November 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Using Lua Dialplan hunting
by Tim St. Pierre 11 Nov '10

11 Nov '10

Hello, I am using lua extensively for dynamic dialplan generatian. In some cases, a dialplan is run with execute_extension instead of transfer. How do I get the Caller-Hunt-Destination from within the lua script? It doesn't seem to be a session variable, and I can't pass arguments to the lua script when it is used as a dialplan. session:getVariable(caller-destination) returns the caller destination, which is usually different than the hunt-destination when execute_extension is used. Any suggestions? Can this be added as a session variable somehow?

3 5

Help! FS Unable to Handle Incoming Calls Through SIP Gateway
by Yasuro 11 Nov '10

11 Nov '10

Hi, FreeSWITCH gurus! I need your help! First off, I am new to FS and I am new to Internet telephony as well. Heck, I am new to the concept of NAT, UPnP, etc., so please bear with my ignorance. I subscribe to a VoIP service at home, with which I get one DID. They supply me a VoIP adapter. Their expected usage is for you to plug in analog phones to the analog phone jacks in the VoIP adapter. However, It also has four Ethernet LAN ports and it acts as a router. You can also access it from the LAN side and register with its built-in SIP gateway. What I would like to do is to run FS (Windows version) on one of the Windows PCs, have it register with the SIP gateway, and have it act as an AA or IVR. For testing, I am having it just play music. When I tried the same idea with AsterikWin32, it worked just as I had hoped; it answered incoming calls automatically. However, I somehow cannot make it work with FS. I simulate incoming calls to my DID number with Skype's Sypeout. It fails after a short while with such error messages as "network error." It appears the call was never answered. FS is assigned an extension number 7 at the gateway. When I call extension 7 from a different extension (at the gateway level, not an extension inside FS), FS does answer the call and I hear music. FS fails to answer only incoming calls from outside. I think my FS configuration is fairly standard. I created an external SIP profile for the gateway under conf/sip_profiles/external/ and modified conf/dialplan/public/00_inbound_did.xml so incoming calls to the gateway will be transferred to an extension within FS. FS's messages and logs, plus the result of packet captures indicate that FS /thinks /it has answered the call, and goes on to initiate media communication. I see RTP packets going from FS to the SIP gateway. What's different from AsteriskWin32's case is that there are no RTP packets coming back from the SIP gateway to FS. Turning of the firewall of the PC does not seem to change the result in any way. For your perusal, I have created the following logs of communication between FS/AsteriskWin32 and the SIP gateway: * AsteriskWin32's case o Summary: http://pastebin.freeswitch.org/14457 o Details: http://pastebin.freeswitch.org/14460 * FreeSWITCh's case o Summary: http://pastebin.freeswitch.org/14462 o Details: http://pastebin.freeswitch.org/14463 o Log: http://pastebin.freeswitch.org/14465 The IP address of the SIP gateway is *192.168.11.250*, and that of the PC FS/AsteriskWin32 resides in is *192.168.11.11*. My DID number is masked as ABCDEFGHIJ. I do not know if it gives you any useful information, but those files include the registration phase. /FS's log was taken at a different time/, so it does not entirely match the packet captures. I also have the corresponding Pcap files. Please let me know if you need them. I am not entirely sure, but I think as far as what I'd like to do is concerned, NAT is not going to be an issue, because the FS/AsteriskWin32 PC and the SIP gateway (its LAN side IP address) are on the same subnet (192.168.11/24). At this time, I do not need to access FS from the Internet. Finally, I will give you more details about my setup, which may or may not be relevant to this issue. My home LAN is set up this way: http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetu… Please note that there are /two layers/ of NAT, and that in the inner layer, two NAT devices exist. I know it looks convoluted, but there are logical reasons for this setup. The VoIP service provider only supports the PCMU codec. The music file I prepared for this testing is encoded in PCMU, so codecs will not be an issue. Please do not hesitate to ask if you have any questions. Thanks for your help in advance! Yasuro

1 0

Free SIP client with TLS support
by Farhan Husain 11 Nov '10

11 Nov '10

Hello, Is there any free SIP client other than SIP Communicator which supports SIPS over TLS? I tried to make SIP Communicator work but failed. I know the old Windows Messenger has support for SIP over TLS but Microsoft does not provide it in its download center any more. Thanks, Farhan

7 7

FTDM Sangoma problem with latest git release
by Stephen Wilde 11 Nov '10

11 Nov '10

2 1

FreeTDM - Screening Indicator Field
by Stephen Wilde 11 Nov '10

11 Nov '10

Hi, I have a server with Freeswitch (latest GIT revision) + FreeTDM with some Sangoma A108 boards. I have the necessity to set, for outgoing calls, the ISDN information element "Screening Indicator" of calling party number. I have see that some information element can be set in the "freetdm.conf.xml" file (i.e. calling and called typer of number or numbering plan) but I have not found any reference to the screening indicator. If there is no way to specify this information element for each trunk, it's possible to change its default value (that is 01=user provided, verified and passed)? Thanks, Stephen

2 7

After Call Work
by Bill Kunyiha 11 Nov '10

11 Nov '10

Is it possible to call a lua script after a call to for example write billing records? What’s the best way of doing this?

5 5

Help! FS Unable to Handle Incoming Calls Through SIP Gateway
by Yasuro 11 Nov '10

11 Nov '10

Hi, FreeSWITCH gurus! I need your help! First off, I am new to FS and I am new to Internet telephony as well. Heck, I am new to the concept of NAT, UPnP, etc., so please bear with my ignorance. I subscribe to a VoIP service at home, with which I get one DID. They supply me a VoIP adapter. Their expected usage is for you to plug in analog phones to the analog phone jacks in the VoIP adapter. However, It also has four Ethernet LAN ports and it acts as a router. You can also access it from the LAN side and register with its built-in SIP gateway. What I would like to do is to run FS (Windows version) on one of the Windows PCs, have it register with the SIP gateway, and have it act as an AA or IVR. For testing, I am having it just play music. When I tried the same idea with AsterikWin32, it worked just as I had hoped; it answered incoming calls automatically. However, I somehow cannot make it work with FS. I simulate incoming calls to my DID number with Skype's Sypeout. It fails after a short while with such error messages as "network error." It appears the call was never answered. FS is assigned an extension number 7 at the gateway. When I call extension 7 from a different extension (at the gateway level, not an extension inside FS), FS does answer the call and I hear music. FS fails to answer only incoming calls from outside. I think my FS configuration is fairly standard. I created an external SIP profile for the gateway under conf/sip_profiles/external/ and modified conf/dialplan/public/00_inbound_did.xml so incoming calls to the gateway will be transferred to an extension within FS. FS's messages and logs, plus the result of packet captures indicate that FS /thinks /it has answered the call, and goes on to initiate media communication. I see RTP packets going from FS to the SIP gateway. What's different from AsteriskWin32's case is that there are no RTP packets coming back from the SIP gateway to FS. Turning of the firewall of the PC does not seem to change the result in any way. For your perusal, I have created the following logs of communication between FS/AsteriskWin32 and the SIP gateway: * AsteriskWin32's case o Summary: http://pastebin.freeswitch.org/14457 o Details: http://pastebin.freeswitch.org/14460 * FreeSWITCh's case o Summary: http://pastebin.freeswitch.org/14462 o Details: http://pastebin.freeswitch.org/14463 o Log: http://pastebin.freeswitch.org/14465 The IP address of the SIP gateway is *192.168.11.250*, and that of the PC FS/AsteriskWin32 resides in is *192.168.11.11*. My DID number is masked as ABCDEFGHIJ. I do not know if it gives you any useful information, but those files include the registration phase. /FS's log was taken at a different time/, so it does not entirely match the packet captures. I also have the corresponding Pcap files. Please let me know if you need them. I am not entirely sure, but I think as far as what I'd like to do is concerned, NAT is not going to be an issue, because the FS/AsteriskWin32 PC and the SIP gateway (its LAN side IP address) are on the same subnet (192.168.11/24). At this time, I do not need to access FS from the Internet. Finally, I will give you more details about my setup, which may or may not be relevant to this issue. My home LAN is set up this way: http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetu… Please note that there are /two layers/ of NAT, and that in the inner layer, two NAT devices exist. I know it looks convoluted, but there are logical reasons for this setup. The VoIP service provider only supports the PCMU codec. The music file I prepared for this testing is encoded in PCMU, so codecs will not be an issue. Please do not hesitate to ask if you have any questions. Thanks for your help in advance! Yasuro

