July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

bridge: early media: call waiting
by Anita Hall 06 Jul '12

06 Jul '12

Hi I have been using early media fail-over while bridging the call to multiple endpoints. For example, this is taken from the FreeSWITCH cookbook and works fine for user_busy and destination_out_of_order. {ignore_early_media=true,originate_continue_on_timeout=true,call_timout=60,monitor_early_media_fail=user_busy:2:480+620!destination_out_of_order:2:1776.7} But, it does not work when the telco sends a "call waiting" frequency. What should be the option for call waiting? TIA. regards, Anita

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How to send a tigger and monitoring
by Luis Daniel Lucio Quiroz 06 Jul '12

06 Jul '12

Allo! Just Wondering if there is an easy way to do this: a) when we got an outgoing call that meets a regexp to send a tigger, like an email or a nagios push alert b) how to measure a counter for concurrent calls,and total calls per day. I'm in the willling to do a nagios pluging for this but i i dont know how to grab data. Regards, LD

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Re: [Freeswitch-users] How to send a tigger and monitoring
by Peter Olsson 06 Jul '12

06 Jul '12

ESL is probably the best approach. Read more here; http://wiki.freeswitch.org/wiki/Esl http://wiki.freeswitch.org/wiki/Mod_event_socket /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Luis Daniel Lucio Quiroz Skickat: den 6 juli 2012 13:55 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] How to send a tigger and monitoring Allo! Just Wondering if there is an easy way to do this: a) when we got an outgoing call that meets a regexp to send a tigger, like an email or a nagios push alert b) how to measure a counter for concurrent calls,and total calls per day. I'm in the willling to do a nagios pluging for this but i i dont know how to grab data. Regards, LD _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 !DSPAM:4ff6d16a32761304020267!

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How to get hangup_time, billsec variables inside script
by Neo Cheema 06 Jul '12

06 Jul '12

Hi All, I'm using Lua script to handle calls. For logging purposes, I would want to get hangup_time, billsec info in the script. However, the code doesn't work as expected. local new_session = freeswitch.Session(call_to_create_new_session, session) if new_session:ready() then new_session:streamFile("/home/ubuntu/sounds/ward5_1.wav") new_session:hangup(); local created_time = new_session:getVariable('created_time') //Correct value received local answered_time = new_session:getVariable('answered_time') // Correct value receive local hangup_time = new_session:getVariable('hangup_time') // NULL received. Problem!! Is there anyway to get this info from Inside the script? I understand that I can get this info from xml_cdr etc, but is it possible to get these values inside the script itself? Regards

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Getting started wiki page & external gateway: do I understand correctly?
by Jim 06 Jul '12

06 Jul '12

Hi list, More or less beginner, home situation; trying to set up fs (again).. I've got 2 questions: 1. Posting here to make sure I understand before modifying the wiki. Could you please correct me if I'm wrong? Over here http://wiki.freeswitch.org/wiki/Getting_Started_Guide#External it says "The External (formerly "outbound") profile handles outbound registrations to a SIP provider." However, earlier on it also mentions you can let external devices (i.e. user phones in their own networks) register with that profile... so I'd change this to: "The External (formerly "outbound") profile also handles outbound registrations to a SIP provider." Then this: "The external profile allows anonymous calling, which is required as your provider will never authenticate with you to send you a call." Skimmed through the bridge book p78, Receiving calls, which seems to confirm external profile does not require authentication. Ok, fine. Then this: "In order to secure your FreeSWITCH it is wise to link your outbound profile to a dialplan context other than 'default', which in the default configuration is the where authenticated users are placed." Seems this advice mixes a default situation (default dialplan being sensitive) with conditional advice (your outbound profile which would be external in a default config). I would change outbound to external in order to lessen confusion: "In order to secure your FreeSWITCH it is wise to link your exgternal profile to a dialplan context other than 'default', which in the default configuration is where authenticated users are placed." ... although what is probably really meant is something like: "As mentioned, the profile used for outbound registrations allows anonymous, unauthenticated calling. By default, this profile is the external profile. In order to secure your FreeSWITCH, don't link this profile to a dialplan that allows dialing paid numbers or dialing users (who may be bothered/harrassed) without any further checking. Summary: in a default configuration: don't link your external profile to a 'default' dialplan." ... which is a mouthful. 2. Given the above, if I want to have external users in their own network behind NAT register to me, it would be best if I define an additional profile that does require SIP authentication, right? I can then use the external profile to register with SIP trunks etc. Thanks! -- Regards, jb

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twin events MESSAGE (mod_gsmopen, sms_incoming)
by royj 06 Jul '12

