<meta http-equiv="content-type" content="text/html; charset=utf-8"><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; "><div>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?</div>
<div><br></div><div>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) -<a href="http://gist.github.com/416604" target="_blank" style="color: rgb(87, 151, 176); ">http://gist.github.com/416604</a></div>
<div><br></div><div><span style="font-family: 'Times New Roman'; font-size: medium; "><pre style="white-space: pre-wrap; word-wrap: break-word; ">send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037:
------------------------------------------------------------------------
INVITE <a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a> SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
Max-Forwards: 70
From: "C3 Mgmt" <<a href="mailto:sip%3A9727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:9727289377@63.79.178.192</a>;transport=udp>;tag=tgHN0ScKyNcXr
To: <<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>>
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" <<a href="mailto:sip%3A%2B19727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:+19727289377@63.79.178.192</a>>
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" <<a href="mailto:sip%3A9727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:9727289377@63.79.178.192</a>;transport=udp>;tag=S7QvyyUF1cpaD
t: <<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>>
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" <<a href="mailto:sip%3A9727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:9727289377@63.79.178.192</a>;transport=udp>;tag=S7QvyyUF1cpaD
t: <<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>>;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 <a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a> SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH
Max-Forwards: 70
From: "C3 Mgmt" <<a href="mailto:sip%3A9727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:9727289377@63.79.178.192</a>;transport=udp>;tag=S7QvyyUF1cpaD
To: <<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>>;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 <a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a> SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
Max-Forwards: 70
From: "C3 Mgmt" <<a href="mailto:sip%3A9727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:9727289377@63.79.178.192</a>;transport=udp>;tag=tgHN0ScKyNcXr
To: <<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>>
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" <<a href="mailto:sip%3A%2B19727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:+19727289377@63.79.178.192</a>>
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" <<a href="mailto:sip%3A9727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:9727289377@63.79.178.192</a>;transport=udp>;tag=tgHN0ScKyNcXr
t: <<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>>;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 <a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a> SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
Max-Forwards: 70
From: "C3 Mgmt" <<a href="mailto:sip%3A9727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:9727289377@63.79.178.192</a>;transport=udp>;tag=tgHN0ScKyNcXr
To: <<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>>;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 <a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a> SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr
Max-Forwards: 70
From: "C3 Mgmt" <<a href="mailto:sip%3A9727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:9727289377@63.79.178.192</a>;transport=udp>;tag=tgHN0ScKyNcXr
To: <<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>>
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="<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>", response="1871a9dbd248f92b65df2af84c6057f8"
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 291
X-FS-Support: update_display
P-Asserted-Identity: "C3 Mgmt" <<a href="mailto:sip%3A%2B19727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:+19727289377@63.79.178.192</a>>
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" <<a href="mailto:sip%3A9727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:9727289377@63.79.178.192</a>;transport=udp>;tag=tgHN0ScKyNcXr
t: <<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>>
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" <<a href="mailto:sip%3A9727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:9727289377@63.79.178.192</a>;transport=udp>;tag=tgHN0ScKyNcXr
t: <<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>>;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 <a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a> SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr
Max-Forwards: 70