1 0

No Busy Tone and No Announcement Messages on ISDN-VoIP SIP Calls
by Raymond Chan 11 Nov '10

11 Nov '10

[Freeswitch-users] No Busy Tone and No Announcement Messages on ISDN-VoIP SIP Calls Michael Collins msc <mailto:freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users% 5D%20No%20Busy%20Tone%20and%20No%20Announcement%20Messages%20on%0A%20ISDN-Vo IP%20SIP%20Calls&In-Reply-To=018101cb80a8%24bda43ef0%2438ecbcd0%24%40com> at freeswitch.org Wed Nov 10 09:53:09 PST 2010 * Previous message: [Freeswitch-users] <http://lists.freeswitch.org/pipermail/freeswitch-users/2010-November/064962 .html> No Busy Tone and No Announcement Messages on ISDN-VoIP SIP Calls * Next message: [Freeswitch-users] <http://lists.freeswitch.org/pipermail/freeswitch-users/2010-November/064965 .html> trap ## * Messages sorted by: [ <http://lists.freeswitch.org/pipermail/freeswitch-users/2010-November/date.h tml#65110> date ] [ <http://lists.freeswitch.org/pipermail/freeswitch-users/2010-November/thread .html#65110> thread ] [ <http://lists.freeswitch.org/pipermail/freeswitch-users/2010-November/subjec t.html#65110> subject ] [ <http://lists.freeswitch.org/pipermail/freeswitch-users/2010-November/author .html#65110> author ] _____ Hi Michael, The brief network diagram is like this. TDM---Cisco gateway--(SIP)-Freeswitch---IP Phones Cisco gateway and Freeswitch are on the same subnet. Below is the console log (calling from 31754129 to 92882810(this number is a wrong number)). 2010-11-11 11:30:59.210916 [NOTICE] switch_channel.c:779 New Channel sofia/internal/31754129@203.194.134.81:5060 [f3f06789-8c87-4982-956c-6f2b3317af72] 2010-11-11 11:30:59.210916 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/31754129@203.194.134.81:5060) Running State Change CS_NEW 2010-11-11 11:30:59.214961 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/31754129@203.194.134.81:5060) State NEW 2010-11-11 11:30:59.230913 [DEBUG] sofia.c:4455 Channel sofia/internal/31754129@203.194.134.81:5060 entering state [received][100] 2010-11-11 11:30:59.230913 [DEBUG] sofia.c:4466 Remote SDP: v=0 o=DBLE 1289446299 1289446299 IN IP4 10.0.166.181 s=DBLE c=IN IP4 10.0.166.181 t=0 0 m=audio 16384 RTP/AVP 18 4 0 3 8 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2010-11-11 11:30:59.230913 [DEBUG] sofia_glue.c:4000 Audio Codec Compare [G729:18:8000:20]/[PCMU:0:8000:20] 2010-11-11 11:30:59.230913 [DEBUG] sofia_glue.c:4000 Audio Codec Compare [G729:18:8000:20]/[PCMA:8:8000:20] 2010-11-11 11:30:59.230913 [DEBUG] sofia_glue.c:4000 Audio Codec Compare [G723:4:8000:20]/[PCMU:0:8000:20] 2010-11-11 11:30:59.230913 [DEBUG] sofia_glue.c:4000 Audio Codec Compare [G723:4:8000:20]/[PCMA:8:8000:20] 2010-11-11 11:30:59.230913 [DEBUG] sofia_glue.c:4000 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2010-11-11 11:30:59.230913 [DEBUG] sofia_glue.c:2533 Set Codec sofia/internal/31754129@203.194.134.81:5060 PCMU/8000 20 ms 160 samples 2010-11-11 11:30:59.230913 [DEBUG] sofia_glue.c:4096 Set 2833 dtmf send/recv payload to 101 2010-11-11 11:30:59.230913 [DEBUG] sofia.c:4626 (sofia/internal/31754129@203.194.134.81:5060) State Change CS_NEW -> CS_INIT 2010-11-11 11:30:59.230913 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/31754129@203.194.134.81:5060 [BREAK] 2010-11-11 11:30:59.234925 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/31754129@203.194.134.81:5060) Running State Change CS_INIT 2010-11-11 11:30:59.234925 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/31754129@203.194.134.81:5060) State INIT 2010-11-11 11:30:59.234925 [DEBUG] mod_sofia.c:83 sofia/internal/31754129@203.194.134.81:5060 SOFIA INIT 2010-11-11 11:30:59.234925 [DEBUG] mod_sofia.c:123 (sofia/internal/31754129@203.194.134.81:5060) State Change CS_INIT -> CS_ROUTING 2010-11-11 11:30:59.234925 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/31754129@203.194.134.81:5060 [BREAK] 2010-11-11 11:30:59.234925 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/31754129@203.194.134.81:5060) State INIT going to sleep 2010-11-11 11:30:59.234925 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/31754129@203.194.134.81:5060) Running State Change CS_ROUTING 2010-11-11 11:30:59.234925 [DEBUG] switch_channel.c:1517 (sofia/internal/31754129@203.194.134.81:5060) Callstate Change DOWN -> RINGING 2010-11-11 11:30:59.234925 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/31754129@203.194.134.81:5060) State ROUTING 2010-11-11 11:30:59.234925 [DEBUG] mod_sofia.c:146 sofia/internal/31754129@203.194.134.81:5060 SOFIA ROUTING 2010-11-11 11:30:59.234925 [DEBUG] switch_core_state_machine.c:77 sofia/internal/31754129@203.194.134.81:5060 Standard ROUTING 2010-11-11 11:30:59.234925 [INFO] mod_dialplan_xml.c:331 Processing 31754129 <31754129>->92882810 in context default Dialplan: sofia/internal/31754129@203.194.134.81:5060 parsing [default->unloop] continue=false Dialplan: sofia/internal/31754129@203.194.134.81:5060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/31754129@203.194.134.81:5060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/31754129@203.194.134.81:5060 parsing [default->tod_example] continue=true Dialplan: sofia/internal/31754129@203.194.134.81:5060 Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/31754129@203.194.134.81:5060 Action set(open=true) Dialplan: sofia/internal/31754129@203.194.134.81:5060 parsing [default->holiday_example] continue=true Dialplan: sofia/internal/31754129@203.194.134.81:5060 Date/Time Match (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/31754129@203.194.134.81:5060 parsing [default->pizza_demo] continue=false Dialplan: sofia/internal/31754129@203.194.134.81:5060 Regex (FAIL) [pizza_demo] destination_number(92882810) =~ /^(pizza|74992)$/ break=on-false Dialplan: sofia/internal/31754129@203.194.134.81:5060 parsing [default->local.example.com] continue=false Dialplan: sofia/internal/31754129@203.194.134.81:5060 Regex (FAIL) [local.example.com] ${toll_allow}() =~ /local/ break=on-false Dialplan: sofia/internal/31754129@203.194.134.81:5060 parsing [default->domestic.example.com] continue=false Dialplan: sofia/internal/31754129@203.194.134.81:5060 Regex (FAIL) [domestic.example.com] ${toll_allow}() =~ /domestic/ break=on-false Dialplan: