------------------------ recv 425 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192710: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0 f: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377@63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD t: <sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>>;tag=22703bab i: 348b8efb-e497-122d-278a-000bdb94aab9 CSeq: 131385870 INVITE l: 0 Proxy-Authenticate: DIGEST realm="WCOM",nonce="a586274e395fb9b6f66f1a7829b4531a.1275008409" ------------------------------------------------------------------------ send 350 bytes to udp/[63.79.178.192]:5060 at 01:00:10.192977: ------------------------------------------------------------------------ ACK sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377@63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD To: <sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>>;tag=22703bab Call-ID: 348b8efb-e497-122d-278a-000bdb94aab9 CSeq: 131385870 ACK Content-Length: 0 ------------------------------------------------------------------------ 2010-05-27 21:00:10.193015 [DEBUG] sofia.c:4168 Channel sofia/external/17895551234 entering state [terminated][904] 2010-05-27 21:00:10.193015 [NOTICE] sofia.c:4804 Hangup sofia/external/17895551234 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED] 2010-05-27 21:00:10.193015 [DEBUG] switch_channel.c:2117 Send signal sofia/external/17895551234 [KILL] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_HANGUP 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499 (sofia/external/17895551234) State HANGUP 2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:410 sofia/external/17895551234 Overriding SIP cause 480 with 904 from the other leg 2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:416 Channel sofia/external/17895551234 hanging up, cause: NORMAL_UNSPECIFIED 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:46 sofia/external/17895551234 Standard HANGUP, cause: NORMAL_UNSPECIFIED 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499 (sofia/external/17895551234) State HANGUP going to sleep 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:333 (sofia/external/17895551234) State Change CS_HANGUP -> CS_REPORTING 2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_REPORTING 2010-05-27 21:00:10.193015 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:590 (sofia/external/17895551234) State REPORTING freeswitch@govinteract-fs-dev-2> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:53 sofia/external/17895551234 Standard REPORTING, cause: NORMAL_UNSPECIFIED 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:590 (sofia/external/17895551234) State REPORTING going to sleep 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:327 (sofia/external/17895551234) State Change CS_REPORTING -> CS_DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1165 Session 6 (sofia/external/17895551234) Locked, Waiting on external entities 2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1183 Session 6 (sofia/external/17895551234) Ended 2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1185 Close Channel sofia/external/17895551234 [CS_DESTROY] 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:428 (sofia/external/17895551234) Running State Change CS_DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439 (sofia/external/17895551234) State DESTROY 2010-05-27 21:00:10.299296 [DEBUG] mod_sofia.c:343 sofia/external/17895551234 SOFIA DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:60 sofia/external/17895551234 Standard DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439 (sofia/external/17895551234) State DESTROY going to sleep freeswitch@govinteract-fs-dev-2> lua test1.lua 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 0 = [sip_cid_type=pid] 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 1 = [origination_caller_id_name=C3 Mgmt] 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 2 = [origination_caller_id_number=+19727289377] 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 3 = [ignore_early_media=true] 2010-05-27 21:04:26.088360 [NOTICE] switch_channel.c:669 New Channel sofia/external/17895551234 [f5c8454e-69f4-11df-b69d-099cfc924087] 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:3444 (sofia/external/17895551234) State Change CS_NEW -> CS_INIT 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_INIT 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338 (sofia/external/17895551234) State INIT 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:83 sofia/external/17895551234 SOFIA INIT 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:117 (sofia/external/17895551234) State Change CS_INIT -> CS_ROUTING 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338 (sofia/external/17895551234) State INIT going to sleep 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_ROUTING 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341 (sofia/external/17895551234) State ROUTING 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:140 sofia/external/17895551234 SOFIA ROUTING 2010-05-27 21:04:26.088360 [DEBUG] switch_ivr_originate.c:66 (sofia/external/17895551234) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341 (sofia/external/17895551234) State ROUTING going to sleep 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_CONSUME_MEDIA 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360 (sofia/external/17895551234) State CONSUME_MEDIA 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360 (sofia/external/17895551234) State CONSUME_MEDIA going to sleep send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037: ------------------------------------------------------------------------ INVITE sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377@63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 INVITE Contact: <sip:gw+verizon@0.0.0.0:5080;transport=udp;gw=verizon> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377@63.79.178.192 <sip%3A%2B19727289377@63.79.178.192>> v=0 o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0 s=FreeSWITCH c=IN IP4 0.0.0.0 t=0 0 m=audio 23386 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2010-05-27 21:04:26.089756 [DEBUG] sofia.c:4168 Channel sofia/external/17895551234 entering state [calling][0] recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.167603: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c;received=0.0.0.0 f: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377@63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr t: <sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>>;tag=41e44fd5 i: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 INVITE l: 0 Proxy-Authenticate: DIGEST realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665" ------------------------------------------------------------------------ send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.167913: ------------------------------------------------------------------------ ACK sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377@63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>>;tag=41e44fd5 Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 ACK Content-Length: 0 ------------------------------------------------------------------------ send 1359 bytes to udp/[63.79.178.192]:5060 at 01:04:26.168477: ------------------------------------------------------------------------ INVITE sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377@63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385998 INVITE Contact: <sip:gw+verizon@0.0.0.0:5080;transport=udp;gw=verizon> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Proxy-Authorization: Digest username="9727289377", realm="WCOM", nonce="43ea850af91d419e72d85c0375a60237.1275008665", algorithm=MD5, uri="sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>", response="1871a9dbd248f92b65df2af84c6057f8" Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377@63.79.178.192 <sip%3A%2B19727289377@63.79.178.192>> v=0 o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0 s=FreeSWITCH c=IN IP4 0.0.0.0 t=0 0 m=audio 23386 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2010-05-27 21:04:26.168049 [DEBUG] sofia.c:4168 Channel sofia/external/17895551234 entering state [calling][0] recv 296 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231231: ------------------------------------------------------------------------ SIP/2.0 100 Trying v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0 f: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377@63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr t: <sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>> i: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385998 INVITE l: 0 ------------------------------------------------------------------------ recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231408: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0 f: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377@63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr t: <sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>>;tag=1da20439 i: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385998 INVITE l: 0 Proxy-Authenticate: DIGEST realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665" ------------------------------------------------------------------------ send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.231667: ------------------------------------------------------------------------ ACK sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377@63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192 <sip%3A17895551234@63.79.178.192>>;tag=1da20439 Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385998 ACK Content-Length: 0 ------------------------------------------------------------------------ 2010-05-27 21:04:26.231294 [DEBUG] sofia.c:4168 Channel sofia/external/17895551234 entering state [terminated][904] 2010-05-27 21:04:26.231294 [NOTICE] sofia.c:4804 Hangup sofia/external/17895551234 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED] 2010-05-27 21:04:26.231294 [DEBUG] switch_channel.c:2117 Send signal sofia/external/17895551234 [KILL] 2010-05-27 21:04:26.231294 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_HANGUP 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:499 (sofia/external/17895551234) State HANGUP 2010-05-27 21:04:26.232334 [DEBUG] mod_sofia.c:410 sofia/external/17895551234 Overriding SIP cause 480 with 904 from the other leg 2010-05-27 21:04:26.232334 [DEBUG] mod_sofia.c:416 Channel sofia/external/17895551234 hanging up, cause: NORMAL_UNSPECIFIED 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:46 sofia/external/17895551234 Standard HANGUP, cause: NORMAL_UNSPECIFIED 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:499 (sofia/external/17895551234) State HANGUP going to sleep 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:333 (sofia/external/17895551234) State Change CS_HANGUP -> CS_REPORTING 2010-05-27 21:04:26.232334 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_REPORTING 2010-05-27 21:04:26.232334 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED] 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:590 (sofia/external/17895551234) State REPORTING freeswitch@govinteract-fs-dev-2> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:53 sofia/external/17895551234 Standard REPORTING, cause: NORMAL_UNSPECIFIED 