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Unable to make trunk on CentOS 5.3
by Max Bridgewater 11 Aug '09

11 Aug '09

Hi get the following messages while running "make" on CentOS 5.3 (Of course after bootstrap and configure). os_Linux_x86_64.s: Assembler messages: os_Linux_x86_64.s:46: Error: bad register name `%rdi)' os_Linux_x86_64.s:61: Error: bad register name `%rdi)' os_Linux_x86_64.s:75: Error: bad register name `%rdi)' os_Linux_x86_64.s:89: Error: bad register name `%rdi)' make[10]: *** [os_Linux_x86_64.o] Error 1 make[9]: *** [export] Error 2 make[8]: *** [export] Error 2 make[7]: *** [export] Error 2 make[6]: *** [export] Error 2 make[5]: *** [/home/installs/fstrunk/libs/js/libjs.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_spidermonkey-all] Error 1 make[2]: *** [all-recursive] Error 1 Any idea? Max.

4 15

question about latest version of mod_limit
by mark morreny 10 Aug '09

10 Aug '09

Hello, I have the following setup in the dialplan. Then, I fire up sipp to send 5calls/s and I expect to get limit-pass=false in most of the INFO output. However, I am getting all "limit-pass=pass". Does anyone know what is wrong with my dialplan? <context name="internal"> <extension name="test-limit"> <condition field="destination_number" expression="^9(.*)$"> <action application="limit_hash_execute" data="192.168.1.102 6000 1/1 transfer to-next" /> <action application="set" data="limit-pass=false"/> <action application="info" /> <action application="hangup" /> </condition> </extension> <extension name="to-next"> <condition field="destination_number" expression="^to-next$"> <action application="set" data="limit-pass=true"/> <action application="info" /> <action application="hangup" /> </condition> </extension> </context>

3 2

VoiceMail transcription
by Pete Mueller 10 Aug '09

10 Aug '09

2 1

Freeswitch :: VoIP issues using ILBC audio codec
by David Nembrot 10 Aug '09

10 Aug '09

Hello world, I've got two FS servers configured as gateways for each other and I'm currently testing the telephony. Usinge the ILBC audio codec, I figured out that one of the FS servers doesn't forward RTP streams correctly to the other server. Here is its status-quo: INPUT = proper ILBC payload type (97 or 108) OUTPUT = unknown payload type (97 or 102) I've already changed the parameters in internal.xml & external.xml: <param name="inbound-codec-negotiation" value="greedy"/> <param name="disable-transcoding" value="true"/> When dialing out, I also use the following syntax:{absolute_codec_string='GSM,PCMU'}sofia/gateway/mygateway/mynumber Is there another thing to do to have proper ILBC streams passing through the gateways ? Thanking y'all in advance ;) BR, David N. Hello world, I've got two FS servers configured as gateways for each other and I'm currently testing the telephony. Usinge the ILBC audio codec, I figured out that one of the FS servers doesn't forward RTP streams correctly to the other server. Here is its status-quo: INPUT = proper ILBC payload type (97 or 108) OUTPUT = unknown payload type (97 or 102) I've already changed the parameters in internal.xml and also in external.xml: <param name="inbound-codec-negotiation" value="greedy"/> <param name="disable-transcoding" value="true"/> Is there another thing to do to have proper ILBC streams passing through the gateways ? Thanking y'all in advance ;) BR, David N.