sofia/internal/31754129@203.194.134.81:5060 parsing [default->international.example.com] continue=false Dialplan: sofia/internal/31754129@203.194.134.81:5060 Regex (FAIL) [international.example.com] ${toll_allow}() =~ /international/ break=on-false Dialplan: sofia/internal/31754129@203.194.134.81:5060 parsing [default->NWT_DID_outbound_31754129] continue=false Dialplan: sofia/internal/31754129@203.194.134.81:5060 Regex (PASS) [NWT_DID_outbound_31754129] caller_id_number(31754129) =~ /31754129/ break=on-false Dialplan: sofia/internal/31754129@203.194.134.81:5060 Regex (PASS) [NWT_DID_outbound_31754129] destination_number(92882810) =~ /^(\d{8})$/ break=on-false Dialplan: sofia/internal/31754129@203.194.134.81:5060 Action set(ringback=${hk-ring}) Dialplan: sofia/internal/31754129@203.194.134.81:5060 Action bridge(sofia/external/92882810(a)203.194.134.80) 2010-11-11 11:30:59.234925 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/31754129@203.194.134.81:5060) State Change CS_ROUTING -> CS_EXECUTE 2010-11-11 11:30:59.234925 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/31754129@203.194.134.81:5060 [BREAK] 2010-11-11 11:30:59.234925 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/31754129@203.194.134.81:5060) State ROUTING going to sleep 2010-11-11 11:30:59.234925 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/31754129@203.194.134.81:5060) Running State Change CS_EXECUTE 2010-11-11 11:30:59.234925 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/31754129@203.194.134.81:5060) State EXECUTE 2010-11-11 11:30:59.234925 [DEBUG] mod_sofia.c:239 sofia/internal/31754129@203.194.134.81:5060 SOFIA EXECUTE 2010-11-11 11:30:59.234925 [DEBUG] switch_core_state_machine.c:157 sofia/internal/31754129@203.194.134.81:5060 Standard EXECUTE EXECUTE sofia/internal/31754129@203.194.134.81:5060 set(open=true) 2010-11-11 11:30:59.234925 [DEBUG] mod_dptools.c:854 sofia/internal/31754129@203.194.134.81:5060 SET [open]=[true] EXECUTE sofia/internal/31754129@203.194.134.81:5060 set(ringback=%(400,200,440,480);%(400,3000,440,480)) 2010-11-11 11:30:59.234925 [DEBUG] mod_dptools.c:854 sofia/internal/31754129@203.194.134.81:5060 SET [ringback]=[%(400,200,440,480);%(400,3000,440,480)] EXECUTE sofia/internal/31754129@203.194.134.81:5060 bridge(sofia/external/92882810(a)203.194.134.80) 2010-11-11 11:30:59.234925 [NOTICE] switch_channel.c:779 New Channel sofia/external/92882810(a)203.194.134.80 [dca9cd37-7593-417c-82fc-10d98bd90b16] 2010-11-11 11:30:59.243107 [DEBUG] mod_sofia.c:3950 (sofia/external/92882810(a)203.194.134.80) State Change CS_NEW -> CS_INIT 2010-11-11 11:30:59.243107 [DEBUG] switch_core_session.c:1047 Send signal sofia/external/92882810(a)203.194.134.80 [BREAK] 2010-11-11 11:30:59.243107 [DEBUG] switch_core_state_machine.c:314 (sofia/external/92882810(a)203.194.134.80) Running State Change CS_INIT 2010-11-11 11:30:59.243107 [DEBUG] switch_core_state_machine.c:338 (sofia/external/92882810(a)203.194.134.80) State INIT 2010-11-11 11:30:59.243107 [DEBUG] mod_sofia.c:83 sofia/external/92882810(a)203.194.134.80 SOFIA INIT 2010-11-11 11:30:59.243107 [DEBUG] mod_sofia.c:123 (sofia/external/92882810(a)203.194.134.80) State Change CS_INIT -> CS_ROUTING 2010-11-11 11:30:59.243107 [DEBUG] switch_core_session.c:1047 Send signal sofia/external/92882810(a)203.194.134.80 [BREAK] 2010-11-11 11:30:59.243107 [DEBUG] switch_core_state_machine.c:338 (sofia/external/92882810(a)203.194.134.80) State INIT going to sleep 2010-11-11 11:30:59.243107 [DEBUG] switch_core_state_machine.c:314 (sofia/external/92882810(a)203.194.134.80) Running State Change CS_ROUTING 2010-11-11 11:30:59.243107 [DEBUG] switch_channel.c:1517 (sofia/external/92882810(a)203.194.134.80) Callstate Change DOWN -> RINGING 2010-11-11 11:30:59.243107 [DEBUG] switch_core_state_machine.c:341 (sofia/external/92882810(a)203.194.134.80) State ROUTING 2010-11-11 11:30:59.243107 [DEBUG] mod_sofia.c:146 sofia/external/92882810(a)203.194.134.80 SOFIA ROUTING 2010-11-11 11:30:59.243107 [DEBUG] sofia.c:4455 Channel sofia/external/92882810(a)203.194.134.80 entering state [calling][0] 2010-11-11 11:30:59.243107 [DEBUG] switch_ivr_originate.c:66 (sofia/external/92882810(a)203.194.134.80) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-11-11 11:30:59.243107 [DEBUG] switch_core_session.c:1047 Send signal sofia/external/92882810(a)203.194.134.80 [BREAK] 2010-11-11 11:30:59.243107 [DEBUG] switch_core_state_machine.c:341 (sofia/external/92882810(a)203.194.134.80) State ROUTING going to sleep 2010-11-11 11:30:59.243107 [DEBUG] switch_core_state_machine.c:314 (sofia/external/92882810(a)203.194.134.80) Running State Change CS_CONSUME_MEDIA 2010-11-11 11:30:59.243107 [DEBUG] switch_core_state_machine.c:360 (sofia/external/92882810(a)203.194.134.80) State CONSUME_MEDIA 2010-11-11 11:30:59.243107 [DEBUG] switch_core_state_machine.c:360 (sofia/external/92882810(a)203.194.134.80) State CONSUME_MEDIA going to sleep 2010-11-11 11:30:59.726960 [INFO] sofia.c:709 sofia/external/90226864(a)203.194.134.80 Update Callee ID to "90226864" <90226864> 2010-11-11 11:30:59.726960 [DEBUG] sofia.c:4455 Channel sofia/external/90226864(a)203.194.134.80 entering state [proceeding][180] 2010-11-11 11:30:59.726960 [NOTICE] sofia.c:4527 Ring-Ready sofia/external/90226864(a)203.194.134.80! 2010-11-11 11:30:59.738964 [INFO] switch_ivr_originate.c:1079 Sending early media 2010-11-11 11:30:59.738964 [DEBUG] sofia_glue.c:2773 AUDIO RTP [sofia/internal/31754156(a)203.194.134.81] 203.194.134.81 port 30158 -> 120.83.66.232 port 16430 codec: 0 ms: 30 2010-11-11 11:30:59.738964 [DEBUG] switch_rtp.c:1417 Starting timer [soft] 240 bytes per 30ms 2010-11-11 11:30:59.742998 [DEBUG] sofia_glue.c:2983 Set 2833 dtmf send payload to 101 2010-11-11 11:30:59.742998 [DEBUG] sofia_glue.c:2988 Set 2833 dtmf receive payload to 101 2010-11-11 11:30:59.742998 [DEBUG] mod_sofia.c:2168 Ring SDP: v=0 o=FreeSWITCH 1289416101 1289416102 IN IP4 203.194.134.81 s=FreeSWITCH c=IN IP4 203.194.134.81 t=0 0 m=audio 30158 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv 2010-11-11 11:30:59.742998 [NOTICE] mod_sofia.c:2171 Pre-Answer sofia/internal/31754156(a)203.194.134.81! 