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:590 (sofia/external/17895551234) State REPORTING going to sleep 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:327 (sofia/external/17895551234) State Change CS_REPORTING -> CS_DESTROY 2010-05-27 21:04:26.260365 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.260365 [DEBUG] switch_core_session.c:1165 Session 7 (sofia/external/17895551234) Locked, Waiting on external entities 2010-05-27 21:04:26.260365 [NOTICE] switch_core_session.c:1183 Session 7 (sofia/external/17895551234) Ended 2010-05-27 21:04:26.260365 [NOTICE] switch_core_session.c:1185 Close Channel sofia/external/17895551234 [CS_DESTROY] 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:428 (sofia/external/17895551234) Running State Change CS_DESTROY 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:439 (sofia/external/17895551234) State DESTROY 2010-05-27 21:04:26.260365 [DEBUG] mod_sofia.c:343 sofia/external/17895551234 SOFIA DESTROY 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:60 sofia/external/17895551234 Standard DESTROY 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:439 (sofia/external/17895551234) State DESTROY going to sleep Here's how I've edited our acl.conf.xml file: <configuration name="acl.conf" description="Network Lists"> <network-lists> <!-- These ACL's are automatically created on startup. rfc1918.auto - RFC1918 Space nat.auto - RFC1918 Excluding your local lan. localnet.auto - ACL for your local lan. loopback.auto - ACL for your local lan. --> <list name="lan" default="allow"> <node type="deny" cidr="192.168.42.0/24"/> <node type="allow" cidr="192.168.42.42/32"/> </list> <list name="verizon" default="deny"> <node type="allow" cidr="63.79.178.192"/> </list> .... truncated, the rest is the default file .... <http://gist.github.com/416604>Any help would be greatly appreciated! 2010/5/27 <freeswitch-users-request@lists.freeswitch.org>
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Today's Topics:
1. Re: Don't work playback after bypass media mode. (Sergey Scheglov) 2. Re: Don't work playback after bypass media mode. (Sergey Scheglov) 3. Re: Don't work playback after bypass media mode. (Anthony Minessale) 4. Re: Don't work playback after bypass media mode. (Sergey Scheglov) 5. Re: Don't work playback after bypass media mode. (Sergey Scheglov)
---------- Forwarded message ---------- From: Sergey Scheglov <sid@eltc.ru> To: freeswitch-users@lists.freeswitch.org Date: Fri, 28 May 2010 01:31:26 +0700 Subject: Re: [Freeswitch-users] Don't work playback after bypass media mode.
Hi, Anthony.
Thank's for reply.
You wrote 27 may 2010 г., 20:34:15:
playback executes on line 87 of your trace.
yes, but executed by log after 30 seconds after line 84 and immediately hangup.
Note, duration wav file - 4 sec.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media,
I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case.
If dialplans is:
<extension name="fail_balance">
<condition field="${blabla}" expression="^\-1$">
<action application="pre_answer"/>
<action application="sleep" data="1000"/>
<action application="playback" data="elight/neg_balance.wav"/>
<action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
</condition>
</extension>
then work's fine, no problems (bypass_media=true not set).
change your pre-answer before the playback to answer to verify.
If set answer, problem disappears. But answer it's not for my case.
Try adding sofia profile internal siptrace on to see the sip traffic too.
Call log with sip trace http://pastebin.freeswitch.org/13065
Thanks again :)
--
Regard's
Sergey Scheglov
---------- Forwarded message ---------- From: Sergey Scheglov <sid@eltc.ru> To: Anthony Minessale <freeswitch-users@lists.freeswitch.org> Date: Fri, 28 May 2010 01:35:47 +0700 Subject: Re: [Freeswitch-users] Don't work playback after bypass media mode.
Hi, Anthony.
You wrote 27 may 2010 г., 20:34:15:
playback executes on line 87 of your trace.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media, change your pre-answer before the playback to answer to verify.
Try adding sofia profile internal siptrace on to see the sip traffic too.
playback executes on line 87 of your trace.
yes, but executed by log after 30 seconds after line 84 and immediately hangup.
Note, duration wav file - 4 sec.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media,
I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case.
If dialplans is:
<extension name="fail_balance">
<condition field="${blabla}" expression="^\-1$">
<action application="pre_answer"/>
<action application="sleep" data="1000"/>
<action application="playback" data="elight/neg_balance.wav"/>
<action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
</condition>
</extension>
then work's fine, no problems (bypass_media=true not set).
change your pre-answer before the playback to answer to verify.
If set answer, problem disappears. But answer it's not for my case.
Try adding sofia profile internal siptrace on to see the sip traffic too.
Call log with sip trace http://pastebin.freeswitch.org/13065
Thanks again :)
--
Regard's
Sergey Scheglov
---------- Forwarded message ---------- From: Anthony Minessale <anthony.minessale@gmail.com> To: freeswitch-users@lists.freeswitch.org Date: Thu, 27 May 2010 13:46:05 -0500 Subject: Re: [Freeswitch-users] Don't work playback after bypass media mode. it's against the SIP spec to re-negotiate media before you have answered. the transaction to negotiate the SDP from the original call has not completed so it's illegal to send a re-invite.
instead you should use bypass_media_after_bridge=true so the bypass only happens when the bridge works.
2010/5/27 Sergey Scheglov <sid@eltc.ru>
Hi, Anthony.
