September 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones Crashing w/ mod_av Enabled
by Luis Azedo 08 Sep '17

08 Sep '17

> > i noticed that freeswitch crashes with mod_av enabled in debian stretch > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 07 Sep 2017 17:00:55 +0000 > Subject: Re: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones > Crashing w/ mod_av Enabled > I haven't tested all those end points but for sure we have used video with > latest and beta bria and had it work. If clients are crashing when > receiving video in sdp you should consult with those vendors to find out > why. > > On Wed, Sep 6, 2017 at 3:31 AM roy j <royj(a)yandex.ru> wrote: > >> Hi. Just interested, you say they crash, what exactly is happening, they >> are rebooting? If so I would ask Cisco about it, it's so strange. >> >> 05.09.2017, 10:23, "Joseph Barrero" <joe(a)expert.net>: >> >> Hi, I’m running Freeswitch v1.6.19. >> >> When mod_av is enabled, inbound calls to Cisco SPA504g and SPA514g ip >> phones crash as they start to ring. The crash seems to occur immediately >> after Freeswitch sends SIP Invite with the SDP information that includes >> the video details. I don’t have any problems when I disable mod_av and >> include PCMU, PCMA, and OPUS. The latest versions of X-Lite and Bria 5 >> also crash with mod_av enabled. >> >> Bria 4 and other models of ip phones (Polycom, Yealink, Grandstream) >> don’t seem to have a problem. >> >> Any ideas what could be wrong? >> >> - Joe Barrero >> >> , >> >> _________________________________________________________________________ >> 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 >> >> >

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PortAudio and Windows devices
by Clive Lansink 07 Sep '17

07 Sep '17

Hi everyone. Can someone please help with the following problem? I have spent several hours on this today, just googling around, and although I think I am not the only one with this problem, I haven't actually found the solution. Problem is PortAudio reports there are no input and output devices. I am running Freeswitch on Windows 10 64-bit. The Freeswitch installer is FreeSWITCH-1.6.18-x64-Release.msi which I downloaded recently from files.freeswitch.org. I should explain that I have been using Freeswitch for a number of years and PortAudio used to work, even on this Windows 10 PC. The problem seems to be with the more recent versions of Freeswitch. We know that Freeswitch for Windows includes PortAudio. That's good because I don't know how to build Freeswitch for Windows. I just know how to install and use it. I think I read somewhere that the Freeswitch PortAudio is built for DirectEx. I came across the Dxdiag tool which tells me about DirectEx devices. I have several input and output devices on this machine, but Dxdiag only lists one output device and it lists no input devices. It tells me I have DirectEx version 12. I do a lot of programming in Python and that includes a Python module that provides bindings for PortAudio. That code still works. I read that PortAudio by default is built for MME, or maybe it is PyAudio that is built for MME. Whatever the case, that code works. But if Freeswitch's PortAudio module is built for DirectEx, could there be a problem with DirectEx on Windows 10? I just get the impression that Microsoft is favouring newer Windows sound APIs. Issuing the pa devlist command just lists nothing so it is as if it is just not seeing any devices, but they are certainly there. I think this has been a problem for several years because I have Freeswitch version 1.5.15 on another Windows 10 64-bit PC and PortAudio on that PC has the same problem. I just haven't had to use it on that PC. By the way, I did try running Freeswitch in console mode with administrator privileges and it made no difference. I hope someone can suggest something to help me fix this problem. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150

1 0

v8 failing to build master branch
by John Covici 07 Sep '17

07 Sep '17

Hi. I am trying to update to masterand v8 will not build and I have no idea even what this error means. Here is the complete build log. It is saying ninja subcommand failed. https://covici.com/owncloud/index.php/s/6MBIR4hnWdu8lWG Am I doing something wrong or should I file a bug? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Video Conferencing freezing and lag in v1.8 and master
by Peter Baines (lists) 07 Sep '17