06 Jul '12

Hi all I want to use the mod_event_socket inbound mode to handle incoming SMS, there's a chatplan (http://pastebin.freeswitch.com/19448) and with this simple script (http://pastebin.freeswitch.com/19451) for every SMS I see two identical events (Event-Sequence with the same numbers as the other corresponding Headers do not differ from each other), is this normal? -- Regards, royj

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how to prevent the changes of nonce ttl sip auth ?
by Jack Nikolas 06 Jul '12

06 Jul '12

HI, i find out every time we want to register on sipclient the sip_nonce_auth get the value,and every time we unregister the same sipclient the value of sip_nonce_auth was changed ! but is there any solutions to prevent of changing the value of sip_nonce_auth for same sipclient that registered and unregistered? because suppose we have 25 sip client that is registered on our server,now i want to check the sip client that is registered be the same that is unregistered so if the value of sip_nonce_auth for the sipcliet be the equal with the value of the same sipclient that is unregister ,the problem will solve, in the internal.xml we have <param name="nonce-ttl" value="60"/> it means every 60sec the value of sip_nonce_auth for each sipclient was chnaged , and if the value of nonce-ttl is less than the value of parametere "register every ... " softphone, so the nonce-ttl was changed again. but i don't want change to check every sip client that is registered is the same that is unregistered i want when the status of sipclient was changed to 'unregistered' form 'registered' the value of 'sip_nonce_auth' be the same to find out the sip client that is registered was the same that is unregistered ! is there any solutions? ( i know we can check using fs_cli with sofia status profile internal reg and sofia_count_reg ..) but i want to check with sip header ... for example when sipclient is registered 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A42%3A44 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A12%3A44%20GMT Event-Date-Timestamp: 1341551564411676 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username: jack sip_auth_realm: 192.168.10.89 sip_auth_nonce: 37cb2234-c729-11e1-a170-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user: jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 when the same sip client is unregistered: 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A49%3A56 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A19%3A56%20GMT Event-Date-Timestamp: 1341551996651662 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username:jack sip_auth_realm: 192.168.10.89 sip_auth_nonce:397d1b86-c72a-11e1-a171-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user:jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 as you see the sip_auth_method is the same when user is registered AND unregistered ... but how can i findout that ? is there any method to change the 'sip_auth_method ' to another value when sipclient status was changed to 'unregistered'? thanks in advanced

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Transfer using lua no bleg present
by KPS Maillinglist 06 Jul '12

06 Jul '12

Hi All, I have a issue where when i transfer a call using lua. It tells me that "[WARNING] mod_dptools.c:976 No B-leg present." Below you will find some background info: @@@@@@@@@@ Log: http://pastebin.freeswitch.org/19458 @@@@@@@@@@ lua file relevant potions ########## dx_script.lua ###################### session:setAutoHangup(false) session:answer(); session:set_tts_parms("cepstral", "amy"); -- session:execute("", "") digits = session:read(4, 11, "tone_stream://%(10000,0,350,440)", 5000, "#") write_FS_LOG("Got dtmf: ", digits) --freeswitch.consoleLog("info", "Got dtmf: ".. digits .."\n"); --session:execute("read", "11 11 'tone_stream://%(10000,0,350,440)' digits 5000 #") session:execute("export","orig_transfer_digits=" .. digits) session:setVariable("orig_transfer_digits", digits) session:setVariable("orig_user_context",usr_context) session:execute("export","orig_user_context=" .. usr_context) atoit = string.sub(dst_number, 4) session:execute("transfer", "is_transfer-" .. atoit .. " XML FEATURES") write_FS_LOG("Script End Time: ", os.date("%Y-%m-%d %X")) ############### is_transfer_script.lua ###################### if string.find(dst_number, "-") ~= nil then dst_num_context_str = string.split(dst_number,"-") for dncs_index,dncs_str in pairs(dst_num_context_str) do -- do some stuff write_FS_LOG("DCNS Index", string.format("dcns_ind%s", dncs_index)) write_FS_LOG("DCNS VAR", string.format("%s", dncs_str)) session:setVariable("dcns_ind" .. dncs_index, dncs_str) orig_ucon = dncs_str end end -- orig_ucon = session:getVariable("dcns_ind2") -- orig_ucon = dcns_ind2 write_FS_LOG("orig_ucon", orig_ucon) if orig_ucon == nil then write_FS_ERRLOG("Missing data", "Missing user originating context") write_FS_LOG("Script End Time", os.date("%Y-%m-%d %X")) session:hangup() else write_FS_LOG("Recieved Context - Continuing...", orig_ucon) end if session:getVariable("orig_transfer_digits") == nil then write_FS_ERRLOG("Missing data", "session:getVariable(orig_transfer_digits)") write_FS_LOG("Script End Time", os.date("%Y-%m-%d %X")) session:hangup() else write_FS_LOG("Recieved digits - Continuing...", session:getVariable("orig_transfer_digits")) end -- session:execute("info","") -- session:execute("info","") -- orig_user_context = session:getVariable("orig_user_context") orig_transfer_digits = session:getVariable("orig_transfer_digits") write_FS_LOG("Script: ", "regsrv is_transfer_script.lua") write_FS_LOG("Script Start Time: ", os.date("%Y-%m-%d %X")) session:setAutoHangup(false) session:answer(); session:set_tts_parms("cepstral", "amy"); -- session:execute("", "") session:execute("transfer", "-bleg " .. orig_transfer_digits .. " XML " .. orig_ucon) write_FS_LOG("Script End Time: ", os.date("%Y-%m-%d %X")) @@@@@@@@@@ dialplan<document type="freeswitch/xml"> <section name="dialplan" description="FreeSWITCH Dialplan"> <context name="FEATURES"> <extension name="DX"> <condition field="destination_number" expression="^DX-.*"> <action application="lua" data="/fs/scripts/users/dx_script.lua"/> </condition> </extension> <extension name="is_transfer"> <condition field="destination_number" expression="^is_transfer-.*"> <action application="lua" data="/fs/scripts/users/is_transfer_script.lua"/> <anti-action application="eval" data="cancel transfer"/> </condition> </extension> </context> </section> </document>