From: "C3 Mgmt" <<a href="mailto:sip%3A9727289377@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:9727289377@63.79.178.192</a>;transport=udp>;tag=tgHN0ScKyNcXr
To: <<a href="mailto:sip%3A17895551234@63.79.178.192" target="_blank" style="color: rgb(87, 151, 176); ">sip:17895551234@63.79.178.192</a>>;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</pre></span></div><div>Here's how I've edited our acl.conf.xml file:</div><div><br></div>
<div><div><font face="'courier new', monospace"><configuration name="acl.conf" description="Network Lists"></font></div><div><font face="'courier new', monospace"><network-lists></font></div>
<div><font face="'courier new', monospace"> <!--</font></div><div><font face="'courier new', monospace"> These ACL's are automatically created on startup.</font></div><div><font face="'courier new', monospace"><br>
</font></div><div><font face="'courier new', monospace"> rfc1918.auto - RFC1918 Space</font></div><div><font face="'courier new', monospace"> nat.auto - RFC1918 Excluding your local lan.</font></div>
<div><font face="'courier new', monospace"> localnet.auto - ACL for your local lan.</font></div><div><font face="'courier new', monospace"> loopback.auto - ACL for your local lan.</font></div>
<div><font face="'courier new', monospace"> --></font></div><div><font face="'courier new', monospace"><br></font></div><div><font face="'courier new', monospace"> <list name="lan" default="allow"></font></div>
<div><font face="'courier new', monospace"> <node type="deny" cidr="<a href="http://192.168.42.0/24" target="_blank" style="color: rgb(87, 151, 176); ">192.168.42.0/24</a>"/></font></div>
<div><font face="'courier new', monospace"> <node type="allow" cidr="<a href="http://192.168.42.42/32" target="_blank" style="color: rgb(87, 151, 176); ">192.168.42.42/32</a>"/></font></div>
<div><font face="'courier new', monospace"> </list></font></div><div><font face="'courier new', monospace"> <list name="verizon" default="deny"></font></div><div><font face="'courier new', monospace"> <node type="allow" cidr="63.79.178.192"/></font></div>
<div><font face="'courier new', monospace"> </list></font></div></div><div><font face="'courier new', monospace"> .... truncated, the rest is the default file ....</font></div><div><br></div><div>
<a href="http://gist.github.com/416604" target="_blank" style="color: rgb(87, 151, 176); "></a>Any help would be greatly appreciated!</div><div><br></div></span><br><div class="gmail_quote">2010/5/27 <span dir="ltr"><<a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Send FreeSWITCH-users mailing list submissions to<br>
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<br>Today's Topics:<br>
<br>
1. Re: Don't work playback after bypass media mode. (Sergey Scheglov)<br>
2. Re: Don't work playback after bypass media mode. (Sergey Scheglov)<br>
3. Re: Don't work playback after bypass media mode.<br>
(Anthony Minessale)<br>
4. Re: Don't work playback after bypass media mode. (Sergey Scheglov)<br>
5. Re: Don't work playback after bypass media mode. (Sergey Scheglov)<br>
<br><br>---------- Forwarded message ----------<br>From:Sergey Scheglov <<a href="mailto:sid@eltc.ru">sid@eltc.ru</a>><br>To:<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Date:Fri, 28 May 2010 01:31:26 +0700<br>Subject:Re: [Freeswitch-users] Don't work playback after bypass media mode.<br>
<div>
<p>Hi, Anthony.</p>
<p><br></p>
<p>Thank's for reply.</p>
<p><br></p>
<p>You wrote 27 may 2010 Η., 20:34:15:</p>
<p><br></p>
<p><span>> playback executes on line 87 of your trace.</span></p>
<p><br></p>
<p>yes, but executed by log after 30 seconds after line 84 and immediately hangup.</p>
<p>Note, duration wav file - 4 sec.</p>
<p><br></p>
<p><span>> If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or</span></p>
<p><span>the phone you are on does not support early media,</span></p>
<p><span><br></span></p>
<p><span>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.</span></p>
<p><span>If dialplans is:</span></p>
<p><span><extension name="fail_balance"></span></p>
<p><span> <condition field="${blabla}" expression="^\-1$"></span></p>
<p><span> <action application="pre_answer"/></span></p>
<p><span> <action application="sleep" data="1000"/></span></p>
<p><span> <action application="playback" data="elight/neg_balance.wav"/></span></p>
<p><span> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/></span></p>
<p><span> </condition></span></p>
<p><span></extension></span></p>
<p><span>then work's fine, no problems (bypass_media=true not set).</span></p>
<p><span><br></span></p>
<p><span>> change your pre-answer before the playback to answer to verify.</span></p>
<p><br></p>
<p>If set<span>answer, problem disappears. But answer it's not for my case.</span></p>
<p><span><br></span></p>
<p><span>> Try adding sofia profile internal siptrace on to see the sip traffic too.</span></p>
<p><span><br></span></p>
<p><span>Call log with sip trace</span><span><a href="http://pastebin.freeswitch.org/13065" target="_blank">http://pastebin.freeswitch.org/13065</a></span></p>