2010-11-11 11:30:59.742998 [DEBUG] switch_channel.c:2444 (sofia/internal/31754156(a)203.194.134.81) Callstate Change RINGING -> EARLY 2010-11-11 11:30:59.742998 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/31754156(a)203.194.134.81 [BREAK] 2010-11-11 11:30:59.742998 [DEBUG] switch_ivr_originate.c:1128 Raw Codec Activation Success L16@8000hz 1 channel 30ms 2010-11-11 11:30:59.742998 [DEBUG] switch_core_codec.c:116 sofia/internal/31754156(a)203.194.134.81 Push codec L16:10 2010-11-11 11:30:59.742998 [DEBUG] switch_ivr_originate.c:1193 Play Ringback Tone [%(400,200,440,480);%(400,3000,440,480)] 2010-11-11 11:30:59.742998 [DEBUG] sofia.c:4450 Channel sofia/internal/31754156(a)203.194.134.81 skipping state [early][183] 2010-11-11 11:31:00.083016 [INFO] switch_rtp.c:2521 Auto Changing port from 120.83.66.232:16430 to 120.83.66.232:24822 2010-11-11 11:31:00.382989 [INFO] sofia.c:709 sofia/external/92882810(a)203.194.134.80 Update Callee ID to "92882810" <92882810> 2010-11-11 11:31:00.382989 [DEBUG] sofia.c:4455 Channel sofia/external/92882810(a)203.194.134.80 entering state [proceeding][180] 2010-11-11 11:31:00.382989 [NOTICE] sofia.c:4527 Ring-Ready sofia/external/92882810(a)203.194.134.80! 2010-11-11 11:31:00.391009 [INFO] switch_ivr_originate.c:1079 Sending early media 2010-11-11 11:31:00.391009 [DEBUG] sofia_glue.c:2773 AUDIO RTP [sofia/internal/31754129@203.194.134.81:5060] 203.194.134.81 port 21858 -> 10.0.166.181 port 16384 codec: 0 ms: 20 2010-11-11 11:31:00.391009 [DEBUG] switch_rtp.c:1417 Starting timer [soft] 160 bytes per 20ms 2010-11-11 11:31:00.394994 [DEBUG] sofia_glue.c:2983 Set 2833 dtmf send payload to 101 2010-11-11 11:31:00.394994 [DEBUG] sofia_glue.c:2988 Set 2833 dtmf receive payload to 101 2010-11-11 11:31:00.394994 [DEBUG] mod_sofia.c:2168 Ring SDP: v=0 o=FreeSWITCH 1289424402 1289424403 IN IP4 203.194.134.81 s=FreeSWITCH c=IN IP4 203.194.134.81 t=0 0 m=audio 21858 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-11-11 11:31:00.394994 [NOTICE] mod_sofia.c:2171 Pre-Answer sofia/internal/31754129@203.194.134.81:5060! 2010-11-11 11:31:00.394994 [DEBUG] switch_channel.c:2444 (sofia/internal/31754129@203.194.134.81:5060) Callstate Change RINGING -> EARLY 2010-11-11 11:31:00.394994 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/31754129@203.194.134.81:5060 [BREAK] 2010-11-11 11:31:00.394994 [DEBUG] switch_ivr_originate.c:1128 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2010-11-11 11:31:00.394994 [DEBUG] switch_core_codec.c:116 sofia/internal/31754129@203.194.134.81:5060 Push codec L16:10 2010-11-11 11:31:00.394994 [DEBUG] switch_ivr_originate.c:1193 Play Ringback Tone [%(400,200,440,480);%(400,3000,440,480)] 2010-11-11 11:31:00.400593 [DEBUG] sofia.c:4450 Channel sofia/internal/31754129@203.194.134.81:5060 skipping state [early][183] internal/31754129@203.194.134.81:5060 Restore previous codec PCMU:0. 2010-11-11 11:31:59.010658 [DEBUG] switch_channel.c:2357 (sofia/external/92882810(a)203.194.134.80) Callstate Change RINGING -> HANGUP 2010-11-11 11:31:59.010658 [NOTICE] switch_ivr_originate.c:3307 Hangup sofia/external/92882810(a)203.194.134.80 [CS_CONSUME_MEDIA] [NO_ANSWER] 2010-11-11 11:31:59.010658 [DEBUG] switch_channel.c:2373 Send signal sofia/external/92882810(a)203.194.134.80 [KILL] 2010-11-11 11:31:59.010658 [DEBUG] switch_core_session.c:1047 Send signal sofia/external/92882810(a)203.194.134.80 [BREAK] 2010-11-11 11:31:59.010658 [DEBUG] switch_core_state_machine.c:314 (sofia/external/92882810(a)203.194.134.80) Running State Change CS_HANGUP 2010-11-11 11:31:59.010658 [INFO] mod_dptools.c:2411 Originate Failed. Cause: NO_ANSWER 2010-11-11 11:31:59.010658 [NOTICE] switch_core_state_machine.c:185 sofia/internal/31754129@203.194.134.81:5060 has executed the last dialplan instruction, hanging up. 2010-11-11 11:31:59.010658 [DEBUG] switch_channel.c:2357 (sofia/internal/31754129@203.194.134.81:5060) Callstate Change EARLY -> HANGUP 2010-11-11 11:31:59.010658 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/31754129@203.194.134.81:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2010-11-11 11:31:59.010658 [DEBUG] switch_channel.c:2373 Send signal sofia/internal/31754129@203.194.134.81:5060 [KILL] 2010-11-11 11:31:59.010658 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/31754129@203.194.134.81:5060 [BREAK] 2010-11-11 11:31:59.010658 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/31754129@203.194.134.81:5060) State EXECUTE going to sleep 2010-11-11 11:31:59.010658 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/31754129@203.194.134.81:5060) Running State Change CS_HANGUP 2010-11-11 11:31:59.010658 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/31754129@203.194.134.81:5060) State HANGUP 2010-11-11 11:31:59.010658 [DEBUG] mod_sofia.c:457 Channel sofia/internal/31754129@203.194.134.81:5060 hanging up, cause: NORMAL_CLEARING 2010-11-11 11:31:59.010658 [DEBUG] switch_core_state_machine.c:535 (sofia/external/92882810(a)203.194.134.80) State HANGUP 2010-11-11 11:31:59.010658 [DEBUG] mod_sofia.c:457 Channel sofia/external/92882810(a)203.194.134.80 hanging up, cause: NO_ANSWER 2010-11-11 11:31:59.010658 [DEBUG] mod_sofia.c:510 Sending CANCEL to sofia/external/92882810(a)203.194.134.80 2010-11-11 11:31:59.010658 [DEBUG] switch_core_state_machine.c:46 sofia/external/92882810(a)203.194.134.80 Standard HANGUP, cause: NO_ANSWER 2010-11-11 11:31:59.010658 [DEBUG] switch_core_state_machine.c:535 (sofia/external/92882810(a)203.194.134.80) State HANGUP going to sleep 2010-11-11 11:31:59.010658 [DEBUG] switch_core_state_machine.c:333 (sofia/external/92882810(a)203.194.134.80) State Change CS_HANGUP -> CS_REPORTING 2010-11-11 11:31:59.010658 [DEBUG] switch_core_session.c:1047 Send signal sofia/external/92882810(a)203.194.134.80 [BREAK] 2010-11-11 11:31:59.010658 [DEBUG] switch_core_state_machine.c:314 (sofia/external/92882810(a)203.194.134.80) Running State Change CS_REPORTING 2010-11-11 11:31:59.010658 [DEBUG] switch_core_state_machine.c:595 (sofia/external/92882810(a)203.194.134.80) State REPORTING 2010-11-11 11:31:59.014663 [DEBUG] switch_core_state_machine.c:53 sofia/external/92882810(a)203.194.134.80 Standard REPORTING, cause: NO_ANSWER 2010-11-11 11:31:59.014663 [DEBUG] switch_core_state_machine.c:595 (sofia/external/92882810(a)203.194.134.80) State