You wrote 27 may 2010 г., 20:34:15:
playback executes on line 87 of your trace.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media, change your pre-answer before the playback to answer to verify.
Try adding sofia profile internal siptrace on to see the sip traffic too.
playback executes on line 87 of your trace.
yes, but executed by log after 30 seconds after line 84 and immediately hangup.
Note, duration wav file - 4 sec.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media,
I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case.
If dialplans is:
<extension name="fail_balance">
<condition field="${blabla}" expression="^\-1$">
<action application="pre_answer"/>
<action application="sleep" data="1000"/>
<action application="playback" data="elight/neg_balance.wav"/>
<action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
</condition>
</extension>
then work's fine, no problems (bypass_media=true not set).
change your pre-answer before the playback to answer to verify.
If set answer, problem disappears. But answer it's not for my case.
Try adding sofia profile internal siptrace on to see the sip traffic too.
Call log with sip trace http://pastebin.freeswitch.org/13065
Thanks again :)
--
Regard's
Sergey Scheglov
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
---------- Forwarded message ---------- From: Sergey Scheglov <sid@eltc.ru> To: Anthony Minessale <freeswitch-users@lists.freeswitch.org> Date: Fri, 28 May 2010 01:49:04 +0700 Subject: Re: [Freeswitch-users] Don't work playback after bypass media mode. Hi, Anthony.
Thank's for reply.
You wrote 27 may 2010 г., 20:34:15:
playback executes on line 87 of your trace.
yes, but executed by log after 30 seconds after line 84 and immediately hangup. Note, duration wav file - 4 sec.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or the phone you are on does not support early media, change your pre-answer before the playback to answer to verify.
I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case. If dialplans is: <extension name="fail_balance"> <condition field="${blabla}" expression="^\-1$"> <action application="pre_answer"/> <action application="sleep" data="1000"/> <action application="playback" data="elight/neg_balance.wav"/> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/> </condition> </extension> then work's fine, no problems (bypass_media=true not set).
If set answer, problem disappears. But answer it's not for my case.
Try adding sofia profile internal siptrace on to see the sip traffic too.
Call log with sip trace http://pastebin.freeswitch.org/13065
Thanks again :)
Best regard's Sergey Scheglov
---------- Forwarded message ---------- From: Sergey Scheglov <sid@eltc.ru> To: Anthony Minessale <freeswitch-users@lists.freeswitch.org> Date: Fri, 28 May 2010 02:12:28 +0700 Subject: Re: [Freeswitch-users] Don't work playback after bypass media mode. Hi, Anthony.
Thanks, thanks, thanks )) You wrote 28 may 2010., 1:46:05:
it's against the SIP spec to re-negotiate media before you have answered. the transaction to negotiate the SDP from the original call has not completed so it's illegal to send a re-invite.
Do I understand correctly that the dialplan is not correct?
instead you should use bypass_media_after_bridge=true so the bypass only happens when the bridge works.
As I wrote in first message: "If set bypass_media_after_bridge instead bypass_media, then works fine, BUT changing codec negotiation." And two phones that support codecs, which not present in internal profile, not be able to call each other.
Best regards Sergey Scheglov
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
If 63.79.178.192 is Verizon, It's Verizon sending the 407, so asking you to authenticate. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 28/05/2010 à 03:19, Michael De Lorenzo a écrit :
I've tried updating our ACL conf, but that doesn't seem to help as I still get the 407 error (line 50 from the Gist below). Am I correct that the authentication is failing when Verizon attempts to contact our switch? Or is it an authentication failure when I hit Verizon?
