October 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Upcoming maintenance!
by Brian West 25 Oct '18

25 Oct '18

FreeSWITCHers, Over the weekend you'll see various public facing services for the FreeSWITCH go down for a while. Please be patient while we make changes. Thanks, -- Brian West | Co-founder and Developer Need Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> Email: brian(a)freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>

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Sometimes "hanging" channels
by Markus Bönke 25 Oct '18

25 Oct '18

Hello All, we are running freeswitch 1.6.20, calls are controlled via ESL, CDRs are written with mod_xml_cdrl. Sometimes we see „hanging“ channels. In such a case the CDR via mod_xml_cdr is written and the last log entry for such a call is "Locked, Waiting on external entities“. ... freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263 2018-10-09 10:18:15.197345 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/anonymous(a)anonymous.invalid) State Change CS_REPORTING -> CS_DESTROY freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263 2018-10-09 10:18:15.197345 [DEBUG] switch_core_session.c:1665 Session 166111 (sofia/internal/anonymous(a)anonymous.invalid) Locked, Waiting on external entities ... How can I proceed to further analyze the problem ? In the last log line I also see the session number (Session 166111) - is there a way to find out on which external entity it is waiting? Thanks and regards Markus

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CentOS 7, FreeSWITCH 1.8.2, mod_flite error
by Social Boh 24 Oct '18

24 Oct '18

Hello list, I compiled FreeSWITCH 1.8.2+git~20180926T175525Z~a98a958ac3~64bit en CentOS 7.5.1804 without problem with flite-devel version 2.0.0-1.el7. Now, when I try to load the module: /[CRIT] switch_loadable_module.c:1522 Error Loading module /usr/lib/freeswitch/mod/mod_flite.so// //**/lib64/libflite.so.1: undefined symbol: snd_pcm_hw_params_any**/ Any Hint? Thank you Regards -- --- I'm SoCIaL, MayBe

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SpanDSP V34 Support
by Douglas Davenport 24 Oct '18

24 Oct '18

There appears to be a build option to add V34 faxing support to freeswitch SpanDSP. Does anyone know if this code is functional and how to enable it? Thanks

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[username password with @][http cache]
by Bilal Abbasi 24 Oct '18

24 Oct '18

Hey users, I have to download a sound file from FTP server having password with @ in it. The mod_http_cache has the way username:password@url, i was looking to escape the @ from it. This might be very basic question to ask, may be some one has done that already. Regards Abbasi

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Re: [Freeswitch-users] h264 video call between endpoints with one end at bad resolution of 352x288
by Michael Jerris 23 Oct '18

23 Oct '18

sounds like you might be doing the option to mirror resolution > On Jun 28, 2018, at 9:07 AM, antonio via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > From: antonio <asilva(a)wirelessmundi.com> > Subject: h264 video call between endpoints with one end at bad resolution of 352x288 > Date: June 26, 2018 at 6:05:20 AM EDT > To: freeswitch-users(a)lists.freeswitch.org > > > Hi, > > Calling between two Grandstream GXV3275 i expected to have a video call at 1280x720, but the video that is present at destination as the resolution of 352x288 and the image as poor quality, originator is sending video at 1280×720. If i swap the devices the result is the same, the destination always get a bad resolution of 352x288 but the originator receives the video at expected resolution of 1280x720. > > If i set rtp_direct between the two endpoints it work as expected. > > I also tested a conference room and force the it resolution to 1280*x720 it work well, if i don't force the canvas-size i see: > > 2018-06-26 13:55:02.077824 [WARNING] mod_conference.c:3157 Unspecified video-canvas-size, falling back to 1280x720 > > And the conference is done at 1280x720. > > > I've open a jira where you can find a full log with this behavior: > > https://freeswitch.org/jira/browse/FS-11140 > > > I don't understand why there is transcoding and mod_av is used, the codec negotiated is the same.. i understand that the resolution is at 352x288 for the first frames sent by the phone but then it send them at 1280x720, mod_av should detected and adjust the new resolution, no? > > > Anyone experience this behavior? >

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How to play some file into parked call via inbound ESL connection?
by Eugene Prokopiev 22 Oct '18

