August 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_php needed
by Christian Löschenkohl 09 Aug '09

09 Aug '09

hello i am working for an austrian voip carrier. for a few months i work with freeswitch and it is simply great. it solves our needs in many places (high volume, flexible, stable). the only thing i really miss is the avalibilty of php as a call control language. mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't that good (or even there :-) ). i know there is perl, i also implemented some applications (conference system with provisioning, inbound call routing to our application servers, some tests as pbx), but what should i say - perl and me aren't compatible in the end. the greatest thing for us would be that mod_php comes alive again with the functional state of mod_perl (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes) there is also a bounty entry about mod_php, to pay for this would also be an option and could be discussed. keep on with the excellent work and greetings from austria -- Ing. Christian Löschenkohl Technische Leitung, Forschung& Entwicklung VoIP xpirio Telekommunikation& Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

8 13

Fwd: strange behaviour with scheduled jobs, uuid_transfer hangs up one of the legs if freeswitch is not in the media path, lua freeswitch.Session(uuid) tries to call out if channel references by uuid does not exist any longer
by Benedikt Fraunhofer 08 Aug '09

08 Aug '09

Hello List, Hello *, First of all the usual excuses: sorry for the bad english and the long email, no native speaker and i really tried to make it shorter, but i guess this would result in even more "check back"s than it already does :) we're currently running in a weird "lockup"-scenario in our loadtests. Our setup is the following: three freeswitch servers, let's call them A(-leg), M(aster), and B(-leg) with the goal in mind to initiate calls on M which calls A, play some file, bridge to B, limit call length and play (different) prompts to A and B if they exceed that limit. (Note that A and B work fine, regardless of the amount of load we put on them) A and B are silly dialplan logic, accepting calls on a certain extension after a random delay and playing moh. Before calling playback to a localstream they call a lua script which schedules hangup somewhere in future (which works flawlessly) Calls are initiated on M using some hacked up loadgen-script issuing http requests like originate [sofiaSyntaxToExtensionOn_A] 6000 . The 6000 extension on M has the following (xml) dialplan which essentially does the following: ------ answer() ...playback file... ...set some callerid stuff set bypass_media bridge to extension 6009 on B ------ we use "execute_on_answer" on the b-leg to run a script which limits the length of the call (doesn't matter if it's done via "export nolocal" or "inlined" into the data part of the bridge application "{execute_on_answer=lua ...}") <action application="export" data="nolocal:execute_on_answer=lua lua/schedule-hangup.lua ${uuid}" /> the lua script "schedula-hangup.lua" does essentially the following: ------ api = freeswitch.API(); local res = api:execute("sched_api", "+10 none lua lua/c2c-hangup-timeout.lua " .. argv[1]); ------ the 10 seconds are just to speed up the time until it gets stuck. this is where things start to go wrong. if I comment out the call to the "schedule-hangup" script, everything works fine, even if it's under heavy load. c2c-hangup-timeout.lua does the following: ------------------ local sess = argv[1]; if(sess) then freeswitch.consoleLog("INFO", "c2c-hangup-timeout.lua for uuid " .. sess .. "\n"); api = freeswitch.API(); local stillValid = api:execute("uuid_getvar", sess .. " Dummy-DoesChannelExists"); if(stillValid:sub(1,4) == "-ERR") then log("session uuid " .. sess .. " disappeared (nothing