<p><br></p>
<p>Thanks again :)</p>
<p><br></p>
<p><span>--</span></p>
<p><span>Regard's</span></p>
<p><span>Sergey Scheglov</span></p>
</div>
<br><br>---------- Forwarded message ----------<br>From:Sergey Scheglov <<a href="mailto:sid@eltc.ru">sid@eltc.ru</a>><br>To:Anthony Minessale <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Date:Fri, 28 May 2010 01:35:47 +0700<br>Subject:Re: [Freeswitch-users] Don't work playback after bypass media mode.<br>
<div>
<p>Hi, Anthony.</p>
<p><br></p>
<p>You wrote 27 may 2010 Η., 20:34:15:</p>
<p><br></p>
<div><table border="0" cellpadding="1" cellspacing="2" style="background-color:#ffffff">
<tbody><tr valign="top">
<td width="1" style="background-color:#0000ff"><br>
</td>
<td width="1110">
<p><span>playback executes on line 87 of your trace.</span></p>
<p><br></p>
<p><span>If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or</span></p>
<p><span>the phone you are on does not support early media, change your pre-answer before the playback to answer to verify.</span></p>
<p><br></p>
<p><span>Try adding sofia profile internal siptrace on to see the sip traffic too.</span></p>
<p><br></p>
</td>
</tr>
</tbody></table>
</div>
<p><br></p>
<p><br></p>
<p><span>> playback executes on line 87 of your trace.</span></p>
<p><br></p>
<p>yes, but executed by log after 30 seconds after line 84 and immediately hangup.</p>
<p>Note, duration wav file - 4 sec.</p>
<p><br></p>
<p><span>> If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or</span></p>
<p><span>the phone you are on does not support early media,</span></p>
<p><br></p>
<p><span>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.</span></p>
<p><span>If dialplans is:</span></p>
<p><span><extension name="fail_balance"></span></p>
<p><span><condition field="${blabla}" expression="^\-1$"></span></p>
<p><span><action application="pre_answer"/></span></p>
<p><span><action application="sleep" data="1000"/></span></p>
<p><span><action application="playback" data="elight/neg_balance.wav"/></span></p>
<p><span><action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/></span></p>
<p><span></condition></span></p>
<p><span></extension></span></p>
<p><span>then work's fine, no problems (bypass_media=true not set).</span></p>
<p><br></p>
<p><span>> change your pre-answer before the playback to answer to verify.</span></p>
<p><br></p>
<p>If set<span>answer, problem disappears. But answer it's not for my case.</span></p>
<p><br></p>
<p><span>> Try adding sofia profile internal siptrace on to see the sip traffic too.</span></p>
<p><br></p>
<p><span>Call log with sip trace</span><span><a href="http://pastebin.freeswitch.org/13065" target="_blank">http://pastebin.freeswitch.org/13065</a></span></p>
<p><br></p>
<p>Thanks again :)</p>
<p><br></p>
<p><br></p>
<p><span>--</span></p>
<p><span>Regard's</span></p>
<p><span>Sergey Scheglov</span></p>
</div>
<br><br>---------- Forwarded message ----------<br>From:Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>><br>To:<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Date:Thu, 27 May 2010 13:46:05 -0500<br>Subject:Re: [Freeswitch-users] Don't work playback after bypass media mode.<br>it's against the SIP spec to re-negotiatemedia before you have answered.<div>the transaction to negotiate the SDP from the original call has not completed so it's illegal to send a re-invite.</div>
<div><br></div><div>
instead you should use bypass_media_after_bridge=true so the bypass only happens when the bridge works.</div><div><br><br><div class="gmail_quote">2010/5/27 Sergey Scheglov <span dir="ltr"><<a href="mailto:sid@eltc.ru" target="_blank">sid@eltc.ru</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<p>Hi, Anthony.</p>
<p><br></p>
<p>You wrote 27 may 2010 Η., 20:34:15:</p><div>
<p><br></p>
<div><table border="0" cellpadding="1" cellspacing="2" style="background-color:#ffffff">
<tbody><tr valign="top">
<td width="1" style="background-color:#0000ff"><br>
</td>
<td width="1110">
<p><span>playback executes on line 87 of your trace.</span></p>
<p><br></p>
<p><span>If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or</span></p>
<p><span>the phone you are on does not support early media, change your pre-answer before the playback to answer to verify.</span></p>
<p><br></p>
<p><span>Try adding sofia profile internal siptrace on to see the sip traffic too.</span></p>
<p><br></p>
</td>
</tr>
</tbody></table>
</div>
<p><br></p>
<p><br></p>
<p><span>> playback executes on line 87 of your trace.</span></p>
<p><br></p>
</div><div><p>yes, but executed by log after 30 seconds after line 84 and immediately hangup.</p>
<p>Note, duration wav file - 4 sec.</p>
<p><br></p>
</div><div><p><span>> If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or</span></p>
<p><span>the phone you are on does not support early media,</span></p>
<p><br></p>
</div><div><p><span>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.</span></p>