2 1

Freeswitch pre-compiled for Solaris 10/x86
by vmorales 10 Aug '09

10 Aug '09

Hello, Does anyone have, or know where to get, a pre-compiled copy of FreeSwitch for Solaris 10/x86? # uname -a SunOS hrndvsoi-zm01 5.10 Generic_137112-07 i86pc i386 i86pc I'm stuck trying to run make/gmake/\/opt/gnu/bin/make: <snip> "make" results in: make: Fatal error: Command failed for target `all-recursive' Current working directory /home/vmorales/freeswitch *** Error code 1 make: Fatal error: Command failed for target `all' </snip> "gmake" & "/opt/gnu/bin/make" result in: <snip> creating libfreeswitch.la (cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la libfreeswitch.la) gcc -I/home/vmorales/freeswitch/src/include -I/home/vmorales/freeswitch/libs/libteletone/src -fPIC -Werror -g -ggdb -DPATH_MAX=2048 -g -m32 -I/usr/sfw/include -Wall -std=c99 -pedantic -m32 -o .libs/freeswitch freeswitch-switch.o -L/usr/sfw/lib -lm ./.libs/libfreeswitch.so /home/vmorales/freeswitch/libs/apr/.libs/libapr-1.a -L/home/vmorales/freeswitch/libs/srtp /usr/sfw/lib/libstdc++.so libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt -ldl -lnsl -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lcurses -lsocket -R/home/vmorales/freeswitch-build/lib -R/usr/sfw/lib Undefined first referenced symbol in file XML_Parse ./.libs/libfreeswitch.so XML_ParserCreate ./.libs/libfreeswitch.so XML_ErrorString ./.libs/libfreeswitch.so herror ./.libs/libfreeswitch.so XML_SetUserData ./.libs/libfreeswitch.so XML_ParserFree ./.libs/libfreeswitch.so XML_GetErrorCode ./.libs/libfreeswitch.so XML_SetCharacterDataHandler ./.libs/libfreeswitch.so XML_SetElementHandler ./.libs/libfreeswitch.so ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch collect2: ld returned 1 exit status gmake[2]: *** [freeswitch] Error 1 gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 </snip> Thanks in advance for any information provided. Vladimir

3 3

Error trying to use PHP ESL
by Nicolas Brenner 10 Aug '09

10 Aug '09

Hi, I'm trying to get started with the ESL using PHP. I compiled the ESL, then phpmod according to the wiki instructions, but then when I try the examples in the libs/esl/php dir, they fail saying: PHP Fatal error: Cannot redeclare ESLconnection::__construct() in /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132 Checking ESL.php on line 132, I see there are several different declarations for the function __construct() with different parameters each, which makes sense, but doens't work. I am using PHP 5.1.6, is there a required minimum higher than that or something? What could be the problem? Thanks! Nicolas

5 6

can freeswitch group_call to gateway extension ?
by daqiang wang 10 Aug '09

10 Aug '09

Hello every one : I tested FS. and when I use group_call. But I can't call the extension in group and not registered in FS. Why ? and What ? thanks .

2 2

Fw: FreeSwitch doesn't play music on hold forbriged channel
by Kozak Vladimir 10 Aug '09