REPORTING going to sleep 2010-11-11 11:31:59.014663 [DEBUG] switch_core_state_machine.c:327 (sofia/external/92882810(a)203.194.134.80) State Change CS_REPORTING -> CS_DESTROY 2010-11-11 11:31:59.014663 [DEBUG] switch_core_session.c:1047 Send signal sofia/external/92882810(a)203.194.134.80 [BREAK] 2010-11-11 11:31:59.014663 [DEBUG] switch_core_session.c:1210 Session 6351 (sofia/external/92882810(a)203.194.134.80) Locked, Waiting on external entities 2010-11-11 11:31:59.014663 [NOTICE] switch_core_session.c:1228 Session 6351 (sofia/external/92882810(a)203.194.134.80) Ended 2010-11-11 11:31:59.014663 [NOTICE] switch_core_session.c:1230 Close Channel sofia/external/92882810(a)203.194.134.80 [CS_DESTROY] 2010-11-11 11:31:59.014663 [DEBUG] switch_core_state_machine.c:427 (sofia/external/92882810(a)203.194.134.80) Callstate Change HANGUP -> DOWN 2010-11-11 11:31:59.014663 [DEBUG] switch_core_state_machine.c:430 (sofia/external/92882810(a)203.194.134.80) Running State Change CS_DESTROY 2010-11-11 11:31:59.014663 [DEBUG] switch_core_state_machine.c:440 (sofia/external/92882810(a)203.194.134.80) State DESTROY 2010-11-11 11:31:59.014663 [DEBUG] mod_sofia.c:362 sofia/external/92882810(a)203.194.134.80 SOFIA DESTROY 2010-11-11 11:31:59.014663 [DEBUG] switch_core_state_machine.c:60 sofia/external/92882810(a)203.194.134.80 Standard DESTROY 2010-11-11 11:31:59.014663 [DEBUG] switch_core_state_machine.c:440 (sofia/external/92882810(a)203.194.134.80) State DESTROY going to sleep 2010-11-11 11:31:59.034666 [DEBUG] mod_sofia.c:519 Responding to INVITE with: 480 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:46 sofia/internal/31754129@203.194.134.81:5060 Standard HANGUP, cause: NORMAL_CLEARING 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/31754129@203.194.134.81:5060) State HANGUP going to sleep 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/31754129@203.194.134.81:5060) State Change CS_HANGUP -> CS_REPORTING 2010-11-11 11:31:59.034666 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/31754129@203.194.134.81:5060 [BREAK] 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/31754129@203.194.134.81:5060) Running State Change CS_REPORTING 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/31754129@203.194.134.81:5060) State REPORTING 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:53 sofia/internal/31754129@203.194.134.81:5060 Standard REPORTING, cause: NORMAL_CLEARING 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/31754129@203.194.134.81:5060) State REPORTING going to sleep 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/31754129@203.194.134.81:5060) State Change CS_REPORTING -> CS_DESTROY 2010-11-11 11:31:59.034666 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/31754129@203.194.134.81:5060 [BREAK] 2010-11-11 11:31:59.034666 [DEBUG] switch_core_session.c:1210 Session 6350 (sofia/internal/31754129@203.194.134.81:5060) Locked, Waiting on external entities 2010-11-11 11:31:59.034666 [NOTICE] switch_core_session.c:1228 Session 6350 (sofia/internal/31754129@203.194.134.81:5060) Ended 2010-11-11 11:31:59.034666 [NOTICE] switch_core_session.c:1230 Close Channel sofia/internal/31754129@203.194.134.81:5060 [CS_DESTROY] 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/31754129@203.194.134.81:5060) Callstate Change HANGUP -> DOWN 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/31754129@203.194.134.81:5060) Running State Change CS_DESTROY 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/31754129@203.194.134.81:5060) State DESTROY 2010-11-11 11:31:59.034666 [DEBUG] mod_sofia.c:362 sofia/internal/31754129@203.194.134.81:5060 SOFIA DESTROY 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:60 sofia/internal/31754129@203.194.134.81:5060 Standard DESTROY 2010-11-11 11:31:59.034666 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/31754129@203.194.134.81:5060) State DESTROY going to sleep Thanks Raymond You've left out a lot of information, like how are these devices connected? and what shows up in the console log? -MC On Tue, Nov 9, 2010 at 11:27 PM, Raymond Chan <raymondchan at commverge.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >wrote: > Hi all, > > > > We are using Freeswitch and using Cisco 2811 as gateway, we have SIP phones > registered in our Freeswitch and trying to call a non-exit number (mobile > number). We found the SIP phone still has ringtone. When we use PSTN to call > that number, there is announcement that "this number does not exist." We > cannot make the SIP phone has the same announcement. But it success when we > calling a land-line non exist number. Busy tone is generated. > > > > Below are the Cisco debug and highlight the 2 different ISDN disconnect > cause code. Do I need to configure anything in the Freeswitch? > > > > > ---------------------------------------------------------------------------- ---------------------------------------------------- > > Calling non-exist mobile number: > > > > *Nov 10 05:54:24.019: %ISDN-6-DISCONNECT: Interface Serial0/0/0:22 > disconnected from 95127917 , call lasted 205 seconds > > *Nov 10 05:54:51.727: //216086/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind: > > Call Entry Is Not Found > > *Nov 10 05:54:51.731: //-1/DE6E855A8453/CCAPI/cc_api_display_ie_subfields: > > cc_api_call_setup_ind_common: > > cisco-username=31754251 > > ----- ccCallInfo IE subfields ----- > > cisco-ani=31754251 > > cisco-anitype=0 > > cisco-aniplan=0 > > cisco-anipi=0 > > cisco-anisi=1 > > dest=92882810 > > cisco-desttype=0 > > cisco-destplan=0 > > cisco-rdie=FFFFFFFF > > cisco-rdn= > > cisco-rdntype=0 > > cisco-rdnplan=0 > > cisco-rdnpi=0 > > cisco-rdnsi=0 > > cisco-redirectreason=-1 > > *Nov 10 05:54:51.731: //-1/DE6E855A8453/CCAPI/cc_api_call_setup_ind_common: > > Interface=0x451F6154, Call Info( > > Calling Number=31754251(TON=Unknown, NPI=Unknown, Screening=User, > Passed, Presentation=Allowed), > > Called Number=92882810(TON=Unknown, NPI=Unknown), > > Calling Translated=FALSE, Subscriber Type Str=Unknown, > FinalDestinationFlag=TRUE, > > Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE > Present=TRUE, > > Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID > Transparent=FALSE), Call