July 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

privacy and anonymous@anonymous.invalid
by Panagiotis Skoulikaritis 19 Jul '16

19 Jul '16

Dear all I have some issues with a privacy headers, in my understanding is that it has to do with trusted/untrusted networks FreeSWITCH receives: From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as19e43f0e P-Asserted-Identity: "Anonymous" <sip:anonymous@anonymous.invalid> and sends out: From: "Anonymous" <sip:anonymous@10.10.10.10>;tag=yp9N2QvepFc9K P-Asserted-Identity: "Anonymous" <sip:anonymous@10.10.10.10> The Terminating carrier doesn't want to receive the IP address but rather the anonymous.invalid I searched through the mail list but I didn't find anything that could help me fix it. Bellow is the dialplan I currently use for the carrier: <extension name="Test"> <condition field="network_addr" expression="^(^10\.10\.10\.1|^10\.10\.11\.1)$"/> <condition field="destination_number" expression="^1234#(\d+)$" require-nested="false"> <condition field="caller_id_number" expression="^(00)(\d+)$" break="never"> <action application="set" data="effective_caller_id_number=+$2" inline="true"/> <action application="set" data="effective_caller_id_name=+$2" inline="true"/> </condition> <condition field="caller_id_number" expression="^([1-9]\d+)$" break="never"> <action application="set" data="effective_caller_id_number=+$1" inline="true"/> <action application="set" data="effective_caller_id_name=+$1" inline="true"/> </condition> <action application="set" data="call_direction=outgoing"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="sip_cid_in_1xx=false"/> <action application="set" data="inherit_codec=true"/> <action application="set" data="accountcode=${sip_h_X-ACCOUNTCODE}" inline="true"/> <action application="export" data="nolocal:accountcode=${sip_h_X-ACCOUNTCODE}" inline="true"/> <action application="unset" data="sip_h_X-ACCOUNTCODE" inline="true"/> <action application="set" data="execute_on_answer=sched_hangup +21600 alloted_timeout"/> <action application="bridge" data="sofia/external/+$1@192.168.157.10:5060"/> </condition> </extension> any help will be greatly appreciated. Best Regards Panagiotis

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how to get my vanilla config for new module generated during make install into conf/autoload_configs
by Dave Horton 18 Jul '16

18 Jul '16

I am building a new asr_tts module and I am not clear on how to get the associated *.conf.xml file generated into the ${install_dir}/conf/autoload_configs directory. I tried added it to {srcdir}/conf/vanilla/autoload_configs and a reference to it in ${srcdir}/freeswitch.spec but that didn’t seem to do it.

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Registration to FreeSWITCH through multiple proxy problem
by Dmitry Saratsky 18 Jul '16

18 Jul '16

Hi, I have a scenario, where I need to register to freeswitch through 2 or more stateless forwarding kamailio proxies. <User> => <Proxy1> => <Proxy2> => <FreeSWITCH> Registration packet which comes to freeswitch have 3 VIA Headers (please see below): REGISTER sip:146.148.54.153 SIP/2.0 Via: SIP/2.0/TCP 159.203.230.115:5060;branch=z9hG4bKea26.3936c0f1c658d2da02a7c46a86186077.0;i=1;rport Via: SIP/2.0/TCP 188.166.22.58:5060;received=188.166.22.58;branch=z9hG4bKea26.98aaf5f1397dccdf6f103ae4150d3ce0.0;i=14;rport=36525 Via: SIP/2.0/TCP 216.72.41.53:63675;branch=z9hG4bK-524287-1---e991c17828928028 Max-Forwards: 68 Contact: <sip:219@216.72.41.53:63675;rinstance=e6186ad594e06aa9;transport=tcp> To: "219"<sip:219@146.148.54.153> From: "219"<sip:219@146.148.54.153>;tag=8c8fc855 Call-ID: NTE1OWIwNjYyODg0YTNiNDVkZDZkNjQxMjlhM2IxYzE CSeq: 1 REGISTER Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, MESSAGE, SUBSCRIBE User-Agent: Bria iOS release 3.6.5 stamp 34728 Content-Length: 0 Expires: 120 P-hint: outbound P-hint: outbound User register to freeswitch successfully, however back calling to end-user is impossible bcz it is not going to reach end-phone using VIA haeder from REGISTER. If I use aggressive-nat: freeswitch send invite calls to end-user to proxy2, which have then no idea where where to send the call (no route haeder in invite from freeswitch) If I do not use aggressive-nat - freeswitch sends it's invite directly to end-user phone((( In this example I'm using SIP over TCP, however the same is happened over UDP... So, the question is how to register correctly to FS and preserve all VIA headers with fs_nat in contact db or some other way, so I could reach the end-user phone in same patch it is registering? Thanks a lot, Dmitry.

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Fax detection for outbound calls
by Gonzalo Gasca Meza 17 Jul '16