10 Aug '09

2009-08-12 03:02:03 [DEBUG] switch_core_session.c:706 switch_core_session_queue_private_event() Send signal sofia/internal/sip:1000@172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK] 2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:228 audio_bridge_thread() sofia/internal/sip:1000@172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes receive message [UNBRIDGE] 2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000@172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK] 2009-08-12 03:02:03 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/internal/sip:1000@172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes Command Execute hold() 2009-08-12 03:02:03 [DEBUG] switch_ivr.c:1054 switch_ivr_hold() sofia/internal/sip:1000@172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes receive message [HOLD] 2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000@172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK] 2009-08-12 03:02:03 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/internal/sip:1000@172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes entering state [calling] 2009-08-12 03:02:03 [WARNING] switch_ivr_async.c:2239 switch_ivr_broadcast() Channel [sofia/internal/1001@172.26.200.250][local_stream://moh] already broadcasting...broadcast aborted 2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:231 audio_bridge_thread() sofia/internal/sip:1000@172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes receive message [BRIDGE] 2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000@172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK] 2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:233 audio_bridge_thread() Send signal sofia/internal/1001(a)172.26.200.250 [BREAK] 2009-08-12 03:02:03 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/internal/sip:1000@172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes entering state [ready] 2009-08-12 03:02:03 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 6 3 IN IP4 172.26.10.39 s=CounterPath eyeBeam 1.5 c=IN IP4 172.26.10.39 t=0 0 m=audio 15712 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=recvonly 2009-08-12 03:02:03 [DEBUG] sofia_glue.c:2483 sofia_glue_negotiate_sdp() Our existing sdp is still good [PCMU 172.26.10.39:15712], let's keep it. 2009-08-12 03:02:03 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 ----- Original Message ----- From: Michael Jerris To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, August 09, 2009 12:19 AM Subject: Re: [Freeswitch-users] FreeSwitch doesn't play music on holdforbriged channel What do the debug logs on fs say when you try to put the call on hold? Mike On Aug 6, 2009, at 1:01 PM, Kozak Vladimir wrote: The scenario is the following: FS User A dial an extension Extention opens outbound socket channel to my application My application bridges the call to FS User B The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id User B is placed on hold but no music on hold is played to the caller (User A) I have outbound socket channel and the following sequence of commands/event: listening on [any] 8084 ... connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000 connect ------------------------------------------------------------------------------ _______________________________________________ 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 The scenario is the following: FS User A dial an extension Extention opens outbound socket channel to my application My application bridges the call to FS User B The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id User B is placed on hold but no music on hold is played to the caller (User A) I have outbound socket channel and the following sequence of commands/event: listening on [any] 8084 ... connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000 connect myevents SendMsg call-command: execute execute-app-name: bridge execute-app-arg:user/1000@uat.agent.starpoundtech.net Channel-Username: 1001 Channel-Dialplan: XML Channel-Caller-ID-Name: 1001 Channel-Caller-ID-Number: 1001 Channel-Network-Addr: 172.26.10.39 Channel-Destination-Number: 6666 Channel-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2 Channel-Source: mod_sofia Channel-Context: default Channel-Channel-Name: sofia/internal/1001%40172.26.200.250 Channel-Profile-Index: 1 Channel-Profile-Created-Time: 1249142681680114 Channel-Channel-Created-Time: 1249142681680114 Channel-Channel-Answered-Time: 0 Channel-Channel-Progress-Time: 0 Channel-Channel-Progress-Media-Time: 1249142681809352 