September 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Detecting speech over ESL using mod_unimrcp.
by Anthony Minessale 21 Sep '18

21 Sep '18

Its async so send the call to do something like park so it doesn't end. On Thu, Sep 20, 2018 at 4:21 PM Ilya Spirin via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Ilya Spirin <splva(a)mail.ru> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Thu, 20 Sep 2018 13:12:49 -0700 (MST) > Subject: Detecting speech over ESL using mod_unimrcp. > Hello everyone! > > I'm trying to recognize speech with my application. FS built with > mod_unimrcp. Also I have installed UniMRCP Server with Google Speech > Recognition plugin. > > play_and_detect_speech application works fine. But I have to stop playback > manually that's why I'm sending some commands via ESL. > > Here is ESL dump: > https://pastebin.com/p1jUrhn6 <https://pastebin.com/p1jUrhn6> > After sending answer, playback and detect_speech commands I immediately got > "CHANNEL_EXECUTE_COMPLETE" event for detect_speech application. > > Please help. What I'm doing wrong? > Thank you. > > -- > Best regards > Ilya Spirin > > > > -- > Sent from: http://freeswitch-users.2379917.n2.nabble.com/ > > > > > ---------- Forwarded message ---------- > From: Ilya Spirin via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Thu, 20 Sep 2018 14:21:25 -0700 (PDT) > Subject: [Freeswitch-users] Detecting speech over ESL using mod_unimrcp. > _________________________________________________________________________ > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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video conference call, member audio only
by Bastien LE REST 21 Sep '18

21 Sep '18

Hello, I’m trying to make a conference call. (A) bridge a video call to (B) and (B) answer, I transfer to the conference just before bridging using: api_before_bridge=uuid_transfer ${uuid} -both 'conference:TEST’ inline. Everything works fine. (B) bridge another audio call to (C) and (C) answer, I transfer to the conference using the same command: api_before_bridge=uuid_transfer ${uuid} -both 'conference:TEST’ inline. Audio is fine, I can hear (A), (B) and (C). The video between (A) and (B) stop and I’m getting the following warning on FS console: [WARNING] switch_core_media.c:14282 sofia/external/+33969321515 has no video codec. Does it possible to have 2 video members and 1 audio only member in the same conference? Here full traces: https://pastebin.com/SExxV6EC Thank you,

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Detecting speech over ESL using mod_unimrcp.
by Ilya Spirin 20 Sep '18

20 Sep '18

Hello everyone! I'm trying to recognize speech with my application. FS built with mod_unimrcp. Also I have installed UniMRCP Server with Google Speech Recognition plugin. play_and_detect_speech application works fine. But I have to stop playback manually that's why I'm sending some commands via ESL. Here is ESL dump: https://pastebin.com/p1jUrhn6 <https://pastebin.com/p1jUrhn6> After sending answer, playback and detect_speech commands I immediately got "CHANNEL_EXECUTE_COMPLETE" event for detect_speech application. Please help. What I'm doing wrong? Thank you. -- Best regards Ilya Spirin -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Facing issue while using uuid_media_reneg on answered call
by Ravindrakumar Bhatt 20 Sep '18

20 Sep '18

Hello, I am using freeswitch 1.6.20. it is fresh setup with basic configuration. i want to renegotiate codec when call is answered and i am using uuid_media_reneg for same. SCENARIO: user a calls to user b and both user are using code G722. when i use uuid_meida_reneg to change codec to G729 for A leg it is working fine but when using same api with bleg uuid for B leg it is using G722 for renegotiate. Here are console log for same. https://pastebin.freeswitch.org/view/342e4eed Any suggestions for this or setting needed to be done in freeswitch. -- *Thanks and Regards,* *Ravindrakumar Bhatt* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+918460692402*

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Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI
by Ben Kaufman 20 Sep '18

20 Sep '18

Setting up Freeswitch to connect to Microsoft Teams' "Direct Routing" (https://docs.microsoft.com/en-us/microsoftteams/direct-routing-plan) General call setup-teardown is successful, but I'm encountering a problem with call transfers if the transfer target is another Microsoft Teams user. The most immediate obvious difference between the transfer to an external party and the transfer to another Teams user is in the Refer-To header of the Refer message on the transfer event. In the case of transfer to an external party, the Refer-To is (phone number redacted to 555-555-1000): > REFER-TO: <sip:+15555551000@sip.pstnhub.microsoft.com:5061;user=phone;transport=tls> In the case of transfer to another Teams user, the transfer Refer-To: header looks like this: > REFER-TO: <sip:sip.pstnhub.microsoft.com:5061;transport=tls;x-m=8:orgid:13bc82d9-003a-43b8-aad2-8f2f561a9b8d> When this occurs, the last line in the Freeswitch log for the call is: > [DEBUG] sofia.c:8544 Process REFER to [(null)(a)sip.pstnhub.microsoft.com] There is nothing else in the log until this times out, no indication it has entered any dialplan, etc. Even on timeout, it is just the teams client saying that the call has failed, and unless one end hangs up, or reconnects to the failed transfer, there is no action in freeswitch. I believe this to be a bug, but wanted to get an opinion before opening a formal bug report. Regards, Ben Kaufman AltiGen Communications, Inc. STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, or the person responsible for delivering the e-mail to the intended recipient, be advised you have received this message in error and that any use, dissemination, forwarding, printing, or copying is strictly prohibited. Please notify AltiGen Communications immediately at either (888)258-4436 or via email to administrator(a)altigen.com, and destroy all copies of this message and any attachments.

