[Freeswitch-users] 407 Proxy Authentication

Michael De Lorenzo delorenzodesign at gmail.com
Thu May 27 18:19:13 PDT 2010


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 at 63.79.178.192
<sip%3A17895551234 at 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 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   To: <sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192>>
   Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
   CSeq: 131385997 INVITE
   Contact: <sip:gw+verizon at 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 at 63.79.178.192
<sip%3A%2B19727289377 at 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 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD
   t: <sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 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 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD
   t: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 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 at 63.79.178.192 <sip%3A17895551234 at 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 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD
   To: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 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 at 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 at 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 at 63.79.178.192
<sip%3A17895551234 at 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 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   To: <sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192>>
   Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
   CSeq: 131385997 INVITE
   Contact: <sip:gw+verizon at 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 at 63.79.178.192
<sip%3A%2B19727289377 at 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 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   t: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 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 at 63.79.178.192 <sip%3A17895551234 at 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 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   To: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 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 at 63.79.178.192
<sip%3A17895551234 at 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 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   To: <sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192>>
   Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
   CSeq: 131385998 INVITE
   Contact: <sip:gw+verizon at 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 at 63.79.178.192 <sip%3A17895551234 at 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 at 63.79.178.192
<sip%3A%2B19727289377 at 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 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   t: <sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 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 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   t: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 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 at 63.79.178.192 <sip%3A17895551234 at 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 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   To: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 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 at 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 at lists.freeswitch.org>

> Send FreeSWITCH-users mailing list submissions to
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> or, via email, send a message with subject or body 'help' to
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> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> 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 at eltc.ru>
> To: freeswitch-users at 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 at eltc.ru>
> To: Anthony Minessale <freeswitch-users at 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 at gmail.com>
> To: freeswitch-users at 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 at 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 at 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 at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: Anthony Minessale <freeswitch-users at 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 at eltc.ru>
> To: Anthony Minessale <freeswitch-users at 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 at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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