22 Oct '18

Hi, Is it possible to play some file into parked call via inbound ESL connection? I have dialplan like this: <extension name="in"> <condition> <action application="park"/> </condition> </extension> I tried: $ telnet localhost 8021 auth ClueCon Content-Type: command/reply Reply-Text: +OK accepted event plain CHANNEL_PARK Content-Type: command/reply Reply-Text: +OK event listener enabled plain Content-Length: 5397 Content-Type: text/event-plain Event-Name: CHANNEL_PARK ... Unique-ID: 783e9456-d1f2-11e8-a77f-95321f80797f ... api uuid_answer 783e9456-d1f2-11e8-a77f-95321f80797f Content-Type: api/response Content-Length: 4 +OK api uuid_displace 783e9456-d1f2-11e8-a77f-95321f80797f start /usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav Content-Type: api/response Content-Length: 12 +OK Success api uuid_broadcast 783e9456-d1f2-11e8-a77f-95321f80797f playback::/usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav Content-Type: api/response Content-Length: 17 +OK Message sent api uuid_kill 783e9456-d1f2-11e8-a77f-95321f80797f Content-Type: api/response Content-Length: 4 So, both api uuid_displace and api uuid_broadcast returns immediately with +OK. Is it possible to play and wait until the end of file? Or this is possible via sendmsg instead of api and suscribing on PLAYBACK_STOP event? -- WBR, Eugene Prokopiev

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Freeswitch failed to initiate outbound call using SIPs + SRTP (SRTP unprotect )
by Chhorm Chhatra 22 Oct '18

22 Oct '18

Hello, Currently, I faced a problem regarding SRTP outbound call to user (Leg B). The scenario is like this, - We set up our own root CA to an IP address (e.g 192.168.0.13) - We create a server certificate for freeswitch at 192.168.0.13 - Linphone is used as SIP client and is configured to trust our root CA by default. - Linphone A is configured to register to Freeswitch vis TLS + SRTP. (One leg call to server has both SIPs and SRTP – completely secure) - Linphone B is registered to Freeswitch via TLS + SRTP, and waiting for Linphone A to call to. (One leg call to server, e.g. 9196 (echo test), is completely secure with SRTP + SIPs) - Unfortunately, if A call to B, only A leg has SIPs + SRTP, but Leg B is not encrypted with SRTP and SIPs at all. This causes *SRTP unprotect failed with code 7 (auth check failed)**.* + Dialplan Configuration <action application="set" data="rtp_secure_media=true"/> <action application="export" data="rtp_secure_media=true"/> The dial-string is <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> + Directory Configruation: <param name="dial-string" value="{rtp_secure_media=${ regex(${sofia_contact(${dialed_user}@${dialed_domain}) }|transport=tls)},presence_id=${dialed_user}@${dialed_ domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> My question is that, is there any configuration left that I have to set up in order to let freeswitch initiate an outbound call to Leg B correctly with SRTP and SIPs (tls)? Any help would be really appreciated. Thank you so much. Best Regard,

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VideoCall hangup always after 32 seconds to answer
by Jose David Jurado Alonso 20 Oct '18

20 Oct '18

Hi, I'ḿ get the next dialplan configuration in order to set the call max limit timeout on 120 seconds after answer: <extension name="Local_Extension"> <condition field="destination_number" expression="^(.*)$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ignore_early_media=false"/> <action application="set" data="execute_on_answer=sched_hangup +120" /> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> </condition> </extension> But the call always disconnect after 32 seconds... (calling linphone to linphone android clients). I tried always using "call_timeout", "originate_timeout" but the result is the same. What I really want to do is: + Set timeout of 30 seconds to hangup if NO answer (nobody responds). + Set a maximum total time of 120 seconds since the call is answered. Can someone help me with this configuration? FS version: FreeSWITCH Version 1.9.0-654-ed4920e~64bit (-654-ed4920e 64bit) Thanks! José David Jurado Alonso.

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Accepting an "optional" SRTP offer (crypto in RTP/AVP) and establishing SRTP
by William Simon 16 Oct '18

16 Oct '18

We are working with an endpoint that offers optional SRTP in the non-RFC-compliant way of an a=crypto attribute within RTP/AVP. I have told FreeSWITCH to allow this with rtp_allow_crypto_in_avp=true at the right place in the dialplan. Immediately after that in the dialplan I have to reject SRTP by using rtp_secure_media=forbidden, otherwise the call setup still fails. By setting the value to forbidden, the call does proceed unencrypted. We want FreeSWITCH to proceed with media encryption. Setting rtp_secure_media to any other value results in FreeSWITCH rejecting the offer like this: v=0 o=FreeSWITCH 1538660754 1538660755 IN IP4 192.168.100.104 s=FreeSWITCH c=IN IP4 192.168.100.104 t=0 0 m=audio 0 RTP/AVP 19 m=video 0 RTP/AVP 19 Is there anything else I can do to force SRTP in the answer? “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”

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