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ZRTP and Video in proxy-mode
by Yaroslav Krivopishin 16 Mar '17

16 Mar '17

Good afternoon. Have anyone else drop issue after trying to activate video during call with zrtp encryption? Every time I have INCOMPATIBLE_DESTINATION and hangup after that. In siptrace I found that after receiving sip/sdp invite, Freeswitch send SIP 100 trying and right after it send SIP 488 Not Acceptable Here. Than it follows by bye message to both side. In sofia debug all 9 I didn't realize issue too, but here it is: tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7f9850004440): events IN tport.c:2864 tport_recv_event() tport_recv_event(0x7f9850004440) tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f9850004440) msg 0x7f9850519470 from (udp/194.149.101.23:5060) has 2160 bytes, veclen = 1 tport.c:3023 tport_deliver() tport_deliver(0x7f9850004440): msg 0x7f9850519470 (2160 bytes) from udp/88.208.76.34:5060/sip next=(nil) nta.c:2880 agent_recv_request() nta: received INVITE sip:9468349698@194.149.101.23:5060;transport=udp SIP/2.0 (CSeq 22) nta.c:3174 agent_check_request_via() nta: Via check: received=88.208.76.34 nta.c:3248 agent_aliases() nta: canonizing sip:9468349698@194.149.101.23:5060 with contact nta.c:3060 agent_recv_request() nta: INVITE (22) going to existing leg nta.c:1348 set_timeout() nta: timer shortened to 200 ms nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f98503181f0) called soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7f98503181f0, (nil), 0x7f985054899a, 1206) called tport.c:3257 tport_tsend() tport_tsend(0x7f9850004440) tpn = UDP/ 88.208.76.34:52522 tport.c:4046 tport_resolve() tport_resolve addrinfo = 88.208.76.34:52522 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7f9850004440): not found by name UDP/88.208.76.34:52522 tport.c:3594 tport_vsend() tport_vsend(0x7f9850004440): 318 bytes of 318 to udp/88.208.76.34:52522 tport.c:3492 tport_send_msg() tport_vsend returned 318 nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (22) nua_stack.c:271 nua_stack_event() nua(0x7f985003baa0): event i_invite 100 Trying nua_session.c:4145 signal_call_state_change() nua(0x7f985003baa0): ready call updated: received received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7f98503181f0, [0x7f986753f590], [0x7f986753f598], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0x7f985003baa0): event i_state 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:529 nua_signal() nua(0x7f985003baa0): sent signal r_respond 2017-03-15 11:07:32.110686 [NOTICE] sofia.c:9717 Hangup sofia/default/ 9462793697(a)phone.pricall.eu [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] nua_stack.c:573 nua_stack_signal() nua(0x7f985003baa0): recv signal r_respond 488 Not Acceptable Here nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7f98503181f0, ...) called nua_session.c:2320 nua_invite_server_respond() nua: nua_invite_server_respond: entering soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0x7f98503181f0) called tport.c:3257 tport_tsend() tport_tsend(0x7f9850004440) tpn = UDP/ 88.208.76.34:52522 tport.c:4046 tport_resolve() tport_resolve addrinfo = 88.208.76.34:52522 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7f9850004440): not found by name UDP/88.208.76.34:52522 tport.c:3594 tport_vsend() tport_vsend(0x7f9850004440): 493 bytes of 493 to udp/88.208.76.34:52522 tport.c:3492 tport_send_msg() tport_vsend returned 493 nta.c:6791 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE (22) nua_session.c:4145 signal_call_state_change() nua(0x7f985003baa0): ready call updated: init nua_stack.c:271 nua_stack_event() nua(0x7f985003baa0): event i_state 488 Not Acceptable Here nua_stack.c:271 nua_stack_event() nua(0x7f985003baa0): event i_active 488 Call active soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f98503181f0) called nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering Without ZRTP encryption it works correctly. -- Best regards / S pozdravem / С наилучшими пожеланиями, Krivopishin Yaroslav.

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Re: [Freeswitch-users] [SOLVED] SIP profile set dialplan xml curl gateway-url?
by Richard Chan 15 Mar '17

15 Mar '17

Thanks for the confirmation — indeed, it didn't seem possible with a glance throught the xml curl source. I found Caller-Context in the POST parameters and that should suffice for the web app that provides the dialplan. - Richard

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FreeSWITCH blog RSS feed problems
by Daniel Pocock 15 Mar '17

15 Mar '17

Hi, I had been following the FreeSWITCH blog using the RSS feed https://freeswitch.org/feed/ but my feed reader is complaining with error 404 now when accessing that URL. It looks like the web site has been re-arranged and the blog page doesn't have the RSS or Atom icons nor does it have the meta-data for them in the page. What is the new URL for RSS? Maybe it would be a good idea to put a redirect in the web server so you don't lose followers? Regards, Daniel

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Updates are not passed upstream
by Jose Serrano 15 Mar '17

