May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Any Idea how to let the phone ring with a different ring tone
by Peter P GMX 28 May '10

28 May '10

Hello, for emergency issues I would like to let a phone ring with a different ringer tone so that anybody can hear by the ringer tone that this call is an alarm call. I know that for this purpose an "Alert-Info" can be added to the SIP header. But how has this to be invoked in FS, especially when I want make a group call to a number of phones? Anybody has an idea how to solve this? Best regards Peter

3 3

407 Proxy Authentication
by Michael De Lorenzo 28 May '10

28 May '10

I've tried updating our ACL conf, but that doesn't seem to help as I still get the 407 error (line 50 from the Gist below). Am I correct that the authentication is failing when Verizon attempts to contact our switch? Or is it an authentication failure when I hit Verizon? Here's the trace of the call attempt (I've replaced an IP address and phone number, so I do realize that they're not correct) - http://gist.github.com/416604 send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037: ------------------------------------------------------------------------ INVITE sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377(a)63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192>> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 INVITE Contact: <sip:gw+verizon@0.0.0.0:5080;transport=udp;gw=verizon> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377@63.79.178.192 <sip%3A%2B19727289377(a)63.79.178.192>> v=0 o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0 s=FreeSWITCH c=IN IP4 0.0.0.0 t=0 0 m=audio 21600 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2010-05-27 21:00:10.128406 [DEBUG] sofia.c:4168 Channel sofia/external/17895551234 entering state [calling][0] recv 296 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192551: ------------------------------------------------------------------------ SIP/2.0 100 Trying v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0 f: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377(a)63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD t: <sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192>> i: 348b8efb-e497-122d-278a-000bdb94aab9 CSeq: 131385870 INVITE l: 0 ------------------------------------------------------------------------ recv 425 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192710: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0 f: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377(a)63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD t: <sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192>>;tag=22703bab i: 348b8efb-e497-122d-278a-000bdb94aab9 CSeq: 131385870 INVITE l: 0 Proxy-Authenticate: DIGEST realm="WCOM",nonce="a586274e395fb9b6f66f1a7829b4531a.1275008409" ------------------------------------------------------------------------ send 350 bytes to udp/[63.79.178.192]:5060 at 01:00:10.192977: ------------------------------------------------------------------------ ACK sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377(a)63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD To: <sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192>>;tag=22703bab Call-ID: 348b8efb-e497-122d-278a-000bdb94aab9 CSeq: 131385870 ACK Content-Length: 0 ------------------------------------------------------------------------ 2010-05-27 21:00:10.193015 [DEBUG] sofia.c:4168 Channel sofia/external/17895551234 entering state [terminated][904] 2010-05-27 21:00:10.193015 [NOTICE] sofia.c:4804 Hangup sofia/external/17895551234 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED] 2010-05-27 21:00:10.193015 [DEBUG] switch_channel.c:2117 Send signal sofia/external/17895551234 [KILL] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_HANGUP 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499 (sofia/external/17895551234) State HANGUP 2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:410 sofia/external/17895551234 Overriding SIP cause 480 with 904 from the other leg 2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:416 Channel sofia/external/17895551234 hanging up, cause: NORMAL_UNSPECIFIED 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:46 sofia/external/17895551234 Standard HANGUP, cause: NORMAL_UNSPECIFIED 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499 (sofia/external/17895551234) State HANGUP going to sleep 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:333 (sofia/external/17895551234) State Change CS_HANGUP -> CS_REPORTING 2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_REPORTING 2010-05-27 21:00:10.193015 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED] 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:590 (sofia/external/17895551234) State REPORTING freeswitch@govinteract-fs-dev-2> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:53 sofia/external/17895551234 Standard REPORTING, cause: NORMAL_UNSPECIFIED 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:590 (sofia/external/17895551234) State REPORTING going to sleep 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:327 (sofia/external/17895551234) State Change CS_REPORTING -> CS_DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1165 Session 6 (sofia/external/17895551234) Locked, Waiting on external entities 2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1183 Session 6 (sofia/external/17895551234) Ended 2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1185 Close Channel sofia/external/17895551234 [CS_DESTROY] 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:428 (sofia/external/17895551234) Running State Change CS_DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439 (sofia/external/17895551234) State DESTROY 2010-05-27 21:00:10.299296 [DEBUG] mod_sofia.c:343 sofia/external/17895551234 SOFIA DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:60 sofia/external/17895551234 Standard DESTROY 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439 (sofia/external/17895551234) State DESTROY going to sleep freeswitch@govinteract-fs-dev-2> lua test1.lua 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 0 = [sip_cid_type=pid] 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 1 = [origination_caller_id_name=C3 Mgmt] 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 2 = [origination_caller_id_number=+19727289377] 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885 variable string 3 = [ignore_early_media=true] 2010-05-27 21:04:26.088360 [NOTICE] switch_channel.c:669 New Channel sofia/external/17895551234 [f5c8454e-69f4-11df-b69d-099cfc924087] 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:3444 (sofia/external/17895551234) State Change CS_NEW -> CS_INIT 