bad)"); else -- this is important!!! Otherwise the aleg get's just hung up! api:execute("uuid_media", sess); api:execute("uuid_transfer", sess .. " -both timeout"); end else -- /if(sess) log("called with nil session?"); end -- /if(sess) ------------------ i guess this needs some explanation: we get the uuid of the channel as argument in argv[1]. We don't use local session = freeswitch.Session(uuid); since if the channel referenced by "uuid" does not exist any longer, freeswitch (or the lua bindings) try to interpret the uuid as an "originate string" and can't figure out how to call that. So we use a dummy api call to get some channel variable. If the channel does not exist any longer (A or B already hung up), we get an error message starting with "-ERR", otherwise the channel still exists (we get "_unset_" as the value, if it's not set) and we continue by getting freeswitch back in the media path (uuid_media) and then transferring both legs to an extension called "timeout" which plays some prompt and finally calls hangup(). If we don't do the uuid_media call, one of the legs gets hung up when we transfer them to the extension. This looks like the following on the console after issuing "uuid_transfer [uuid] -both timeout" (extensions are not the same as in our loadgen example above) -------------- 2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1334 Hangup (*) sofia/internal/1000 [CS_HIBERNATE] [BLIND_TRANSFER] 2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer sofia/internal/1004@192.168.179.177:5060 to XML[timeout@default] 2009-07-23 19:57:19.865703 [INFO] mod_dialplan_xml.c:310 Processing BFR1004->timeout in context default API CALL [uuid_transfer(73812082-77b1-11de-b9f8-a10bb0eb9f69 -both timeout)] output: +OK 2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer (**) sofia/internal/1000 to XML[timeout@default] 2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1084 Session 60 (sofia/internal/1000) Ended 2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1086 Close Channel sofia/internal/1000 [CS_DESTROY] ----------- note that it first does Hangup (denoted by *, no that's not an asterisk :) on extension 1000 and then tries to Transfer (**) the hung up channel to the dial plan. this could be the same as in an earlier post to the list "SIP re-invite / bypass_media // Phillip Jones // Wed, 01 Jul 2009 13:30:53 -0700)" This is why we do not directly call sched_transfer() but call a script in between to do the uuid_media() call. I couldn't figure out how to call that directly from the xml dialplan and/or how to check if the channel still exists. so... after using uuid_media(), both legs are transferred without an (intermediate|bogus) hangup() call. This only works fine if we've few concurrent calls. There is no magic borderline where it starts to refuse work. Some of the Symptoms are: traffic decreased to zero as no new channels are successfully brought up, some of the signaling traffic is not ACKed or OKed, scheduled jobs are not run. if i read the output of "show channels" correctly, they're all stuck in different applications like hangup(), some are calling lua but most of them are in signaling_bridge(). Freeswitch is still responding on the console and there's almost no load on the machine (no busy polling or some other kind of running amok). if i kill one of them using uuid_kill() or kill all of them using"fsctl hupall" i get "Task was executed late by 866 seconds 12379 sched_api_function (none)" messages and the usual cleanup takes place. As a quick hack i tried to schedule a uuid_kill() call 20 seconds after the scheduling call to the lua script but that job is not executed either. So what am I doing wrong? Is it some deadlock where uuid_media() and uuid_transfer() are waiting for the other to finish? Or some other silly simple thing i missed? Thx in advance Benedikt.