07 Sep '17

Hello, I'm trying to get a simple video conference between two endpoints. I'm constantly seeing freeze and lag but I don't see any issues around CPU/memory/network that may cause this, has anyone come across something similar? I do notice in the debug log it keeps changing to large resolutions: e.g. VPX reset encoder picture from 352x288 to 720x1280 2048 BW Is there away to define the max resolution ? These are the current settings I'm using, in vars.xml I have: <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=2mb"/> <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=2mb"/> This is the profile from conference.conf.xml (tried both transcode and mux): <profile name="default"> <param name="domain" value="$${domain}"/> <param name="rate" value="8000"/> <param name="channels" value="2"/> <param name="interval" value="20"/> <param name="energy-level" value="0"/> <param name="caller-id-name" value="$${outbound_caller_name}"/> <param name="caller-id-number" value="$${outbound_caller_id}"/> <param name="comfort-noise" value="false"/> <param name="conference-flags" value="minimize-video- encoding|mintwo|manage-inbound-video-bitrate" /> <param name="video-mode" value="mux"/> <param name="video-layout-name" value="2x1"/> <param name="video-canvas-size" value="320x240"/> <param name="video-canvas-bgcolor" value="#FFA500"/> <param name="video-layout-bgcolor" value="#800080"/> <param name="video-codec-bandwidth" value="2mb" /> <param name="video-fps" value="10" /> <param name="video-quality" value="1" /> <param name="video-kps-debounce" value="5000" /> </profile> The dialplan bridges like: <action application="set" data="video_use_dedicated_encoder=true"/> <action application="conference" data="bridge:${uuid}@default: sofia/from_opensips/${destination_number}@${domain}+flags{mintwo}" /> Regards, Peter

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Cisco IP Phones & Counterpath Softphones Crashing w/ mod_av Enabled
by Joseph Barrero 07 Sep '17

07 Sep '17

Hi, I’m running Freeswitch v1.6.19. When mod_av is enabled, inbound calls to Cisco SPA504g and SPA514g ip phones crash as they start to ring. The crash seems to occur immediately after Freeswitch sends SIP Invite with the SDP information that includes the video details. I don’t have any problems when I disable mod_av and include PCMU, PCMA, and OPUS. The latest versions of X-Lite and Bria 5 also crash with mod_av enabled. Bria 4 and other models of ip phones (Polycom, Yealink, Grandstream) don’t seem to have a problem. Any ideas what could be wrong? - Joe Barrero

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SIP-Client / IOS / PJSIP / Error when CANCEL
by Thorsten Göllner 07 Sep '17