Here's the trace of the call attempt (I've replaced an IP address and phone number, so I do realize that they're not correct) - http://gist.github.com/416604
send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037:
------------------------------------------------------------------------ INVITE sip:17895551234@63.79.178.192 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c Max-Forwards: 70 From: "C3 Mgmt" <sip: 9727289377@63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 INVITE Contact: <sip:gw+verizon@0.0.0.0:5080;transport=udp;gw=verizon> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377@63.79.178.192>
v=0 o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0 s=FreeSWITCH c=IN IP4 0.0.0.0 t=0 0 m=audio 21600 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
------------------------------------------------------------------------ 2010-05-27 21:00:10.128406 [DEBUG] sofia.c:4168 Channel sofia/ external/17895551234 entering state [calling][0] recv 296 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192551:
------------------------------------------------------------------------ SIP/2.0 100 Trying v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0 f: "C3 Mgmt" <sip: 9727289377@63.79.178.192;transport=udp>;tag=S7QvyyUF1cpaD t: <sip:17895551234@63.79.178.192> i: 348b8efb-e497-122d-278a-000bdb94aab9 CSeq: 131385870 INVITE l: 0
------------------------------------------------------------------------ recv 425 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192710:
------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0 f: "C3 Mgmt" <sip: 9727289377@63.79.178.192;transport=udp>;tag=S7QvyyUF1cpaD t: <sip:17895551234@63.79.178.192>;tag=22703bab i: 348b8efb-e497-122d-278a-000bdb94aab9 CSeq: 131385870 INVITE l: 0 Proxy-Authenticate: DIGEST realm="WCOM",nonce="a586274e395fb9b6f66f1a7829b4531a.1275008409"
------------------------------------------------------------------------ send 350 bytes to udp/[63.79.178.192]:5060 at 01:00:10.192977:
------------------------------------------------------------------------ ACK sip:17895551234@63.79.178.192 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH Max-Forwards: 70 From: "C3 Mgmt" <sip: 9727289377@63.79.178.192;transport=udp>;tag=S7QvyyUF1cpaD To: <sip:17895551234@63.79.178.192>;tag=22703bab Call-ID: 348b8efb-e497-122d-278a-000bdb94aab9 CSeq: 131385870 ACK Content-Length: 0
------------------------------------------------------------------------ 2010-05-27 21:00:10.193015 [DEBUG] sofia.c:4168 Channel sofia/ external/17895551234 entering state [terminated][904] 2010-05-27 21:00:10.193015 [NOTICE] sofia.c:4804 Hangup sofia/ external/17895551234 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED] 2010-05-27 21:00:10.193015 [DEBUG] switch_channel.c:2117 Send signal sofia/external/17895551234 [KILL] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_HANGUP 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499 (sofia/external/17895551234) State HANGUP 2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:410 sofia/external/ 17895551234 Overriding SIP cause 480 with 904 from the other leg 2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:416 Channel sofia/ external/17895551234 hanging up, cause: NORMAL_UNSPECIFIED 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:46 sofia/external/17895551234 Standard HANGUP, cause: NORMAL_UNSPECIFIED 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499 (sofia/external/17895551234) State HANGUP going to sleep 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:333 (sofia/external/17895551234) State Change CS_HANGUP -> CS_REPORTING 2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_REPORTING 2010-05-27 21:00:10.193015 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:590 (sofia/external/17895551234) State REPORTING
freeswitch@govinteract-fs-dev-2> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:53 sofia/external/17895551234 Standard REPORTING, cause: NORMAL_UNSPECIFIED 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:590 (sofia/external/17895551234) State REPORTING going to sleep 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:327 (sofia/external/17895551234) State Change CS_REPORTING -> CS_DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1165 Session 6 (sofia/external/17895551234) Locked, Waiting on external entities 2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1183 Session 6 (sofia/external/17895551234) Ended 2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1185 Close Channel sofia/external/17895551234 [CS_DESTROY] 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:428 (sofia/external/17895551234) Running State Change CS_DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439 (sofia/external/17895551234) State DESTROY 2010-05-27 21:00:10.299296 [DEBUG] mod_sofia.c:343 sofia/external/ 17895551234 SOFIA DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:60 sofia/external/17895551234 Standard DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439 (sofia/external/17895551234) State DESTROY going to sleep