Channel-Channel-Hangup-Time: 0 Channel-Channel-Transfer-Time: 0 Channel-Screen-Bit: true Channel-Privacy-Hide-Name: false Channel-Privacy-Hide-Number: false Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1001%40172.26.200.250 Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2 Call-Direction: inbound Answer-State: early Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 1001 Caller-Dialplan: XML Caller-Caller-ID-Name: 1001 Caller-Caller-ID-Number: 1001 Caller-Network-Addr: 172.26.10.39 Caller-Destination-Number: 6666 Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1001%40172.26.200.250 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1249142681680114 Caller-Channel-Created-Time: 1249142681680114 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 1249142681809352 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false variable_sip_received_ip: 172.26.10.39 variable_sip_received_port: 13488 variable_sip_via_protocol: udp variable_sip_authorized: true variable_sip_mailbox: 1001 variable_sip_auth_username: 1001 variable_sip_auth_realm: 172.26.200.250 variable_mailbox: 1001 variable_toll_allow: domestic,international,local variable_accountcode: 1001 variable_user_context: default variable_effective_caller_id_name: Extension%201001 variable_effective_caller_id_number: 1001 variable_outbound_caller_id_name: StarPound%20FreeSWITCH variable_outbound_caller_id_number: 0000000000 variable_callgroup: techsupport variable_sip_from_user: 1001 variable_sip_from_uri: 1001%40172.26.200.250 variable_sip_from_host: 172.26.200.250 variable_sip_from_user_stripped: 1001 variable_sip_from_tag: bd11f93c variable_sofia_profile_name: internal variable_sip_req_user: 6666 variable_sip_req_uri: 6666%40172.26.200.250 variable_sip_req_host: 172.26.200.250 variable_sip_to_user: 6666 variable_sip_to_uri: 6666%40172.26.200.250 variable_sip_to_host: 172.26.200.250 variable_sip_contact_user: 1001 variable_sip_contact_port: 13488 variable_sip_contact_uri: 1001%40172.26.10.39%3A13488 variable_sip_contact_host: 172.26.10.39 variable_channel_name: sofia/internal/1001%40172.26.200.250 variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc. variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117 variable_sip_via_host: 172.26.10.39 variable_sip_via_port: 13488 variable_sip_via_rport: 13488 variable_max_forwards: 70 variable_presence_id: 1001%40172.26.200.250 variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A variable_remote_media_ip: 172.26.10.39 variable_remote_media_port: 29826 variable_read_codec: PCMU variable_read_rate: 8000 variable_write_codec: PCMU variable_write_rate: 8000 variable_use_profile: nat variable_record_stereo: true variable_transfer_fallback_extension: operator variable_numbering_plan: US variable_default_areacode: 918 variable_default_gateway: example.com variable_user_name: default variable_domain_name: 172.26.200.250 variable_current_application_data: 172.26.200.251%3A8084%20async%20full variable_current_application: socket variable_socket_host: 172.26.200.251 variable_local_media_ip: 172.26.200.250 variable_local_media_port: 29370 variable_endpoint_disposition: EARLY%20MEDIA variable_sip_nat_detected: true Content-Type: command/reply Reply-Text: %2BOK%0A Socket-Mode: async Control: full Content-Type: command/reply Reply-Text: +OK Events Enabled Content-Type: command/reply Reply-Text: +OK Content-Length: 1541 Content-Type: text/event-plain Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1001%40172.26.200.250 Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2 Call-Direction: inbound Answer-State: early Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 1001 Caller-Dialplan: XML Caller-Caller-ID-Name: 1001 Caller-Caller-ID-Number: 1001 Caller-Network-Addr: 172.26.10.39 Caller-Destination-Number: 6666 Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1001%40172.26.200.250 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1249142681680114 Caller-Channel-Created-Time: 1249142681680114 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 1249142681809352 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Application: bridge Application-Data: user/1000%40uat.agent.starpoundtech.net Event-Name: CHANNEL_EXECUTE Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b FreeSWITCH-Hostname: centos4-4-vm FreeSWITCH-IPv4: 172.26.200.250 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-08-01%2020%3A04%3A51 Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A51%20GMT Event-Date-Timestamp: 1249142691754598 Event-Calling-File: switch_core_session.c