15 Mar '17

Hello, I am using freeswitch as class IV switch. I have observed that updates from SIP servers are nos passed upstream in an scenario as the following IVR------FS-------SBC----------GW1 | _____GW"

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Busy tone if busy
by Alexander Haugg 15 Mar '17

15 Mar '17

Hi together, I try to play a busy tone on leg A if the leg B is busy. For this I'm using the inline dialplan after the originate command: originate sofia/gateway/GW_H3KSip/myMobileNumber 'set:hangup_after_bridge=false,bridge:sofia/gateway/GW_H3KSip/2640,managed:C4B_AfterBridge' inline The managed dialplan app "C4B_AfterBridge" includes only the check, if the cause is busy and the playback command for the tone stream. In the first line of this app is a logging line for which reason I know, the app will not called. WHY? Thanks for your help!

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pickup vs intercept
by Luke Milbourne 14 Mar '17

14 Mar '17

Hi Everyone, Using pickup endpoints, if the UA picking up a call is utilising a different codec (G722) FreeSWITCH raises a codec error. However the same test using the intercept function instead of a pickup endpoint doesn't and the call works fine with 2 way audio etc. Anyone else experienced/aware of this? Thanks, Luke

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Re: [Freeswitch-users] Cepstral will provide an Allison-8kHz for Linux license to anyone donating $100 or more.
by agoulis＠opensips.org 14 Mar '17

14 Mar '17

I thought it was a license per 100? Thanks, Alex GoulisOpenSIPS Solutions4016411072 ------ Original message------From: Brian WestDate: Mon, Mar 13, 2017 2:22 PMTo: Luis Daniel Lucio Quiroz;Cc: FreeSWITCH Users Help;Subject:Re: [Freeswitch-users] Cepstral will provide an Allison-8kHz for Linux license to anyone donating $100 or more. Its what I've used in the past, seems to have worked well. On Mon, Mar 13, 2017 at 2:06 PM, Luis Daniel Lucio Quiroz <luis.daniel.lucio(a)gmail.com> wrote: Why did you select gofoundme for this crowdfunding? --Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH On Mon, Feb 13, 2017 at 5:31 PM, Brian West <brian(a)freeswitch.org> wrote: See that? Cepstral will provide an Allison-8kHz for Linux license to anyone donating $100 or more. You get an Allison 8kHz license for Linux if you donate more than 100 dollars. https://www.gofundme.com/allison-prompts-for-freeswitch Thanks, -- Brian West brian(a)freeswitch.orgTwitter: @FreeSWITCH , @briankwesthttp://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH:https://www.gofundme.com/allison-prompts-for-freeswitch Wish to schedule a meeting?http://app.timebridge.com/#/meet/freeswitchGot Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com 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 -- Brian West brian(a)freeswitch.orgTwitter: @FreeSWITCH , @briankwesthttp://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH:https://www.gofundme.com/allison-prompts-for-freeswitchGot Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest

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SIP profile set dialplan xml curl gateway-url?
by Richard Chan 14 Mar '17

14 Mar '17

Is there a way for a SIP profile to change the XML curl dialplan gateway-url just for that profile? It seems that the gateway-url for dialplan binding is set at start up. Thanks. -- Richard Chan

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Getting call audio stream from a java application
by Thomas, Tom 14 Mar '17

14 Mar '17

Hello Guys, I know your time is valuable, but I am using this mailing list as my last resort. I have tried to look for answers everywhere else, including the Confluence, wiki and google, and previous posts on the mailing list. So here's my dilemma. I have a custom speech recognition engine which I want to use with freeswitch. I am not looking to use the built-in pocketsphinx module or using an mrcp server. What I am doing right now is, when a call is answered, I am recording it to a file using record_session. I have a java application that reads from this file and sends data to my speech engine. This would have been sufficient, if not for the fact that freeswitch writes 4 second chunks of audio at a time. I verified this by writing a small test. The recorded file is updated every 4 seconds with the previous 4 seconds of audio. This is not an acceptable delay for my particular application. So I tried looking into other options where I can directly access the call audio. I know mod_shout can be used to setup an icecast stream, but that's not really what I would like to do. Same for the telecast feature, as I would like to use the raw audio stream instead of transcoded audio. I read in one post on stackoverflow that it is possible to use eavesdropping to listen to the audio stream. But it didn't go into any further details on how this can be achieved through an external application, like Java. Looking up examples for eavesdropping on freeswitch didn't help either. So I'd like to know, if it's possible to directly access the audio stream from, say a Java program using the freeswitch library? If not, how does pocketsphinx and other built in speech engines really read in the audio? Also, if direct access is not possible, is there any way to change the recording write speed? Thanks.

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Perl ESL Windows
by Alexey Gorbachev 14 Mar '17

14 Mar '17

Hi, everybody! I've installed FreeSwitch for Windows from binaries. How is it possible to enable ESL support for Perl? I'm getting an error "can't locate ELS.pm" on string "require ESL;". Am I able to find precompiled binaries for this library? Best regards, Alexey Gorbachev.

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