17 Jul '16

Hi all, I'm configuring FS 1.7 to perform a dialout. If a fax is detected, I would like transfer call to an specific extension/application. I tried example in FreeSwitch cookbook, but doesn't seem to detect the FAX. http://pastebin.com/0F6M1DSe Topology: Freeswitch --> SIP Provider -- PSTN ---> Fax machine Only when I try from CLI invoking spandsp_start_fax_detect it works: originate {ignore_early_media='true',fax_enable_t38=true,fax_verbose=true,fax_use_ecm=false,fax_enable_t38_request=false}sofia/gateway/twilio/+18884732963 'spandsp_start_fax_detect:txfax /tmp/txfax-sample.tiff 10 ced,sleep:10000' inline These are the logs using Freeswitch Cookbook sample: http://pastebin.com/0F6M1DSe I have tried the following combinations without success: <action application="export" data="execute_on_fax_detect='execute_extension fax'"/> <action application="spandsp_start_fax_detect" data="txfax '/tmp/txfax-sample.tiff 10 ced'"/> <action application="spandsp_start_fax_detect" data="transfer 'fax XML default' 2"/> <action application="spandsp_start_fax_detect" data="transfer '9179 XML default' 2"/> This is my config http://pastebin.com/6Awmxk14 <extension name="telephonist"> <condition field="destination_number" expression="^handle_calls$"> <action application="answer"/> <action application="log" data="INFO handle_calls application match"/> <action application="set" data="fax_verbose=true"/> <action application="set" data="transfer_ringback=${us-ring}"/> <action application="export" data="execute_on_fax_detect='execute_extension fax'"/> <action application="playback" data="silence_stream://20000"/> <action application="hangup"/> </condition> </extension> <extension name="fax_handler"> <condition field="destination_number" expression="^fax$"> <action application="log" data="INFO FAX detected"/> <action application="txfax" data="/tmp/txfax-sample.tiff 10 ced"/> <action application="log" data="INFO FAX completed"/> <action application="hangup"/> </condition> </extension> <extension name="fax_receive"> <condition field="destination_number" expression="^9178$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="$${temp_dir}/rxfax.tif"/> <action application="hangup"/> </condition> </extension> <extension name="fax_transmit"> <condition field="destination_number" expression="^9179$"> <action application="txfax" data="$${temp_dir}/txfax-sample.tiff"/> <action application="hangup"/> </condition> </extension> Thanks

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Questions about using httapi playback
by Lin, Xiaoping 17 Jul '16

17 Jul '16

In the FreeSwitch 1.2 book ( chapter 11), examples shows digit-timeout and input-timeout are in seconds, but in Wiki page <document type="xml/freeswitch-httapi"> <params></params> <work> <playback name="exten" file="http://my.com/sounds/exten.wav"<http://my.com/sounds/exten.wav> error-file="http://my.com/sounds/invalid.wav"<http://my.com/sounds/invalid.wav> input-timeout="5000"> <bind strip="#">~\d+#</bind> </playback> </work> </document> They must be ms? Thanks

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Enable voice in DONGLE - GSMOpen
by Luiz Fernando Softov 17 Jul '16

17 Jul '16

Hi, I don't know if this is the correct place, but since this is related with GSMOpen. I already google it, and can't found anything. I know with DCUnlocker I can enable voice, and sometimes with AT^CVOICE=0 i can make voice work. But, there are moments when I use CVOICE, and can't enable voice support I send > AT^CVOICE=0 REPLY < OK Then I send > AT^CVOICE=? REPLY < CVOICE: (1) So, using DCIUnlocker, in the same device, the voice was successful enabled. There is another way to unlock voice without DCUnlocker? Maybe, another AT command has to be make too, or something else?

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INCOMPATIBLE_DESTINATION - GSM (L16) to User
by Luiz Fernando Softov 15 Jul '16

15 Jul '16

Hi, I'm trying to make a call between GSMOpen (GSM) and a user (or a group, it's the same). But I receive a 488 INCOMPATIBLE_DESTINATION I'm thinking this is because the codec negotiation. My profile: https://pastebin.freeswitch.org/view/2f8a0f95 My FS LOG https://pastebin.freeswitch.org/view/f6fc8bac Please, someone know how to solve this?

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T38<>T30 transcoding
by Mirko Brankovic 15 Jul '16

15 Jul '16

Hi, Seems to me that I'm not configuring something correctly or hitting some bug. Reading https://wiki.freeswitch.org/wiki/Mod_spandsp#t38_gateway I have a T.38 incapable provider that will respond with SIP 488 back. If I leave T.38-passtrough and get 488 back, RTP stops and call is dropped. So I was trying to set up correct transcoding. This is the scenario, with 3 identical FS: 1 - T.38 incapable server - PROVIDER 2 - T.38 capable - PROXY 3 - T.38 capable - PBX PBX sends txfax (file.tiff) to PROXY which forwards call to PROVIDER (g.711). PBX sends T.38 re-Invite, PROXY responds with 200 OK but then I have 2 scenarios: either passtrough T.38 which will invoke 488 and dropped call or I tried sip_execute_on_image=t38_gateway peer nocng" but this doesn't send anything to PROVIDER but also call is dropped after BYE and 486 message. Case that I'm hitting is similar to this: "If you want to transcode t.38 on leg a to audio on leg B when leg a invites t.38, but refuse all t.38 offers from leg b " but to refuse t.38 invites from lag a. The other way around, call from PROVIDER to PBX over PROXY is easely transcoded with sip_execute_on_image=t38_gateway self nocng" on the PROXY side. Is it normal behaviour that After failed t.38 negotiation or 488 message, fax call doesn't fall back to previous codec or t.30? -- Regards, Mirko

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assisted transfer on Verto
by Azlain Saavedra 15 Jul '16

15 Jul '16

Verto HTML5/Javascript version 1.1 I would like to do an assisted transfer call using verto. I am already doing audio calls, video calls, blind calls and holds and resume. Is there any example on how to do a assisted transfer? -- T.S.U Azlain Saavedra Movil: +58 04142528099 Skype: azlain.saavedra

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Eavesdrop on only one side of a call
by Tom Fielding 15 Jul '16

15 Jul '16

Hi all, Is there only a way to only eavesdrop on one side of a call? Currently, it seems like eavesdrop listens to both the caller and callee. Thanks, -Tom

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