Id=216086 > > *Nov 10 05:54:51.731: > //216086/DE6E855A8453/CCAPI/cc_api_call_setup_ind_common: > > Set Up Event Sent; > > Call Info(Calling Number=31754251(TON=Unknown, NPI=Unknown, > Screening=User, Passed, Presentation=Allowed), > > Called Number=92882810(TON=Unknown, NPI=Unknown)) > > *Nov 10 05:54:51.735: > //216086/DE6E855A8453/CCAPI/cc_process_call_setup_ind: > > Event=0x456648F0 > > *Nov 10 05:54:51.735: //216086/DE6E855A8453/CCAPI/ccCallSetContext: > > Context=0x467619B4 > > *Nov 10 05:54:51.735: > //216086/DE6E855A8453/CCAPI/cc_process_call_setup_ind: > > >>>>CCAPI handed cid 216086 with tag 0 to app > "_ManagedAppProcess_Default" > > *Nov 10 05:54:51.735: //216086/DE6E855A8453/CCAPI/ccCallProceeding: > > Progress Indication=NULL(0) > > *Nov 10 05:54:51.735: //216086/DE6E855A8453/CCAPI/ccCallSetupRequest: > > Destination=, Calling IE Present=TRUE, Mode=0, > > Outgoing Dial-peer=1, Params=0x4740AA5C, Progress Indication=ORIGINATING > SIDE IS NON ISDN(3) > > *Nov 10 05:54:51.735: //216086/DE6E855A8453/CCAPI/ccCheckClipClir: > > In: Calling Number=31754251(TON=Unknown, NPI=Unknown, Screening=User, > Passed, Presentation=Allowed) > > *Nov 10 05:54:51.735: //216086/DE6E855A8453/CCAPI/ccCheckClipClir: > > Out: Calling Number=31754251(TON=Unknown, NPI=Unknown, Screening=User, > Passed, Presentation=Allowed) > > *Nov 10 05:54:51.735: //216086/DE6E855A8453/CCAPI/ccCallSetupRequest: > > Destination Pattern=.T, Called Number=92882810, Digit Strip=FALSE > > *Nov 10 05:54:51.735: //216086/DE6E855A8453/CCAPI/ccCallSetupRequest: > > Calling Number=31754251(TON=Unknown, NPI=Unknown, Screening=User, > Passed, Presentation=Allowed), > > Called Number=92882810(TON=Unknown, NPI=Unknown), > > Redirect Number=, Display Info=31754251 > > Account Number=31754251, Final Destination Flag=TRUE, > > Guid=DE6E855A-EBC5-11DF-8453-C1363B6B99C1, Outgoing Dial-peer=1 > > *Nov 10 05:54:51.739: > //216086/DE6E855A8453/CCAPI/cc_api_display_ie_subfields: > > ccCallSetupRequest: > > cisco-username=31754251 > > ----- ccCallInfo IE subfields ----- > > cisco-ani=31754251 > > cisco-anitype=0 > > cisco-aniplan=0 > > cisco-anipi=0 > > cisco-anisi=1 > > dest=92882810 > > cisco-desttype=0 > > cisco-destplan=0 > > cisco-rdie=FFFFFFFF > > cisco-rdn= > > cisco-rdntype=0 > > cisco-rdnplan=0 > > cisco-rdnpi=0 > > cisco-rdnsi=0 > > cisco-redirectreason=-1 > > *Nov 10 05:54:51.739: > //216086/DE6E855A8453/CCAPI/ccIFCallSetupRequestPrivate: > > Interface=0x456ACB10, Interface Type=6, Destination=, Mode=0x0, > > Call Params(Calling Number=31754251(TON=Unknown, NPI=Unknown, > Screening=User, Passed, Presentation=Allowed), > > Called Number=92882810(TON=Unknown, NPI=Unknown), Calling > Translated=FALSE, > > Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing > Dial-peer=1, Call Count On=FALSE, > > Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, > Application Call Id=) > > *Nov 10 05:54:51.739: > //216087/DE6E855A8453/CCAPI/ccIFCallSetupRequestPrivate: > > SPI Call Setup Request Is Success; Interface Type=6, FlowMode=1 > > *Nov 10 05:54:51.739: //216087/DE6E855A8453/CCAPI/ccCallSetContext: > > Context=0x4740AA0C > > *Nov 10 05:54:51.739: //216086/DE6E855A8453/CCAPI/ccSaveDialpeerTag: > > Outgoing Dial-peer=1 > > *Nov 10 05:54:51.779: //216087/DE6E855A8453/CCAPI/cc_api_call_proceeding: > > Interface=0x456ACB10, Progress Indication=NULL(0) > > *Nov 10 05:54:52.947: //216087/DE6E855A8453/CCAPI/cc_api_call_alert: > > Interface=0x456ACB10, Progress Indication=NULL(0), Signal Indication=NOT > PRESENT(255) > > *Nov 10 05:54:52.947: //216087/DE6E855A8453/CCAPI/cc_api_call_alert: > > Call Entry(Retry Count=0, Responsed=TRUE) > > *Nov 10 05:54:52.947: //216086/DE6E855A8453/CCAPI/ccCallAlert: > > Progress Indication=NULL(0), Signal Indication=NOT PRESENT(255) > > *Nov 10 05:54:52.947: //216086/DE6E855A8453/CCAPI/ccCallAlert: > > Call Entry(Responsed=TRUE, AlertSent=TRUE) > > *Nov 10 05:55:50.971: //216086/DE6E855A8453/CCAPI/cc_api_call_disconnected: > > Cause Value=16, Interface=0x451F6154, Call Id=216086 > > *Nov 10 05:55:50.971: //216086/DE6E855A8453/CCAPI/cc_api_call_disconnected: > > Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0) > > *Nov 10 05:55:50.971: //216087/DE6E855A8453/CCAPI/ccCallDisconnect: > > Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, > Disconnect Cause=0) > > *Nov 10 05:55:50.971: //216087/DE6E855A8453/CCAPI/ccCallDisconnect: > > Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16) > > *Nov 10 05:55:50.971: //216087/DE6E855A8453/CCAPI/cc_api_get_transfer_info: > > Transfer Number Is Null > > *Nov 10 05:55:50.975: //216086/DE6E855A8453/CCAPI/ccCallDisconnect: > > Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, > Disconnect Cause=16) > > *Nov 10 05:55:50.975: //216086/DE6E855A8453/CCAPI/ccCallDisconnect: > > Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16) > > *Nov 10 05:55:50.983: > //216086/DE6E855A8453/CCAPI/cc_api_call_disconnect_done: > > Disposition=0, Interface=0x451F6154, Tag=0x0, Call Id=216086, > > Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0) > > *Nov 10 05:55:50.983: > //216086/DE6E855A8453/CCAPI/cc_api_call_disconnect_done: > > Call Disconnect Event Sent > > > > > ---------------------------------------------------------------------------- --------------------------------------------- > > > > Calling non-exist land line number: > > > > cvghk-vgw1#sh log > > Syslog logging: enabled (11 messages dropped, 74 messages rate-limited, > > 68247 flushes, 0 overruns, xml disabled, filtering > disabled) > > Console logging: level debugging, 2492150 messages logged, xml > disabled, > > filtering disabled > > Monitor logging: level debugging, 0 messages logged, xml disabled, > > filtering disabled > > Buffer logging: level debugging, 2285346 messages logged, xml disabled, > > filtering disabled > > Logging Exception size (4096 bytes) > > Count and timestamp logging messages: disabled > > > > No active filter modules. > > > > Trap logging: level informational, 109823 message lines logged > > > > Log Buffer (8000 bytes): > > > > *Nov 10 06:01:45.127: //216094/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind: > > Call Entry Is Not Found > > *Nov 10 06:01:45.127: //-1/D4D5BF13845B/CCAPI/cc_api_display_ie_subfields: > > cc_api_call_setup_ind_common: > > cisco-username=31754250 > > ----- ccCallInfo IE subfields ----- > > cisco-ani=31754250 > > cisco-anitype=0 > > cisco-aniplan=0 > > cisco-anipi=0 > > cisco-anisi=1 > > dest=22222222 > > cisco-desttype=0 > > cisco-destplan=0 > > cisco-rdie=FFFFFFFF > > cisco-rdn= > > cisco-rdntype=0 > > cisco-rdnplan=0 > > cisco-rdnpi=0 > > cisco-rdnsi=0 > > cisco-redirectreason=-1 > > *Nov 10 06:01:45.127: //-1/D4D5BF13845B/CCAPI/cc_api_call_setup_ind_common: > > Interface=0x451F6154, Call Info( > > Calling Number=31754250(TON=Unknown, NPI=Unknown, Screening=User, > Passed, > Pre > sentation=Allowed), > > Called Number=22222222(TON=Unknown, NPI=Unknown), > > Calling Translated=FALSE, Subscriber Type Str=Unknown, > FinalDestinationFlag=T > RUE, > > Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE > Present=TRUE, > > Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID > Transparent=FALS > E), Call Id=216094 > > *Nov 10 06:01:45.131: > //216094/D4D5BF13845B/CCAPI/cc_api_call_setup_ind_common: > > Set Up Event Sent; > > Call Info(Calling Number=31754250(TON=Unknown, NPI=Unknown, > Screening=User, > P > assed, Presentation=Allowed), > > Called Number=22222222(TON=Unknown, NPI=Unknown)) > > *Nov 10 06:01:45.131: > //216094/D4D5BF13845B/CCAPI/cc_process_call_setup_ind: > > Event=0x45661160 > > *Nov 10 06:01:45.131: //216094/D4D5BF13845B/CCAPI/ccCallSetContext: > > Context=0x48B4C074 > > *Nov 10 06:01:45.131: > //216094/D4D5BF13845B/CCAPI/cc_process_call_setup_ind: > > >>>>CCAPI handed cid 216094 with tag 0 to app > "_ManagedAppProcess_Default" > > *Nov 10 06:01:45.131: //216094/D4D5BF13845B/CCAPI/ccCallProceeding: > > Progress Indication=NULL(0) > > *Nov 10 06:01:45.135: //216094/D4D5BF13845B/CCAPI/ccCallSetupRequest: > > Destination=, Calling IE Present=TRUE, Mode=0, > > Outgoing Dial-peer=1, Params=0x46AD67F0, Progress Indication=ORIGINATING > SIDE > IS NON ISDN(3) > > *Nov 10 06:01:45.135: //216094/D4D5BF13845B/CCAPI/ccCheckClipClir: > > In: Calling Number=31754250(TON=Unknown, NPI=Unknown, Screening=User, > Passed, > Presentation=Allowed) > > *Nov 10 06:01:45.135: //216094/D4D5BF13845B/CCAPI/ccCheckClipClir: > > Out: Calling Number=31754250(TON=Unknown, NPI=Unknown, Screening=User, > Passed > , Presentation=Allowed) > > *Nov 10 06:01:45.135: //216094/D4D5BF13845B/CCAPI/ccCallSetupRequest: > > Destination Pattern=.T, Called Number=22222222, Digit Strip=FALSE > > *Nov 10 06:01:45.135: //216094/D4D5BF13845B/CCAPI/ccCallSetupRequest: > > Calling Number=31754250(TON=Unknown, NPI=Unknown, Screening=User, > Passed, Pre > sentation=Allowed), > > Called Number=22222222(TON=Unknown, NPI=Unknown), > > Redirect Number=, Display Info=31754250 > > Account Number=31754250, Final Destination Flag=TRUE, > > Guid=D4D5BF13-EBC6-11DF-845B-C1363B6B99C1, Outgoing Dial-peer=1 > > *Nov 10 06:01:45.135: > //216094/D4D5BF13845B/CCAPI/cc_api_display_ie_subfields: > > ccCallSetupRequest: > > cisco-username=31754250 > > ----- ccCallInfo IE subfields ----- > > cisco-ani=31754250 > > cisco-anitype=0 > > cisco-aniplan=0 > > cisco-anipi=0 > > cisco-anisi=1 > > dest=22222222 > > cisco-desttype=0 > > cisco-destplan=0 > > cisco-rdie=FFFFFFFF > > cisco-rdn= > > cisco-rdntype=0 > > cisco-rdnplan=0 > > cisco-rdnpi=0 > > cisco-rdnsi=0 > > cisco-redirectreason=-1 > > *Nov 10 06:01:45.135: > //216094/D4D5BF13845B/CCAPI/ccIFCallSetupRequestPrivate: > > Interface=0x456ACB10, Interface Type=6, Destination=, Mode=0x0, > > Call Params(Calling Number=31754250(TON=Unknown, NPI=Unknown, > Screening=User, > Passed, Presentation=Allowed), > > Called Number=22222222(TON=Unknown, NPI=Unknown), Calling > Translated=FALSE, > > Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing > Dial-peer=1, > Call Count On=FALSE, > > Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, > Appl > ication Call Id=) > > *Nov 10 06:01:45.139: > //216095/D4D5BF13845B/CCAPI/ccIFCallSetupRequestPrivate: > > SPI Call Setup Request Is Success; Interface Type=6, FlowMode=1 > > *Nov 10 06:01:45.139: //216095/D4D5BF13845B/CCAPI/ccCallSetContext: > > Context=0x46AD67A0 > > *Nov 10 06:01:45.139: //216094/D4D5BF13845B/CCAPI/ccSaveDialpeerTag: > > Outgoing Dial-peer=1 > > *Nov 10 06:01:45.183: //216095/D4D5BF13845B/CCAPI/cc_api_call_proceeding: > > Interface=0x456ACB10, Progress Indication=NULL(0) > > *Nov 10 06:01:45.443: //216095/D4D5BF13845B/CCAPI/cc_api_call_disconnected: > > Cause Value=1, Interface=0x456ACB10, Call Id=216095 > > *Nov 10 06:01:45.443: //216095/D4D5BF13845B/CCAPI/cc_api_call_disconnected: > > Call Entry(Responsed=TRUE, Cause Value=1, Retry Count=0) > > *Nov 10 06:01:45.443: //216095/D4D5BF13845B/CCAPI/ccCallSetAAA_Accounting: > > Accounting=0, Call Id=216095 > > *Nov 10 06:01:45.443: //216095/D4D5BF13845B/CCAPI/ccCallDisconnect: > > Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, > Disconnect > Ca > use=1) > > *Nov 10 06:01:45.443: //216095/D4D5BF13845B/CCAPI/ccCallDisconnect: > > Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1) > > *Nov 10 06:01:45.443: //216095/D4D5BF13845B/CCAPI/cc_api_get_transfer_info: > > Transfer Number Is Null > > *Nov 10 06:01:45.463: > //216095/D4D5BF13845B/CCAPI/cc_api_call_disconnect_done: > > Disposition=0, Interface=0x456ACB10, Tag=0x0, Call Id=216095, > > Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0) > > *Nov 10 06:01:45.463: > //216095/D4D5BF13845B/CCAPI/cc_api_call_disconnect_done: > > Call Disconnect Event Sent > > *Nov 10 06:01:45.463: //216094/D4D5BF13845B/CCAPI/ccCallDisconnect: > > Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, > Disconnect > Ca > use=0) > > *Nov 10 06:01:45.463: //216094/D4D5BF13845B/CCAPI/ccCallDisconnect: > > Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1) > > *Nov 10 06:01:45.471: > //216094/D4D5BF13845B/CCAPI/cc_api_call_disconnect_done: > > Disposition=0, Interface=0x451F6154, Tag=0x0, Call Id=216094, > > Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0) > > > > > > Please help. > > > > Thanks > > > > Raymond Chan > > Technical Manager (Hong Kong and Macau) > > Commverge Solutions Limited > > Mobile: +852 98403073 > > Direct: +852 29022420 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org <http://www.freeswitch.org/> > >