freeswitch@govinteract-fs-dev-2> lua test1.lua 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 0 = [sip_cid_type=pid] 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 1 = [origination_caller_id_name=C3 Mgmt] 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 2 = [origination_caller_id_number=+19727289377] 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 3 = [ignore_early_media=true] 2010-05-27 21:04:26.088360 [NOTICE] switch_channel.c:669 New Channel sofia/external/17895551234 [f5c8454e-69f4-11df-b69d-099cfc924087] 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:3444 (sofia/external/ 17895551234) State Change CS_NEW -> CS_INIT 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_INIT 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338 (sofia/external/17895551234) State INIT 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:83 sofia/external/ 17895551234 SOFIA INIT 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:117 (sofia/external/ 17895551234) State Change CS_INIT -> CS_ROUTING 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338 (sofia/external/17895551234) State INIT going to sleep 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_ROUTING 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341 (sofia/external/17895551234) State ROUTING 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:140 sofia/external/ 17895551234 SOFIA ROUTING 2010-05-27 21:04:26.088360 [DEBUG] switch_ivr_originate.c:66 (sofia/ external/17895551234) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341 (sofia/external/17895551234) State ROUTING going to sleep 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_CONSUME_MEDIA 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360 (sofia/external/17895551234) State CONSUME_MEDIA 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360 (sofia/external/17895551234) State CONSUME_MEDIA going to sleep send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037:
------------------------------------------------------------------------ INVITE sip:17895551234@63.79.178.192 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c Max-Forwards: 70 From: "C3 Mgmt" <sip: 9727289377@63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 INVITE Contact: <sip:gw+verizon@0.0.0.0:5080;transport=udp;gw=verizon> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377@63.79.178.192>
v=0 o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0 s=FreeSWITCH c=IN IP4 0.0.0.0 t=0 0 m=audio 23386 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
------------------------------------------------------------------------ 2010-05-27 21:04:26.089756 [DEBUG] sofia.c:4168 Channel sofia/ external/17895551234 entering state [calling][0] recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.167603:
------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c;received=0.0.0.0 f: "C3 Mgmt" <sip: 9727289377@63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr t: <sip:17895551234@63.79.178.192>;tag=41e44fd5 i: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 INVITE l: 0 Proxy-Authenticate: DIGEST realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665"
------------------------------------------------------------------------ send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.167913:
------------------------------------------------------------------------ ACK sip:17895551234@63.79.178.192 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c Max-Forwards: 70 From: "C3 Mgmt" <sip: 9727289377@63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192>;tag=41e44fd5 Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 ACK Content-Length: 0
------------------------------------------------------------------------ send 1359 bytes to udp/[63.79.178.192]:5060 at 01:04:26.168477:
------------------------------------------------------------------------ INVITE sip:17895551234@63.79.178.192 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr Max-Forwards: 70 From: "C3 Mgmt" <sip: 9727289377@63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385998 INVITE Contact: <sip:gw+verizon@0.0.0.0:5080;transport=udp;gw=verizon> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Proxy-Authorization: Digest username="9727289377", realm="WCOM", nonce="43ea850af91d419e72d85c0375a60237.1275008665", algorithm=MD5, uri="sip:17895551234@63.79.178.192", response="1871a9dbd248f92b65df2af84c6057f8" Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377@63.79.178.192>
v=0 o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0 s=FreeSWITCH c=IN IP4 0.0.0.0 t=0 0 m=audio 23386 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
------------------------------------------------------------------------ 2010-05-27 21:04:26.168049 [DEBUG] sofia.c:4168 Channel sofia/ external/17895551234 entering state [calling][0] recv 296 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231231:
------------------------------------------------------------------------ SIP/2.0 100 Trying v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0 f: "C3 Mgmt" <sip: 9727289377@63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr t: <sip:17895551234@63.79.178.192> i: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385998 INVITE l: 0
------------------------------------------------------------------------ recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231408:
------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0 f: "C3 Mgmt" <sip: 9727289377@63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr t: <sip:17895551234@63.79.178.192>;tag=1da20439 i: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385998 INVITE l: 0 Proxy-Authenticate: DIGEST realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665"
------------------------------------------------------------------------ send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.231667:
------------------------------------------------------------------------ ACK sip:17895551234@63.79.178.192 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr Max-Forwards: 70 From: "C3 Mgmt" <sip: 9727289377@63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192>;tag=1da20439 Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385998 ACK Content-Length: 0