3 5

Skypiax, Skype endpoint and trunk, revamped
by Giovanni Maruzzelli 08 Aug '09

08 Aug '09

Ciao FreeSWITCHers, please have a look at the much changed wiki page: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and checkout and test the code in svn. Much has happened, various bug fixes and features added. Most notable: - multiple instances of the same Skype username on Linux (eg: running 20 concurrent channels as "Bob" Skype user) - adding and removing interfaces on the fly (patch sent by Muhammad Shahzad) - easier creation of Skype clients configuration directory - reduced latency - better robustness - running as Windows Service - customized ALSA driver for more devices with less IRQs and context switches - custom kernel tickless and 100HZ (eg. solves high load problems in CentOS and in virtualization) - interactive command line client for prototyping Also, please note that ALSA drivers version 1.0.20 seems to be much more stable in our kind of usage (snd-dummy). Various other enhancements will come, but in the mean time please give feedback on the current svn code (we want to be robust for the 1.0.4 Release :-) ) See you all at www.cluecon.com, talk on Skypiax August 4th at 4.30 pm ! -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039

3 2

Calling Multiple Destinations with Failover
by Phillip Jones 08 Aug '09

08 Aug '09

Hi there, I am trying to implement a scenario where I can terminate calls to multiple destinations AND have termination carrier fail over. Currently I can see how to do one or the other. But not both. Multiple destinations is easy: <action application="bridge" data="sofia/SIP_PROVIDER_1/6095551234,sofia/SIP_PROVIDER_1/7325551234"/> Failover appears to use the same mechanism: <action application="bridge" data="sofia/gateway/primary/blah|sofia/gateway/secondary/blah"/> I can not get my head around how use these together, such that: - 6095551234 is dialed through SIP_PROVIDER_1 - if NO_ROUTE_DESTINATION then dial 6095551234 through SIP_PROVIDER_2 - Called party does not answer 7325551234 is dialed through SIP_PROVIDER_1 - if NO_ROUTE_DESTINATION then dial 7325551234 through SIP_PROVIDER_2 - Called party answers This must be a fairly common requirement so any ideas on what I might be missing would be very welcome. Thanks Phillip Jones

3 7

DTMF disable a few secs after call starts
by Frank ＠ Impact 08 Aug '09

08 Aug '09

FS is in the media path of an IVR call. At the moment, the call is ulaw with DTMF in the audio I think coming into FS and leaving FS. The call is coming from an Asterisk server and going to an Asterisk server. Is there a way to disable FS from passing DTMF at some point in the call? For example, after 15 seconds, is there a way to get FS to stop passing DTMF events? Would I have to try to force asterisk to use rfc2833 when sending the call to FS and when accepting it back from FS?

2 1

State of originated call
by Max Bridgewater 08 Aug '09

08 Aug '09

Hi, using javascript, i do originate the call this way: Session s= new Session(originateStr); >From this point, is it possible to know what states the call is going through? In a previous message it was suggested that variable_originate_disposition would give me the response code. Now, how to i use this in practice in a script? How do i for instance retrieve a 180 response code when rining is hapening on the remote end? Thanks, Max.

2 1

FreeSwitch doesn't play music on hold forbriged channel
by Kozak Vladimir 08 Aug '09

08 Aug '09

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.

2 1

Call transfer/forward PSTN-SIP-PSTN
by Vladimir Rodionov 08 Aug '09

08 Aug '09

Good morning, This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes through PSTN Gateway (1) to freeSWITCH application server (AS) (2). AS does some logic and transfers call (or forward) out of Voip provider network to another PSTN number User2. This is call bridge UA1 (PSTN) - -> UA2 (PSTN) - - - (1) - (4) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH -> This is what I want to acomplish (4) UA1 (PSTN) ------------------------------- -> UA2 (PSTN) - - (1) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH -> 1. Can it be implemented in FreeSWITCH? 2. Does anybody know Voip providers which support out of network call transfer/forwarding to PSTN? TIA -Vladimir Rodionov

3 6

Lua on Windows and additional modules
by Vladimir Rodionov 08 Aug '09

08 Aug '09

Good evening, This is newbie question. The FreeSWITCH lua module does not support sockets and sql out of box that is why I just installed LuaBinaries (including socket, sql modules). My dev environment is Win XP not Linux/Unix. I am trying to understand what will happen when lua_module get this: require "socket" or require "luasql.mysql" ? How does lua_module look up additional lua modules on Windows platform? Do I have to set some env variables? TIA -Vladimir Rodionov

2 2

question on att_xfer
by Rocco Lucente 08 Aug '09

08 Aug '09

hi, during a call made with att_xfer, you can resume the user A on ring? Let me explain better, with bind_meta_app and att_xfer can do this: during a bridge if B dials *1 (for example), A is parked and B can call C. After C to answer, B can to dial # and return in conversation with A. B could resume A while is calling C (for example because I noticed that I've dialed an incorrect number)? I tried with execute_on_ring, but without result. Any ideas? Thank you and good job to all. -- Rocco Lucente

1 0