07 Sep '17

Hi, currently we are developing a SIP-Client for iOS (using PJSIP/PJSUA with the current version). We use Freeswitch 1.6.18 (64 bit) on a bare metal server. Windows-Clients (such as Blink or Zoiper) work fine. And our iOS-App works fine also. BUT when we initialze an outgoing call, wait for ringing on the other hand and hangup up when it's ringing, we face an error. Freeswitch gives us "481 Call/Transaction Does Not Exist". I attached the corresponding SIP-Trace and you can see the error at the last block. The interesting thing: If we activate PJSIP with TCP and(!) UDP we face the error. BUT when we use only UDP it works fine. Is it a bug within PJSIP? Or anyone see an hint wihtin the SIP-Trace? Thanks in advance! recv 1247 bytes from udp/[77.119.248.251]:5060 at 11:36:21.330337: ------------------------------------------------------------------------ INVITE sip:+436641540180@telequest-occ.0049.org SIP/2.0 Via: SIP/2.0/UDP 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 Via: SIP/2.0/UDP 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh From: <sip:102@telequest-occ.0049.org>;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu To: <sip:+436641540180@telequest-occ.0049.org> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z CSeq: 17632 INVITE Contact: <sip:102@77.119.248.251> max-forwards: 69 supported: replaces supported: 100rel supported: timer supported: norefersub session-expires: 1800 min-se: 90 user-agent: AdCall Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Type: application/sdp Content-Length: 480 v=0 o=- 3713765781 3713765781 IN IP4 77.119.248.251 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 7070 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 77.119.248.251 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.178.110 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 ------------------------------------------------------------------------ send 432 bytes to udp/[77.119.248.251]:5060 at 11:36:21.330679: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 Via: SIP/2.0/UDP 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh From: <sip:102@telequest-occ.0049.org>;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu To: <sip:+436641540180@telequest-occ.0049.org> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z CSeq: 17632 INVITE User-Agent: FreeSWITCH Content-Length: 0 ------------------------------------------------------------------------ send 945 bytes to udp/[77.119.248.251]:5060 at 11:36:21.331652: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 Via: SIP/2.0/UDP 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh From: <sip:102@telequest-occ.0049.org>;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu To: <sip:+436641540180@telequest-occ.0049.org>;tag=0FgQ57apej24p Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z CSeq: 17632 INVITE User-Agent: FreeSWITCH Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="telequest-occ.0049.org", nonce="01e39fee-93b0-11e7-9b0b-45e8f1c3918a", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 475 bytes from udp/[77.119.248.251]:5060 at 11:36:21.409776: ------------------------------------------------------------------------ ACK sip:+436641540180@telequest-occ.0049.org SIP/2.0 Via: SIP/2.0/UDP 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 Via: SIP/2.0/UDP 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh From: <sip:102@telequest-occ.0049.org>;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu To: <sip:+436641540180@telequest-occ.0049.org>;tag=0FgQ57apej24p Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z CSeq: 17632 ACK max-forwards: 69 Content-Length: 0 ------------------------------------------------------------------------ recv 1439 bytes from tcp/[77.119.248.251]:60186 at 11:36:21.457742: ------------------------------------------------------------------------ INVITE sip:+436641540180@telequest-occ.0049.org SIP/2.0 Via: SIP/2.0/TCP 192.168.178.110:60186;rport;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias Max-Forwards: 70 From: sip:102@telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu To: sip:+436641540180@telequest-occ.0049.org Contact: <sip:102@192.168.178.110:5060;ob> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z CSeq: 17633 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: AdCall Proxy-Authorization: Digest username="102", realm="telequest-occ.0049.org", nonce="01e39fee-93b0-11e7-9b0b-45e8f1c3918a", uri="sip:+436641540180@telequest-occ.0049.org", response="c7fef167bd3ecc40bdaeaaef9c9b8085", algorithm=MD5, cnonce="raNBupUIEw5CyLL8MQbvYrk5txEd3Zep", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 482 v=0 o=- 3713765781 3713765781 IN IP4 192.168.178.110 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 192.168.178.110 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.178.110 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 ------------------------------------------------------------------------ send 380 bytes to tcp/[77.119.248.251]:60186 at 11:36:21.457978: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.178.110:60186;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 From: sip:102@telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu To: sip:+436641540180@telequest-occ.0049.org Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z CSeq: 17633 INVITE User-Agent: FreeSWITCH Content-Length: 0 ------------------------------------------------------------------------ send 1193 bytes to tcp/[77.119.248.251]:60186 at 11:36:24.966185: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 192.168.178.110:60186;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 From: sip:102@telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu To: <sip:+436641540180@telequest-occ.0049.org>;tag=1r9F72USBUrQj Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z CSeq: 17633 INVITE Contact: <sip:+436641540180@88.217.132.183:5060;transport=udp> User-Agent: FreeSWITCH Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 203 Remote-Party-ID: "+436641540180" <sip:+436641540180@telequest-occ.0049.org>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1504753876 1504753877 IN IP4 88.217.132.183 s=FreeSWITCH c=IN IP4 88.217.132.183 t=0 0 m=audio 23108 RTP/AVP 3 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtcp:23109 IN IP4 88.217.132.183 ------------------------------------------------------------------------ recv 401 bytes from tcp/[77.119.248.251]:60186 at 11:36:28.869708: ------------------------------------------------------------------------ CANCEL sip:+436641540180@telequest-occ.0049.org SIP/2.0 Via: SIP/2.0/TCP 192.168.178.110:5060;rport;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias Max-Forwards: 70 From: sip:102@telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu To: sip:+436641540180@telequest-occ.0049.org Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z CSeq: 17633 CANCEL User-Agent: AdCall Content-Length: 0 ------------------------------------------------------------------------ send 400 bytes to tcp/[77.119.248.251]:60186 at 11:36:28.869807: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/TCP 192.168.178.110:5060;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 From: sip:102@telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu To: <sip:+436641540180@telequest-occ.0049.org>;tag=1r9F72USBUrQj Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z CSeq: 17633 CANCEL Content-Length: 0 ------------------------------------------------------------------------ Best regards, Thorsten