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_INIT 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338 (sofia/external/17895551234) State INIT 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:83 sofia/external/17895551234 SOFIA INIT 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:117 (sofia/external/17895551234) State Change CS_INIT -> CS_ROUTING 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338 (sofia/external/17895551234) State INIT going to sleep 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_ROUTING 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341 (sofia/external/17895551234) State ROUTING 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:140 sofia/external/17895551234 SOFIA ROUTING 2010-05-27 21:04:26.088360 [DEBUG] switch_ivr_originate.c:66 (sofia/external/17895551234) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/17895551234 [BREAK] 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341 (sofia/external/17895551234) State ROUTING going to sleep 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314 (sofia/external/17895551234) Running State Change CS_CONSUME_MEDIA 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360 (sofia/external/17895551234) State CONSUME_MEDIA 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360 (sofia/external/17895551234) State CONSUME_MEDIA going to sleep send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037: ------------------------------------------------------------------------ INVITE sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377(a)63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192>> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 INVITE Contact: <sip:gw+verizon@0.0.0.0:5080;transport=udp;gw=verizon> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377@63.79.178.192 <sip%3A%2B19727289377(a)63.79.178.192>> v=0 o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0 s=FreeSWITCH c=IN IP4 0.0.0.0 t=0 0 m=audio 23386 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2010-05-27 21:04:26.089756 [DEBUG] sofia.c:4168 Channel sofia/external/17895551234 entering state [calling][0] recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.167603: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c;received=0.0.0.0 f: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377(a)63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr t: <sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192>>;tag=41e44fd5 i: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 INVITE l: 0 Proxy-Authenticate: DIGEST realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665" ------------------------------------------------------------------------ send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.167913: ------------------------------------------------------------------------ ACK sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377(a)63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192>>;tag=41e44fd5 Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385997 ACK Content-Length: 0 ------------------------------------------------------------------------ send 1359 bytes to udp/[63.79.178.192]:5060 at 01:04:26.168477: ------------------------------------------------------------------------ INVITE sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377(a)63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192>> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385998 INVITE Contact: <sip:gw+verizon@0.0.0.0:5080;transport=udp;gw=verizon> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Proxy-Authorization: Digest username="9727289377", realm="WCOM", nonce="43ea850af91d419e72d85c0375a60237.1275008665", algorithm=MD5, uri="sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192>", response="1871a9dbd248f92b65df2af84c6057f8" Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377@63.79.178.192 <sip%3A%2B19727289377(a)63.79.178.192>> v=0 o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0 s=FreeSWITCH c=IN IP4 0.0.0.0 t=0 0 m=audio 23386 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2010-05-27 21:04:26.168049 [DEBUG] sofia.c:4168 Channel sofia/external/17895551234 entering state [calling][0] recv 296 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231231: ------------------------------------------------------------------------ SIP/2.0 100 Trying v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0 f: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377(a)63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr t: <sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192>> i: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385998 INVITE l: 0 ------------------------------------------------------------------------ recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231408: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required v: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0 f: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377(a)63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr t: <sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192>>;tag=1da20439 i: cd26b853-e497-122d-278a-000bdb94aab9 CSeq: 131385998 INVITE l: 0 Proxy-Authenticate: DIGEST realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665" ------------------------------------------------------------------------ send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.231667: ------------------------------------------------------------------------ ACK sip:17895551234@63.79.178.192 <sip%3A17895551234(a)63.79.178.192> SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr Max-Forwards: 70 From: "C3 Mgmt" <sip:9727289377@63.79.178.192 <sip%3A9727289377(a)63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr To: <sip:17895551234@63.79.178.192 <sip%3A17895551234(a)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>  <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(a)lists.freeswitch.org> > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > 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(a)eltc.ru> > To: freeswitch-users(a)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(a)eltc.ru> > To: Anthony Minessale <freeswitch-users(a)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(a)gmail.com> > To: freeswitch-users(a)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(a)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(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> > pstn:+19193869900 > > > ---------- Forwarded message ---------- > From: Sergey Scheglov <sid(a)eltc.ru> > To: Anthony Minessale <freeswitch-users(a)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(a)eltc.ru> > To: Anthony Minessale <freeswitch-users(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

