If 63.79.178.192 is Verizon, It's Verizon sending the 407, so asking you to authenticate.

David Ponzone  Direction Technique
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26

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>
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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


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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
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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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