------------------------------------------------------------------------ 2010-05-27 21:04:26.231294 [DEBUG] sofia.c:4168 Channel sofia/ external/17895551234 entering state [terminated][904] 2010-05-27 21:04:26.231294 [NOTICE] sofia.c:4804 Hangup sofia/ external/17895551234 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED] 2010-05-27 21:04:26.231294 [DEBUG] switch_channel.c:2117 Send signal sofia/external/17895551234 [KILL] 2010-05-27 21:04:26.231294 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_HANGUP 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:499 (sofia/external/17895551234) State HANGUP 2010-05-27 21:04:26.232334 [DEBUG] mod_sofia.c:410 sofia/external/ 17895551234 Overriding SIP cause 480 with 904 from the other leg 2010-05-27 21:04:26.232334 [DEBUG] mod_sofia.c:416 Channel sofia/ external/17895551234 hanging up, cause: NORMAL_UNSPECIFIED 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:46 sofia/external/17895551234 Standard HANGUP, cause: NORMAL_UNSPECIFIED 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:499 (sofia/external/17895551234) State HANGUP going to sleep 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:333 (sofia/external/17895551234) State Change CS_HANGUP -> CS_REPORTING 2010-05-27 21:04:26.232334 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_REPORTING 2010-05-27 21:04:26.232334 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED] 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:590 (sofia/external/17895551234) State REPORTING
freeswitch@govinteract-fs-dev-2> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:53 sofia/external/17895551234 Standard REPORTING, cause: NORMAL_UNSPECIFIED 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:590 (sofia/external/17895551234) State REPORTING going to sleep 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:327 (sofia/external/17895551234) State Change CS_REPORTING -> CS_DESTROY 2010-05-27 21:04:26.260365 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.260365 [DEBUG] switch_core_session.c:1165 Session 7 (sofia/external/17895551234) Locked, Waiting on external entities 2010-05-27 21:04:26.260365 [NOTICE] switch_core_session.c:1183 Session 7 (sofia/external/17895551234) Ended 2010-05-27 21:04:26.260365 [NOTICE] switch_core_session.c:1185 Close Channel sofia/external/17895551234 [CS_DESTROY] 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:428 (sofia/external/17895551234) Running State Change CS_DESTROY 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:439 (sofia/external/17895551234) State DESTROY 2010-05-27 21:04:26.260365 [DEBUG] mod_sofia.c:343 sofia/external/ 17895551234 SOFIA DESTROY 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:60 sofia/external/17895551234 Standard DESTROY 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:439 (sofia/external/17895551234) State DESTROY going to sleep Here's how I've edited our acl.conf.xml file:
<configuration name="acl.conf" description="Network Lists"> <network-lists> <!-- These ACL's are automatically created on startup.
rfc1918.auto - RFC1918 Space nat.auto - RFC1918 Excluding your local lan. localnet.auto - ACL for your local lan. loopback.auto - ACL for your local lan. -->
<list name="lan" default="allow"> <node type="deny" cidr="192.168.42.0/24"/> <node type="allow" cidr="192.168.42.42/32"/> </list> <list name="verizon" default="deny"> <node type="allow" cidr="63.79.178.192"/> </list> .... truncated, the rest is the default file ....
Any help would be greatly appreciated!
2010/5/27 <freeswitch-users-request@lists.freeswitch.org> Send FreeSWITCH-users mailing list submissions to freeswitch-users@lists.freeswitch.org
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Today's Topics:
1. Re: Don't work playback after bypass media mode. (Sergey Scheglov) 2. Re: Don't work playback after bypass media mode. (Sergey Scheglov) 3. Re: Don't work playback after bypass media mode. (Anthony Minessale) 4. Re: Don't work playback after bypass media mode. (Sergey Scheglov) 5. Re: Don't work playback after bypass media mode. (Sergey Scheglov)
---------- Forwarded message ---------- From: Sergey Scheglov <sid@eltc.ru> To: freeswitch-users@lists.freeswitch.org Date: Fri, 28 May 2010 01:31:26 +0700 Subject: Re: [Freeswitch-users] Don't work playback after bypass media mode. Hi, Anthony.
Thank's for reply.
You wrote 27 may 2010 г., 20:34:15:
playback executes on line 87 of your trace.
yes, but executed by log after 30 seconds after line 84 and immediately hangup.
Note, duration wav file - 4 sec.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media,
I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case.
If dialplans is:
<extension name="fail_balance">
<condition field="${blabla}" expression="^\-1$">
<action application="pre_answer"/>
<action application="sleep" data="1000"/>
<action application="playback" data="elight/neg_balance.wav"/>
<action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
</condition>
</extension>
then work's fine, no problems (bypass_media=true not set).
change your pre-answer before the playback to answer to verify.
If set answer, problem disappears. But answer it's not for my case.
Try adding sofia profile internal siptrace on to see the sip traffic too.
Call log with sip trace http://pastebin.freeswitch.org/13065
Thanks again :)
--
Regard's
Sergey Scheglov
---------- Forwarded message ---------- From: Sergey Scheglov <sid@eltc.ru> To: Anthony Minessale <freeswitch-users@lists.freeswitch.org> Date: Fri, 28 May 2010 01:35:47 +0700 Subject: Re: [Freeswitch-users] Don't work playback after bypass media mode. Hi, Anthony.
You wrote 27 may 2010 г., 20:34:15:
playback executes on line 87 of your trace.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media, change your pre- answer before the playback to answer to verify.
Try adding sofia profile internal siptrace on to see the sip traffic too.
playback executes on line 87 of your trace.
yes, but executed by log after 30 seconds after line 84 and immediately hangup.