<p><span>If dialplans is:</span></p>
<p><span><extension name="fail_balance"></span></p>
<p><span><condition field="${blabla}" expression="^\-1$"></span></p>
</div><p><span><action application="pre_answer"/></span></p><div>
<p><span><action application="sleep" data="1000"/></span></p>
</div><p><span><action application="playback" data="elight/neg_balance.wav"/></span></p><div>
<p><span><action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/></span></p>
<p><span></condition></span></p>
<p><span></extension></span></p>
<p><span>then work's fine, no problems (bypass_media=true not set).</span></p>
<p><br></p>
</div><div><p><span>> change your pre-answer before the playback to answer to verify.</span></p>
<p><br></p>
</div><div><p>If set<span>answer, problem disappears. But answer it's not for my case.</span></p>
<p><br></p>
</div><div><p><span>> Try adding sofia profile internal siptrace on to see the sip traffic too.</span></p>
<p><br></p>
</div><div><p><span>Call log with sip trace</span><span><a href="http://pastebin.freeswitch.org/13065" target="_blank">http://pastebin.freeswitch.org/13065</a></span></p>
<p><br></p>
<p>Thanks again :)</p>
<p><br></p>
<p><br></p>
<p><span>--</span></p>
<p><span>Regard's</span></p>
<p><span>Sergey Scheglov</span></p>
</div></div>
<br>_______________________________________________<br>
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<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
Twitter: <a href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br><br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com" target="_blank">MSN:anthony_minessale@hotmail.com</a><br>
GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com" target="_blank">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a><br>
<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org" target="_blank">googletalk:conf+888@conference.freeswitch.org</a><br>
pstn:+19193869900<br>
</div>
<br><br>---------- Forwarded message ----------<br>From:Sergey Scheglov <<a href="mailto:sid@eltc.ru">sid@eltc.ru</a>><br>To:Anthony Minessale <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Date:Fri, 28 May 2010 01:49:04 +0700<br>Subject:Re: [Freeswitch-users] Don't work playback after bypass media mode.<br>Hi, Anthony.<br>
<br>
Thank's for reply.<br>
<br>
You wrote 27 may 2010 Η., 20:34:15:<br>
<br>
> playback executes on line 87 of your trace.<br>
<br>
yes, but executed by log after 30 seconds after line 84 and immediately hangup.<br>
Note, duration wav file - 4 sec.<br>
<br>
> If you do not hear the audio, It means the re-establishment of media fails<br>
> somehow based on your topology or<br>
> the phone you are on does not support early media, change your pre-answer<br>
> before the playback to answer to verify.<br>
<br>
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.<br>
If dialplans is:<br>
<extension name="fail_balance"><br>
<condition field="${blabla}" expression="^\-1$"><br>
<action application="pre_answer"/><br>
<action application="sleep" data="1000"/><br>
<action application="playback" data="elight/neg_balance.wav"/><br>
<action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/><br>
</condition><br>
</extension><br>
then work's fine, no problems (bypass_media=true not set).<br>
<br>
If set answer, problem disappears. But answer it's not for my case.<br>
<br>
> Try adding sofia profile internal siptrace on to see the sip traffic too.<br>
<br>
Call log with sip trace <a href="http://pastebin.freeswitch.org/13065" target="_blank">http://pastebin.freeswitch.org/13065</a><br>
<br>
Thanks again :)<br>
<br>
Best regard's<br>
Sergey Scheglov<br>
<br>
<br>
<br>
<br>
<br><br>---------- Forwarded message ----------<br>From:Sergey Scheglov <<a href="mailto:sid@eltc.ru">sid@eltc.ru</a>><br>To:Anthony Minessale <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Date:Fri, 28 May 2010 02:12:28 +0700<br>Subject:Re: [Freeswitch-users] Don't work playback after bypass media mode.<br>Hi, Anthony.<br>
<br>
Thanks, thanks, thanks ))<br>
You wrote 28 may 2010., 1:46:05:<br>
<br>
> it's against the SIP spec to re-negotiate media before you have answered.<br>
> the transaction to negotiate the SDP from the original call has not<br>
> completed so it's illegal to send a re-invite.<br>
<br>
Do I understand correctly that the dialplan is not correct?<br>
<br>
> instead you should use bypass_media_after_bridge=true so the bypass only<br>
> happens when the bridge works.<br>
<br>
As I wrote in first message:<br>
"If set bypass_media_after_bridge instead bypass_media, then works<br>
fine, BUT changing codec negotiation." And two phones that support<br>
codecs, which not present in internal profile, not be able to call<br>
each other.<br>
<br>
Best regards<br>
Sergey Scheglov<br>
<br>
<br>
<br>
<br>
<br>_______________________________________________<br>
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<br></blockquote></div><br>