Event-Calling-Function: switch_core_session_exec Event-Calling-Line-Number: 1333 Content-Length: 5242 Content-Type: text/event-plain Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1001%40172.26.200.250 Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2 Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 1001 Caller-Dialplan: XML Caller-Caller-ID-Name: 1001 Caller-Caller-ID-Number: 1001 Caller-Network-Addr: 172.26.10.39 Caller-Destination-Number: 6666 Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1001%40172.26.200.250 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1249142681680114 Caller-Channel-Created-Time: 1249142681680114 Caller-Channel-Answered-Time: 1249142692414509 Caller-Channel-Progress-Time: 1249142691898434 Caller-Channel-Progress-Media-Time: 1249142681809352 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Other-Leg-Username: 1001 Other-Leg-Dialplan: XML Other-Leg-Caller-ID-Name: Extension%201001 Other-Leg-Caller-ID-Number: 1001 Other-Leg-Network-Addr: 172.26.10.39 Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119 Other-Leg-Source: mod_sofia Other-Leg-Context: default Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes Other-Leg-Screen-Bit: true Other-Leg-Privacy-Hide-Name: false Other-Leg-Privacy-Hide-Number: false variable_sip_received_ip: 172.26.10.39 variable_sip_received_port: 13488 variable_sip_via_protocol: udp variable_sip_authorized: true variable_sip_mailbox: 1001 variable_sip_auth_username: 1001 variable_sip_auth_realm: 172.26.200.250 variable_mailbox: 1001 variable_toll_allow: domestic,international,local variable_accountcode: 1001 variable_user_context: default variable_effective_caller_id_name: Extension%201001 variable_effective_caller_id_number: 1001 variable_outbound_caller_id_name: StarPound%20FreeSWITCH variable_outbound_caller_id_number: 0000000000 variable_callgroup: techsupport variable_sip_from_user: 1001 variable_sip_from_uri: 1001%40172.26.200.250 variable_sip_from_host: 172.26.200.250 variable_sip_from_user_stripped: 1001 variable_sip_from_tag: bd11f93c variable_sofia_profile_name: internal variable_sip_req_user: 6666 variable_sip_req_uri: 6666%40172.26.200.250 variable_sip_req_host: 172.26.200.250 variable_sip_to_user: 6666 variable_sip_to_uri: 6666%40172.26.200.250 variable_sip_to_host: 172.26.200.250 variable_sip_contact_user: 1001 variable_sip_contact_port: 13488 variable_sip_contact_uri: 1001%40172.26.10.39%3A13488 variable_sip_contact_host: 172.26.10.39 variable_channel_name: sofia/internal/1001%40172.26.200.250 variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc. variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117 variable_sip_via_host: 172.26.10.39 variable_sip_via_port: 13488 variable_sip_via_rport: 13488 variable_max_forwards: 70 variable_presence_id: 1001%40172.26.200.250 variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A variable_remote_media_ip: 172.26.10.39 variable_remote_media_port: 29826 variable_read_codec: PCMU variable_read_rate: 8000 variable_write_codec: PCMU variable_write_rate: 8000 variable_use_profile: nat variable_record_stereo: true variable_transfer_fallback_extension: operator variable_numbering_plan: US variable_default_areacode: 918 variable_default_gateway: example.com variable_user_name: default variable_domain_name: 172.26.200.250 variable_socket_host: 172.26.200.251 variable_local_media_ip: 172.26.200.250 variable_local_media_port: 29370 variable_endpoint_disposition: EARLY%20MEDIA variable_current_application_data: user/1000%40uat.agent.starpoundtech.net variable_current_application: bridge variable_dialed_user: 1000 variable_dialed_domain: uat.agent.starpoundtech.net variable_originate_disposition: failure variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119 variable_sip_redirect_contact_user_0: 1000 variable_sip_redirect_contact_host_0: 172.26.10.39 variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A variable_sip_nat_detected: true Event-Name: CHANNEL_ANSWER Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b FreeSWITCH-Hostname: centos4-4-vm FreeSWITCH-IPv4: 172.26.200.250 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-08-01%2020%3A04%3A52 Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT Event-Date-Timestamp: 1249142692414509 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_perform_mark_answered Event-Calling-Line-Number: 1776 Content-Length: 5233 Content-Type: text/event-plain Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1001%40172.26.200.250 Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2 Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 1001 Caller-Dialplan: XML Caller-Caller-ID-Name: 1001 Caller-Caller-ID-Number: 1001 Caller-Network-Addr: 172.26.10.39 Caller-Destination-Number: 6666 Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1001%40172.26.200.250 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1249142681680114 Caller-Channel-Created-Time: 1249142681680114 Caller-Channel-Answered-Time: 1249142692414509 Caller-Channel-Progress-Time: 1249142691898434 Caller-Channel-Progress-Media-Time: 1249142681809352 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Other-Leg-Username: 1001 Other-Leg-Dialplan: XML Other-Leg-Caller-ID-Name: Extension%201001 Other-Leg-Caller-ID-Number: 1001 Other-Leg-Network-Addr: 172.26.10.39 Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119 Other-Leg-Source: mod_sofia Other-Leg-Context: default Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes Other-Leg-Screen-Bit: true Other-Leg-Privacy-Hide-Name: false Other-Leg-Privacy-Hide-Number: false variable_sip_received_ip: 172.26.10.39 variable_sip_received_port: 13488 variable_sip_via_protocol: udp variable_sip_authorized: true variable_sip_mailbox: 1001 variable_sip_auth_username: 1001 variable_sip_auth_realm: 172.26.200.250 variable_mailbox: 1001 variable_toll_allow: domestic,international,local variable_accountcode: 1001 variable_user_context: default variable_effective_caller_id_name: Extension%201001 variable_effective_caller_id_number: 1001 variable_outbound_caller_id_name: StarPound%20FreeSWITCH variable_outbound_caller_id_number: 0000000000 variable_callgroup: techsupport variable_sip_from_user: 1001 variable_sip_from_uri: 1001%40172.26.200.250 variable_sip_from_host: 172.26.200.250 variable_sip_from_user_stripped: 1001 variable_sip_from_tag: bd11f93c variable_sofia_profile_name: internal variable_sip_req_user: 6666 variable_sip_req_uri: 6666%40172.26.200.250 variable_sip_req_host: 172.26.200.250 variable_sip_to_user: 6666 variable_sip_to_uri: 6666%40172.26.200.250 variable_sip_to_host: 172.26.200.250 variable_sip_contact_user: 1001 variable_sip_contact_port: 13488 variable_sip_contact_uri: 1001%40172.26.10.39%3A13488 variable_sip_contact_host: 172.26.10.39 variable_channel_name: sofia/internal/1001%40172.26.200.250 variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc. variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117 variable_sip_via_host: 172.26.10.39 variable_sip_via_port: 13488 variable_sip_via_rport: 13488 variable_max_forwards: 70 variable_presence_id: 1001%40172.26.200.250 variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A variable_remote_media_ip: 172.26.10.39 variable_remote_media_port: 29826 variable_read_codec: PCMU variable_read_rate: 8000 variable_write_codec: PCMU variable_write_rate: 8000 variable_use_profile: nat variable_record_stereo: true variable_transfer_fallback_extension: operator variable_numbering_plan: US variable_default_areacode: 918 variable_default_gateway: example.com variable_user_name: default variable_domain_name: 172.26.200.250 variable_socket_host: 172.26.200.251 variable_local_media_ip: 172.26.200.250 variable_local_media_port: 29370 variable_current_application_data: user/1000%40uat.agent.starpoundtech.net variable_current_application: bridge variable_dialed_user: 1000 variable_dialed_domain: uat.agent.starpoundtech.net variable_sip_redirect_contact_user_0: 1000 variable_sip_redirect_contact_host_0: 172.26.10.39 variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A variable_sip_nat_detected: true variable_endpoint_disposition: ANSWER variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119 variable_originate_disposition: SUCCESS Event-Name: CHANNEL_BRIDGE Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b FreeSWITCH-Hostname: centos4-4-vm FreeSWITCH-IPv4: 172.26.200.250 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-08-01%2020%3A04%3A52 Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT Event-Date-Timestamp: 1249142692414509 Event-Calling-File: switch_ivr_bridge.c Event-Calling-Function: switch_ivr_multi_threaded_bridge Event-Calling-Line-Number: 828 SendMsg 94b59a38-57c4-4703-9c6e-9985d832d119 call-command: execute execute-app-name: hold Content-Type: command/reply Reply-Text: +OK - I don't see the variable hold_music ... did you remove it? - I didn't. Moreover, I tried to set it explicitly using api uuid_setvar. ------------------------------------------------------------------------------ _______________________________________________ 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