Note, duration wav file - 4 sec.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media,
I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case.
If dialplans is:
<extension name="fail_balance">
<condition field="${blabla}" expression="^\-1$">
<action application="pre_answer"/>
<action application="sleep" data="1000"/>
<action application="playback" data="elight/neg_balance.wav"/>
<action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
</condition>
</extension>
then work's fine, no problems (bypass_media=true not set).
change your pre-answer before the playback to answer to verify.
If set answer, problem disappears. But answer it's not for my case.
Try adding sofia profile internal siptrace on to see the sip traffic too.
Call log with sip trace http://pastebin.freeswitch.org/13065
Thanks again :)
--
Regard's
Sergey Scheglov
---------- Forwarded message ---------- From: Anthony Minessale <anthony.minessale@gmail.com> To: freeswitch-users@lists.freeswitch.org Date: Thu, 27 May 2010 13:46:05 -0500 Subject: Re: [Freeswitch-users] Don't work playback after bypass media mode. it's against the SIP spec to re-negotiate media before you have answered. the transaction to negotiate the SDP from the original call has not completed so it's illegal to send a re-invite.
instead you should use bypass_media_after_bridge=true so the bypass only happens when the bridge works.
2010/5/27 Sergey Scheglov <sid@eltc.ru> Hi, Anthony.
You wrote 27 may 2010 г., 20:34:15:
playback executes on line 87 of your trace.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media, change your pre- answer before the playback to answer to verify.
Try adding sofia profile internal siptrace on to see the sip traffic too.
playback executes on line 87 of your trace.
yes, but executed by log after 30 seconds after line 84 and immediately hangup.
Note, duration wav file - 4 sec.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media,
I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case.
If dialplans is:
<extension name="fail_balance">
<condition field="${blabla}" expression="^\-1$">
<action application="pre_answer"/>
<action application="sleep" data="1000"/>
<action application="playback" data="elight/neg_balance.wav"/>
<action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
</condition>
</extension>
then work's fine, no problems (bypass_media=true not set).
change your pre-answer before the playback to answer to verify.
If set answer, problem disappears. But answer it's not for my case.
Try adding sofia profile internal siptrace on to see the sip traffic too.
Call log with sip trace http://pastebin.freeswitch.org/13065
Thanks again :)
--
Regard's
Sergey Scheglov
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
---------- Forwarded message ---------- From: Sergey Scheglov <sid@eltc.ru> To: Anthony Minessale <freeswitch-users@lists.freeswitch.org> Date: Fri, 28 May 2010 01:49:04 +0700 Subject: Re: [Freeswitch-users] Don't work playback after bypass media mode. Hi, Anthony.
Thank's for reply.
You wrote 27 may 2010 г., 20:34:15:
playback executes on line 87 of your trace.
yes, but executed by log after 30 seconds after line 84 and immediately hangup. Note, duration wav file - 4 sec.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or the phone you are on does not support early media, change your pre- answer before the playback to answer to verify.
I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case. If dialplans is: <extension name="fail_balance"> <condition field="${blabla}" expression="^\-1$"> <action application="pre_answer"/> <action application="sleep" data="1000"/> <action application="playback" data="elight/neg_balance.wav"/> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/> </condition> </extension> then work's fine, no problems (bypass_media=true not set).
If set answer, problem disappears. But answer it's not for my case.
Try adding sofia profile internal siptrace on to see the sip traffic too.
Call log with sip trace http://pastebin.freeswitch.org/13065
Thanks again :)
Best regard's Sergey Scheglov
---------- Forwarded message ---------- From: Sergey Scheglov <sid@eltc.ru> To: Anthony Minessale <freeswitch-users@lists.freeswitch.org> Date: Fri, 28 May 2010 02:12:28 +0700 Subject: Re: [Freeswitch-users] Don't work playback after bypass media mode. Hi, Anthony.
Thanks, thanks, thanks )) You wrote 28 may 2010., 1:46:05:
it's against the SIP spec to re-negotiate media before you have answered. the transaction to negotiate the SDP from the original call has not completed so it's illegal to send a re-invite.
Do I understand correctly that the dialplan is not correct?
instead you should use bypass_media_after_bridge=true so the bypass only happens when the bridge works.
As I wrote in first message: "If set bypass_media_after_bridge instead bypass_media, then works fine, BUT changing codec negotiation." And two phones that support codecs, which not present in internal profile, not be able to call each other.
Best regards Sergey Scheglov
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_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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David Ponzone
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Michael De Lorenzo