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http_cache use of local cache directory
by Francisco Caraballo 20 Sep '18

20 Sep '18

Hello. I'm trying to figure out how mod_htp_cache uses the local cache directory when uploading a file to S3. Speciffically I would like to know how long will the file be kept in the local file system cache (by default /usr/local/freeswitch/cache) in the case of an error while uploading to S3. Will it be deleted next time freeswitch starts? Will it stay there forever? The reason I'm asking is because we are trying to implement a fallback system to recover and upload recordings to S3 in case there was a problem when http_cache first attempted to upload the file (for example a 403 error, or a connectivity issue). Thanks a lot for your help. Francisco Caraballo

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Re: [Freeswitch-users] How mod_verto is invoked to handle incoming call?
by Michael Jerris 19 Sep '18

19 Sep '18

I’ve answered your question on this week’s community corner on cluecon weekly. > On Sep 14, 2018, at 11:07 AM, kaiduan xie via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > From: kaiduan xie <kaiduanx(a)yahoo.ca> > Subject: How mod_verto is invoked to handle incoming call? > Date: September 13, 2018 at 12:10:28 PM EDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Hi all, > > I am learning Freeswitch webrtc, and setup an environment with demo configuration as below. > > 1. User 1008 is logged in from browser with Verto. > > 2. User 1000 is logged in from X-Lite SIP phone. > > "sofia status profile internal reg" only shows 1000. > > 1000 makes a call to 1008, 1008 answers the call, and call is established between browser and X-Lite. > > The following is from freeswitch log, > 2662 7fbee0e2-b766-11e8-b3d6-f9bb4821b253 2018-09-13 15:05:49.296720 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] > > 2663 7fc69cd8-b766-11e8-b3f8-f9bb4821b253 2018-09-13 15:05:49.296720 [DEBUG] mod_rtc.c:392 () State Change CS_NEW -> CS_INIT > > I am wondering how mod_verto is invoked to handle the incoming call. Can someone explain please? > > Many thanks, > > /Kaiduan >

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Date Time parsing question
by Ryan Delgrosso 13 Sep '18

13 Sep '18

Hello all, Ive been puzzling over a date time parsing issue in FS and maybe someone can tell me what ive done wrong, since this log message makes little sense: XML DateTime Check: date time[2018-09-13 12:27:50] =~ 2018-09-13 13:00~2018-09-18 08:00 (PASS) I would expect that to evaluate to false/fail since the datetime is not between the specified parameters. FWIW I also tried specifying seconds in the argument with the same result. Any hints are much appreciated. -Ryan

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How mod_verto is invoked to handle incoming call?
by kaiduan xie 13 Sep '18

13 Sep '18

Hi all, I am learning Freeswitch webrtc, and setup an environment with demo configuration as below. 1. User 1008 is logged in from browser with Verto. 2. User 1000 is logged in from X-Lite SIP phone. "sofia status profile internal reg" only shows 1000. 1000 makes a call to 1008, 1008 answers the call, and call is established between browser and X-Lite. The following is from freeswitch log, 2662 7fbee0e2-b766-11e8-b3d6-f9bb4821b253 2018-09-13 15:05:49.296720 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2663 7fc69cd8-b766-11e8-b3f8-f9bb4821b253 2018-09-13 15:05:49.296720 [DEBUG] mod_rtc.c:392 () State Change CS_NEW -> CS_INIT I am wondering how mod_verto is invoked to handle the incoming call. Can someone explain please? Many thanks, /Kaiduan

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Gateway SIP OPTIONS ping Contact header
by Sebastian Zenzerović 13 Sep '18

13 Sep '18

Hello, is it possible to add (configuration option/variable?) Contact header to outgoing gateway ping messages sent by Freeswitch? Remote SIP peer is answering "403 forbidden" as there is no Contact header present in SIP OPTIONS. Not a big deal since FS sees the peer as available/UP anyway but could be a interoperability enhancement between FS and Microsoft Direct routing service (https://aka.ms/dr/) Here are the requirements: • Must: When placing calls to the Direct Routing interface, the 'CONTACT' header must have the SBC FQDN in the URI hostname • Syntax: Contact: <phone number>@<FQDN of the SBC>:<SBC Port>;<transport type> • If the parameter is not configured correctly, OPTIONS are rejected with a '403 Forbidden' message Freeswitch log: OPTIONS sip:sip-du-a-euno.pstnhub.microsoft.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS Max-Forwards: 70 From: <sip:sbc.xxx.xx>;tag=0aB5jm3SF07ve To: <sip:sbc.xxx.xx> Call-ID: 101b723c-3113-1237-6092-00155d000d46 CSeq: 128016309 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.8.1-2-4f54cff36a~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 SIP/2.0 403 Forbidden FROM: <sip:sbc.xxx.xx>;tag=0aB5jm3SF07ve TO: <sip:sbc.xxx.xx> CSEQ: 128016309 OPTIONS CALL-ID: 101b723c-3113-1237-6092-00155d000d46 VIA: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS REASON: Q.850;cause=21;text="4da4bb47-5ace-41b2-ba4b-6516fb4771c2;Record-Route and Contact headers are missing" CONTENT-LENGTH: 0 ALLOW: INVITE ALLOW: ACK ALLOW: OPTIONS ALLOW: CANCEL ALLOW: BYE ALLOW: NOTIFY SERVER: Microsoft.PSTNHub.SIPProxy v.2018.9.10.4 i.EUNO.4 Thanks. BR, Sebastian

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