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Re: [Freeswitch-users] Softphone control
by Artem Vasiliev 10 Aug '09

10 Aug '09

No, I don't want to make softphone from FreeSwitch I have FS and several users with eyeBeam softphones. I need to control those eyeBeams >You can run FreeSWITCH as a softphone and control it. >http://wiki.freeswitch.org/wiki/Freeswitch_softphone >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com> >> Hi >> >> I have FreeSwitch and external application, which communicates to it via >> event socket - listens for events for certain number and gives some >> commands. >> Is it possible for this application to control client softphones, for >> example, make them answer or hold, using the event socket or other >> FreeSwitch capabilities? >>

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New to Freeswitch - some help needed
by Alan Chandler 09 Aug '09

09 Aug '09

I apologize, as my first post to this list, that I ask a detailed set of questions, but I have spend some time looking at all the docs and can't get what I need to do completely sorted in my head. I am definitely one who likes to UNDERSTAND what is happening rather than follow blank recipies, so please bear with me as I try understand all the details. I do understand about networking, NAT etc - but I am new to SIP/RTP and in particular what I think is a double NAT problem Firstly - what am I trying to achieve: I am in the UK and have a small home network behind a D-Link DIR-100 Router/NAT/Firewall one of those machines, running Debian Lenny, acts as my main server for everything (and in an earlier incarnation was the firewall/router/nat box too - I only say this is because I had all this working using Asterisk a year or so ago, but with this important difference in configuration). Many of the ports on the firewall are port forwarded to this machine. I have set Freeswitch up on this server to act as a small voip pbx for the home - but MORE IMPORTANTLY - to enable my daughter from her house to talk to us. At my house locally I have a Linksys PAP2T two phone SIP box - and that is working with Freeswitch's default configuration (I set up to be 1000 and 1001 and used all the facilities). I will later add a Linksys SPA 3102 - although I DO NOT intend to use its facility to bridge to the normal phone network. My daughter, living in another house, also has a Nat box (unknown - its part of her ADSL modem/router/wireless access point) and also has a PAP2T which she will connect to the her network. This will be her phone. There is a family relation living in Australia who will load up a whatever softphone that we tell him to use. I expect, but don't know, that he will behind a NAT box too. Later, I have some friends in the USA that I might wish to add it too - especially so that we can hold some teleconferences. They will have a mixture of Windows and MACs, and I will need to recommend softphone clients for them. I want to set this up as a small private voice network, so anyone can ring anyone else. I will add fancy facilities such as conferencing and voicemail later - I just want to get the basics working first. Secondly I installed a stun client on my home machine and ran it against stun.freeswitch.org. It reported:- Primary: Independent Mapping, Independent Filter, preserves ports, no hairpin But I have no idea what this means - I can't find any clear statement via googling for it - how this set of answers maps to the different types of NAT that might be required to get this to all work. CAN SOMEONE ENLIGHTEN me please. Thirdly I have set up a sip profile called "double nat" from the recipe in the wiki. This defines the SIP port to be 5090. However, what I DO NOT UNDERSTAND is how the PAP2T box in my daughters house will initiate a connection to my server. Presumably, I have to port forward 5090 from the nat box to my server. IS THAT CORRECT? I also assume I will have to tell her to use STUN (I believe this is an option on the PAP2T) Fourthly If I understand SIP correctly, it just initiates the session and the two end points then communicate directly via RTP. What I don't understand is how does a session transition from SIP to RTP via the connection set up in the the first phase (in terms of passing through the NAT boxes). In particular WHICH OF THE TEST RESULTS from my stun client indicate it will do the right thing. (I am going to take a laptop to my daughters house with a stun client in to test her network this weekend). Could someone explain please. Fifthly Is there a recommended SIP softphone with all the right facilities (STUN support?) that works on MAC and WINDOWS (I only use linux myself). Apologies for the length of this. I am eager to get the answers so I can use an opportunity this weekend to get it working. -- Alan Chandler http://www.chandlerfamily.org.uk

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