2 1

Don't work playback after bypass media mode.
by Sergey Scheglov 28 May '10

28 May '10

Hi, Freeswitch-users. My dialplan: <extension name="Local_Extension"> <condition field="destination_number" expression="^([1-9]\d{4})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="ringback=${ru-ring}"/> <action application="set" data="bypass_media=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=SUBSCRIBER_ABSENT,USER_NOT_REGISTERED"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="pre_answer"/> <action application="sleep" data="1000"/> <action application="playback" data="elight/${originate_disposition}.wav"/> </condition> </extension> If hangup cause is SUBSCRIBER_ABSENT or USER_NOT_REGISTERED, then playback don't work. Piece of log 2010-05-26 17:35:08.218253 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2010-05-26 17:35:08.218253 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] 2010-05-26 17:35:08.218253 [INFO] mod_dptools.c:2356 Originate Failed. Cause: SUBSCRIBER_ABSENT EXECUTE sofia/internal/39663@10.110.1.8:5060 pre_answer() 2010-05-26 17:35:08.219282 [INFO] mod_dptools.c:746 Sending early media 2010-05-26 17:35:08.219282 [NOTICE] mod_sofia.c:2032 Pre-Answer sofia/internal/39663@10.110.1.8:5060! 2010-05-26 17:35:08.219282 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/39663@10.110.1.8:5060 [BREAK] EXECUTE sofia/internal/39663@10.110.1.8:5060 sleep(1000) 2010-05-26 17:35:08.219282 [DEBUG] sofia.c:4195 Channel sofia/internal/39663@10.110.1.8:5060 skipping state [early][183] 2010-05-26 17:35:09.219252 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/39663@10.110.1.8:5060 [BREAK] 2010-05-26 17:35:39.226271 [ERR] switch_core_io.c:121 sofia/internal/39663@10.110.1.8:5060 has no read codec. 2010-05-26 17:35:39.226271 [DEBUG] switch_core_session.c:1759 Application playback Requires media on channel sofia/internal/39663@10.110.1.8:5060! EXECUTE sofia/internal/39663@10.110.1.8:5060 playback(elight/SUBSCRIBER_ABSENT.wav) 2010-05-26 17:35:39.226271 [NOTICE] switch_core_state_machine.c:185 sofia/internal/39663@10.110.1.8:5060 has executed the last dialplan instruction, hanging up. 2010-05-26 17:35:39.226271 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/39663@10.110.1.8:5060 [CS_EXECUTE] [NORMAL_CLEARING] What am I doing wrong? If set bypass_media_after_bridge instead bypass_media, then works fine, BUT changing codec negotiation. Thanks. -- Regard Sergey Scheglov

4 13

Get on IRC day!!
by Michael Collins 27 May '10

27 May '10

Okay everyone, please join #freeswitch on IRC! We want everyone to come hang out with us. Also, we have many new ones coming in asking for advice. Please join up and help spread the knowledge. See you online! -Michael (IRC: mercutioviz)

1 0

Re: [Freeswitch-users] libtool version mismatch
by David Swardstrom 27 May '10

27 May '10

The only local changes to the code were added (with modifications) by Anthony under JIRA FSCORE-612. So I just did a rm -r * while in /usr/src/freeswitch. Then redid the git to re-create everything. The significant change that I made was to delete a file (or files). I have used TortiseSVN on Windows to rebuild (by doing a full revert) the directory structure. Have not figured out how to use git to do this. Regards, Paul David Swardstrom

1 0

[Out of curiosity] Why not use a voice modem for FXO?
by GillesToo 27 May '10

27 May '10

Hello In case there are system developers out there with good experience writing modem softare, considering that... 1. even a cheap OpenVox PCI card starts at $90 for just one FXO port 2. even entry-level PC's theses days have ample CPU power especially when just used a basic server with Linux 3. we've been using so-called softmodems (ie. controller-less modems that offload processing to the computer's CPU) for over a decade and they sell for about $10 as OEM 4. and finally, Digium/Asterisk's Zaptel driver was precisely meant to run on cheap voice cards ... I was wondering why no one has gone forth and written a driver for this kind of hardware, which would be great for SOHO users to handle just one landline. Is it because the hardware is not standard enough, and writing this kind of driver takes a lot of work? Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Out-of-curiosity-Why-not-use-… Sent from the freeswitch-users mailing list archive at Nabble.com.