2 3

Stop packet radius
by Nicolas Paleari 07 Sep '17

07 Sep '17

Friends, I need help, I get incorrect information in the stop packet of raduis that sends freeswitch, in the field dst-number-out the number has %23 instead of #, I do not understand why it replaces it,dst-number-out should be equal to field dst-number-in = 492482 # 541151995330 Send stop package where you see the problem: [2017-09-01 19:03:46,119] DEBUG WORKER 38 - [pkt#175515/ACCT-STOP] --- START: [pkt#175515/ACCT-STOP] -------------------------------------- pdd-time => 0 called-station-id => 492482%23541151995330 nas-port => 0 dst-number-out => 492482%23541151995330 h323-conf-id => cd0f2a0d-4b7f-414c-b5be-d1b937a414db h323-setup-time => 2017-09-01T15:03:36.328927-0400 acct-status-type => Stop h323-disconnect-cause => 10 dst-gw-ip => 190.210.240.37 h323-disconnect-time => 2017-09-01T15:03:43.929039-0400 h323-call-id => 7a34a7be-d643-4f82-a66a-cd1eb0658b6c src-gw-ip => 190.210.240.37 dst-number-in => 492482#541151995330 h323-call-origin => originate nas-ip-address => 107.170.35.75 src-number-in => 7680858053 dst-gw-name => 492482#541151995330 h323-connect-time => 2017-09-01T15:03:38.268917-0400 src-gw-name => 7680858053 acct-session-time => 5 acct-delay-time => 0 src-number-out => 7680858053 calling-station-id => 7680858053 --- END: [pkt#175515/ACCT-STOP] -------------------------------------- I hope you can help me Thank you <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Libre de virus. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

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Timer issue
by Sergey Safarov 06 Sep '17

06 Sep '17

I have FreeSwitch docker container and cannot receive fax. In FreeSwitch log i see error 2017-09-06 20:08:06.963387 [ERR] switch_core_timer.c:117 Timer is not properly configured. But during freeswitch boots up i not see any timer related errors. This error arise during rxfax application. I have tested timer using "timer_test" command and log results like freeswitch(a)freeswitch1.kazoo> timer_test Avg: 19.999ms Total Time: 999.999ms 2017-09-06 20:09:50.293389 [CONSOLE] mod_commands.c:955 Timer Test: 1 sleep 20 20000 2017-09-06 20:09:50.313385 [CONSOLE] mod_commands.c:955 Timer Test: 2 sleep 20 20003 2017-09-06 20:09:50.333385 [CONSOLE] mod_commands.c:955 Timer Test: 3 sleep 20 20005 2017-09-06 20:09:50.353384 [CONSOLE] mod_commands.c:955 Timer Test: 4 sleep 20 19995 2017-09-06 20:09:50.373387 [CONSOLE] mod_commands.c:955 Timer Test: 5 sleep 20 19998 Could you advice where to look to find root of issue?

1 0

session timer - enable/disable per call
by Gabriel Kuri 06 Sep '17

06 Sep '17

Is it possible to either: a) enable/disable the session timer per call, via the dialplan? b) change the session timeout value per call, via the dialplan? Currently, it's set in the sofia profile, either: <param name="enable-timer" value="false"/> or <param name="session-timeout" value="xxx"/> However, it would be great if it could be changed/set per call. If you ask why, a particular device when it receives a re-invite mid-call decides to re-invite an already established T.38 call to PCMU which breaks the FAX transmission. Yes, this is a long fax :) Unfortunately, the vendor won't fix, so it'd be nice for these devices to be able to simply disable or set the timer to a really long time in the dialplan. Thanks ...

2 4

Send a Client SIP Register to a PBX or to a provider
by Alexander Haugg 06 Sep '17

06 Sep '17

Hi All, is it possible, that the FS receive a client SIP register and send for this a client register to a PBX? Thanks a lot!

2 1