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No ring/can't answer during early media playback of wav
by Mario G 10 Nov '10

10 Nov '10

During early media the steps below work except the target stops ringing during step 3 and if the phone is picked up it does not answer the call, other extensions continue ringing. Is there a way to continue ringing the target while the wav is played to the caller so it can be answered? No luck on searching wiki and the few things I tried so far. Thanks, Mario G 1. I play 2 wav files 2. ring 17 seconds 3. play 2 wav files <- at this point the target phones stop ringing and cannot answer the call, after this point no problem. 4. set music on hold while bridging to cell phones. step 1 <action application="set" data="ringback=${us-ring}"/ <action application="execute_extension" data="2921 XML default"/> <<<-play wav based on caller id,no ringing of target step 2 <action application="set" data="call_timeout=17"/> <action application="bridge" data="group/users1@${domain_name}"/> steps 3. <<<-play wav based on caller id, ringing stops and can't answer <action application="execute_extension" data="2922 XML default"/> <action application="playback" data="../mvsounds/we_trying_please_hold.wav"/> step 4. <<<-ringing starts agsin and can answer <action application="set" data="ringback=$${hold_music}"/>

3 9

Issue with Invites without SDP
by SIC FS LIST 10 Nov '10

10 Nov '10

Hello, I have a working FS instance I am using as a redirect server (to serve up LNP requests). It works fine on calls with invites that have SDP and does not work with invites without SDP. I enabled 3pcc to true thinking that would fix the issue. Version info is FreeSWITCH Version 1.0.6 (hacked-20100921T052029Z). With the console log level set to debug the only thing I see is this message (just before returning a 480): freeswitch@lnpdal0001> 2010-11-10 08:58:50.949818 [CRIT] switch_core_state_machine.c:382 e390f5b0-ecda-11df-906a-13020c8dafd9 Timeout waiting for next instruction in CS_NEW! I also do not ever see FS to an xml http req (which is how we control the dialplan). freeswitch@lnpdal0001> recv 495 bytes from udp/[X.X.X.X]:5060 at 14:57:50.934431: ------------------------------------------------------------------------ INVITE sip:19033226103@X.X.X.X:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 Max-Forwards: 70 Contact: <sip:X.X.X.X:5060> To: <sip:19033226103@X.X.X.X:5060> From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 Call-ID: 993186-34928005397-80464(a)ens.com CSeq: 693501 INVITE Date: Tue, 09 Nov 2010 15:29:25 GMT Supported: 100rel User-Agent: ENSR3.0.63.0-IS1-RMRG2101-RG21-CPO11152 Content-Length: 0 ------------------------------------------------------------------------ send 336 bytes to udp/[X.X.X.X]:5060 at 14:57:50.935324: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060> Call-ID: 993186-34928005397-80464(a)ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Content-Length: 0 ------------------------------------------------------------------------ send 714 bytes to udp/[X.X.X.X]:5060 at 14:58:50.951429: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060>;tag=t7ycB64mX33mQ Call-ID: 993186-34928005397-80464(a)ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=81;text="INVALID_CALL_REFERENCE" Content-Length: 0 Remote-Party-ID: "19033226103" <sip:19033226103@X.X.X.X >;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ Here is the sofia profile: <profile name="external"> <aliases> <alias name="outbound"/> <alias name="nat"/>  </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="user-agent-string" value="lnpdal0001"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="enable-100rel" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="X.X.X.X"/> <param name="sip-ip" value="X.X.X.X"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="bind-params" value="transport=udp"/> <param name="tls" value="$${external_ssl_enable}"/>   <param name="inbound-bypass-media" value="true"/> <param name="inbound-late-negotiation" value="true"/> <param name="accept-blind-reg" value="false"/> <param name="accept-blind-auth" value="true"/> <param name="nonce-ttl" value="60"/> <param name="disable-transcoding" value="true"/> <param name="auth-calls" value="false"/> <param name="inbound-reg-force-matching-username" value="false"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_to"/> <param name="enable-3pcc" value="true"/> </settings> </profile> Any thoughts / ideas / help would be greatly appreciated. Thanks! SDR

2 6