6 14

Searching mailing lists
by David Swardstrom 27 May '10

27 May '10

I am new to using FreeSwitch but I have a couple of questions about things. But before I send in the questions, I would like to see if there is any discussion either in the Development or the User list. However, there does not seem to be a search engine capability supplied. Is there a search capability? If so, is there a web page for this? Regards, Paul David Swardstrom

5 7

Disabling voice mail
by Emil Berg 27 May '10

27 May '10

Hello, I am a newbie and I'd like to disable the voice mail. All I want is that when calling to 1000-1010 users, the voice mail will not answer in case that the user is not available. Instead, the server will send a standard reject. Is there any simple way to make it through configuration? I was trying to change the dialplans, but it's very confusing and it didn't work. Thank you, Emil.

4 4

CDRs for Gateways
by Code Ghar 27 May '10

27 May '10

First off, I want to apologize for breaking up this thread into three threads. This is the first time I have ever used mailing lists and it's taken me some time to get things right. Now things should work as they must. David, in response to your last message, in my hurry I forgot to make the necessary modifications in the copy-pasting of config. Here it is rectified. ---Before---   ---After---  <param name="master-file-only" value="true"/> So all I did was enable the master file. After reloadxml all CDRs are added to master file, including those of gateways. Well, according to your mail, you did not change anything :) I guess it is a typo. Can you confirm what you changed ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 22/05/2010 à 03:17, Code Ghar a écrit : > Hi David > > I looked at what you suggested. Before your suggestion, in /usr/ > local/freeswitch/conf/autoload_configs/cdr_csv.conf file, I had the > following lines > > >  >  > > After your suggestion, I changed it to > >  >  > > and then did a reloadxml. Now all CDRs were created in /usr/local/ > freeswitch/log/cdr-csv/Master.csv and CDRs for inbound calls from > unregistered sources (i.e. external gateways) were also created in > this file. Thanks for your help. > > > > Code, > > I think you have a major issue or configuration error with your FS > setup. > FS, as any B2BUA, generates a CDR for any inbound leg, either from a > registered user or from a non-registered gateway. > It doesn't care where it comes from, it just does it. > I can barely imagine a softswitch that would not write the CDR > matching a call that was routed through it. > > The CDR for the outbound leg is optional. > > Can you confirm you are looking in Master.csv ? > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pièces jointes sont confidentiels et établis > à l'intention exclusive de ses destinataires. Toute utilisation ou > diffusion non autorisée est interdite. Tout message électronique est > susceptible d'altération. IPeva décline toute responsabilité au titre > de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes > pas destinataire de ce message, merci de le détruire immédiatement et > d'avertir l'expéditeur. > > > > > Le 21/05/2010 à 05:25, Code Ghar a écrit : > > > Using mod_cdr_csv, it is easy to create CDRs for registered users. > > However, how can we get CDRs for external gateways? For example, > > FreeSWITCH is configured to receive calls from one gateway and send > > them out through another. In this scenario, when neither inbound nor > > outbound gateway registers itself to FreeSWITCH, how can we generate > > and store CDRs for inbound gateway? > > > > Similarly, let's say a SIP provider sends a call meant for a > > registered user. For example, if a cell phone calls 6175550000, it > > is routed through the telephone network to the SIP provider. This > > provider routes the call to FreeSWITCH, which looks up in its dial > > plan that this number should be routed to extension (or registered > > user) 1000. There are no CDRs generated for either the SIP > > provider's gateway or the registered user in this case. > > > > In both these scenarios, if a call is inbound to FreeSWITCH but has > > not been generated by a registered user, mod_cdr_csv does not create > > CDRs. I have even tried to set the following parameter in > > cdr_csv.conf.xml but it didn't help. > > > > <param name="legs" value="ab"/> > > > > My question is this: can mod_cdr_csv generate CDRs for inbound calls > > from sources other than registered users? If it can't, how can this > > be achieved? > > _______________________________________________ > > 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 > >

2 1

Re: [Freeswitch-users] libtool version mismatch
by David Swardstrom 27 May '10

27 May '10

I think I understand what "wipe your tree" means in a sense. But I don't know what I need to do this. I suspect that this means the I need to delete certain files in every subdirectory but what files. While I have used Unix based systems in the past, I never had to use make because either some other build process was used or make was embedded in other tools. For the last couple of years, I have been working with Visual Studio and C++. I think that this "clean-up" everything related to the make and remake would be something that could be useful to add to one of the Wiki pages. Regards, Paul David Swardstrom

2 1