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Problem with Polycom behind firewall with freeswitch
by Akhil Chandra 06 Jul '12

06 Jul '12

Hi all, I come from a asterisk background and having trouble keeping the firewall hole opened up for Polycom registering to freeswitch server. I am running freeswitch ( with bluebox UI) and trying to register a Polycom behind a router. I have exhausted all the options in sip_profiles directory for the interface I am using. The issue I am seeing is that the freeswitch sends the OPTIONS message to the internal IP ie ( 192.168.X.X) rather than the public IP, behind which the Polycom phones are. These phones register fine with asterisk in the stated scenario. I have tried changing these options in the xml file but still the OPTIONS message ( as seen in tcpdump) is sent to 192.168.X.X <param name="nat-options-ping" value="true"/> <param name="all-reg-options-ping" value="true"/> <param name="apply-nat-acl" value="rfc1918.auto"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> Any help appreciated !! .. Thanks in advance. Regards, Akhil

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Re: [Freeswitch-users] Crash on incoming E1 call
by Peter Olsson 06 Jul '12

06 Jul '12

Please report to Jira, with all needed information http://wiki.freeswitch.org/wiki/Reporting_Bugs /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Neo Cheema Skickat: den 5 juli 2012 07:53 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Crash on incoming E1 call Facing exactly the same issue. But in my case, freeswitch crashes at startup itself while configuring spandsp. My stack is Feetdm E1 with libpri. On Thu, Jul 5, 2012 at 11:00 AM, Anton Kvashenkin <anton.jugatsu(a)gmail.com> wrote: > Post this output to backtracker. > > 2012/7/5 Robin Gilks <robin.gilks(a)taitradio.com> >> >> Current running from commit 04bd463d12bf4c0116bec3cee537749b2040ed40 >> >> This occurs on all incoming calls from a Digium Wildcard TE210P with >> a zaptel E1 interface with libpri. >> >> I've tried updating to dahdi-linux-complete-2.6.1+2.6.1.tar.gz but >> failed miserably - couldn't even get the E1 link itself to come up :( >> >> I've captured a backtrace of the event on the zaptel drivers, >> hopefully useful to someone. >> >> gdb /usr/local/freeswitch_TR/bin/freeswitch -x ${HOME}/gdbcommands >> >> >> $ cat gdbcommands >> handle SIGPIPE nostop noprint >> handle SIG33 nostop noprint >> set logging on >> set pagination off >> set breakpoint pending on >> run >> thread apply all bt full >> set logging off >> >> Attached is the output of the backtrace. >> >> >> >> -- >> Robin Gilks >> >> -- >> >> ------------------------------ >> This email, including any attachments, is only for the intended recipient. >> It is subject to copyright, is confidential and may be the subject of >> legal or other privilege, none of which is waived or lost by reason >> of this transmission. >> If you are not an intended recipient, you may not use, disseminate, >> distribute or reproduce such email, any attachments, or any part thereof. >> If you have received a message in error, please notify the sender >> immediately and erase all copies of the message and any attachments. >> Unfortunately, we cannot warrant that the email has not been altered >> or corrupted during transmission nor can we guarantee that any email >> or any attachments are free from computer viruses or other conditions >> which may damage or interfere with recipient data, hardware or >> software. The recipient relies upon its own procedures and assumes >> all risk of use and of opening any attachments. >> ------------------------------ >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> 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-us >> ers >> http://www.freeswitch.org >> > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > 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-use > rs > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 !DSPAM:4ff5389332761306118382!

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