February 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Sip Session Timer on the fly
by Mickael Hubert 06 Feb '20

06 Feb '20

Hi all, I would like to know, if there is a way to FS is "transparent" with SST ? Currently, you can activate or deactivate SST on profile: <param name="enable-timer" value="false / true"/> But I want if client use SST, FS sends it to the B leg, and if client doesn't use SST, FS doesn't send it. thanks in advance

1 0

Verto ice restart
by David P 06 Feb '20

06 Feb '20

I'm looking for best practices about how to detect a) that an ice restart is needed (in verto), and b) how to invoke the restart. FWIW, we use FS v1.8 and will soon try 1.10.2. For detection, I think it could be done either of these ways: a1) In the 1st arg to jQuery.verto(), provide an object with property deviceParams.onICEComplete set to a callback. In the callback, iterate over mediaData.candidateList. For example, check if all candidates are private IPs but all your callers are expected to have public IPs. a2) In the 2nd arg to jQuery.verto(), provide an object with property 'onDialogState' set to a callback. In the callback, if arg1.state.name is "hangup" and arg1.cause is "INCOMPATIBLE DESTINATION". (Ignoring that this hangup cause occurs for other reasons also.) To do the actual restart... In method (a1) we should have access to the peer connection object via arg1.peer. Would it mess up the state of verto to do arg1.peer.createOffer({iceRestart:true}) ? Do we want to limit this reaction to happen just once per call attempt in order to avoid possibly many chained failures? In method (a2), I'm not sure there's any way to access the peer connection. I searched jquery.FSRTC.js for "restart" in case there's a standard way of doing this, but found no match. Cheers, David

1 0

Hostname not being considered by fsctl recover
by Guilherme Lima Hernandez Rincão 04 Feb '20

04 Feb '20

Hello. I've got 3 FS 1.6.20 instances sharing a postgre db, with different hostnames, two of them receiving calls and the third one acting as a failover. Calls are correctly added with their respective hostname to the recovery table but running fsctl recover on the failover, all calls are recovered, not just the ones that match its hostname. Is this the expected behaviour? Looking at the source code <https://github.com/signalwire/freeswitch/blob/cd5d72ac744208ae63b0b81d18410…>, it seems that hostname is not used as a filter, but the profile name is. That does not match the wiki <https://freeswitch.org/confluence/display/FREESWITCH/High+Availability> documentation: *"hostname on both machines needs to be the same, as the sql query to recover calls selects by hostname"*.

1 0

FreeSWITCH and libsng_isdn
by Sam van Herwaarden 04 Feb '20

04 Feb '20

Hi, I'm wondering if anyone has recent experience using FreeSWITCH 1.10 on Debian 10 with libsng_isdn. I'm compiling from source and getting the errors below. I installed the current libsng_isdn, which is version 8.3.4. However, the FreeTDM instructions <https://freeswitch.org/confluence/display/FREESWITCH/FreeTDM> mention I need 1.2.0+ which makes me wonder whether these instructions might be outdated. Would an alternative be to just install DAHDI + libpri? I haven't tried that yet but would be happy to. This is all kind of new to me and I'm not entirely sure which components perform which tasks. I have wanpipe installed and am using a PRI/E1 line, I can see that my first port is connected (service wanrouter status), the card is a Sangoma A104. If I uninstall libsng_isdn I'm able to build and install FreeSWITCH with mod_freetdm but the errors below seem to be blocking me when I try to set it up with an ISDN library. Kind regards, Sam CC ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo In file included from /usr/include/sng_isdn/sng_isdn.h:32, from src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn.h:55, from src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn.c:38: /usr/include/sng_isdn/ssi.x:1253:20: error: inline function ‘SDisInt’ declared but never defined [-Werror] EXTERN INLINE S16 SDisInt ARGS((void )); ^~~~~~~ /usr/include/sng_isdn/ssi.x:1252:20: error: inline function ‘SEnbInt’ declared but never defined [-Werror] EXTERN INLINE S16 SEnbInt ARGS((void )); ^~~~~~~ /usr/include/sng_isdn/ssi.x:1245:19: error: inline function ‘SSetIntPend’ declared but never defined [-Werror] EXTERN INLINE S16 SSetIntPend ARGS((U16 id, Bool flag)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1244:19: error: inline function ‘SChkNxtDBuf’ declared but never defined [-Werror] EXTERN INLINE S16 SChkNxtDBuf ARGS((Buffer *mBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1243:19: error: inline function ‘SGetNxtDBuf’ declared but never defined [-Werror] EXTERN INLINE S16 SGetNxtDBuf ARGS((Buffer *mBuf, Buffer **dBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1242:19: error: inline function ‘SInitNxtDBuf’ declared but never defined [-Werror] EXTERN INLINE S16 SInitNxtDBuf ARGS((Buffer *mBuf)); ^~~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1237:19: error: inline function ‘SUpdMsg’ declared but never defined [-Werror] EXTERN INLINE S16 SUpdMsg ARGS((Buffer *mBuf, Buffer *dBuf, MsgLen mLen)); ^~~~~~~ /usr/include/sng_isdn/ssi.x:1236:19: error: inline function ‘SGetDataTx’ declared but never defined [-Werror] EXTERN INLINE S16 SGetDataTx ARGS((Buffer *dBuf, Data **dat, MsgLen *mLen)); ^~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1235:19: error: inline function ‘SGetDataRx’ declared but never defined [-Werror] EXTERN INLINE S16 SGetDataRx ARGS((Buffer *dBuf, MsgLen pad, Data **dat, MsgLen *mLen)); ^~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1234:19: error: inline function ‘SRemDBufPre’ declared but never defined [-Werror] EXTERN INLINE S16 SRemDBufPre ARGS((Buffer *mBuf, Buffer **dBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1233:19: error: inline function ‘SRemDBufPst’ declared but never defined [-Werror] EXTERN INLINE S16 SRemDBufPst ARGS((Buffer *mBuf, Buffer **dBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1232:19: error: inline function ‘SAddDBufPre’ declared but never defined [-Werror] EXTERN INLINE S16 SAddDBufPre ARGS((Buffer *mBuf, Buffer *dBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1231:19: error: inline function ‘SAddDBufPst’ declared but never defined [-Werror] EXTERN INLINE S16 SAddDBufPst ARGS((Buffer *mBuf, Buffer *dBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1068:20: error: inline function ‘SPutDBuf’ declared but never defined [-Werror] EXTERN INLINE S16 SPutDBuf ARGS((Region region,Pool pool,Buffer *buf)); ^~~~~~~~ /usr/include/sng_isdn/ssi.x:1067:20: error: inline function ‘SGetDBuf’ declared but never defined [-Werror] EXTERN INLINE S16 SGetDBuf ARGS((Region region,Pool pool,Buffer * *bufPtr)); ^~~~~~~~ /usr/include/sng_isdn/ssi.x:1058:20: error: inline function ‘SDequeueLast’ declared but never defined [-Werror] EXTERN INLINE S16 SDequeueLast ARGS((Buffer * *bufPtr,Queue *q)); ^~~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1057:20: error: inline function ‘SDequeueFirst’ declared but never defined [-Werror] EXTERN INLINE S16 SDequeueFirst ARGS((Buffer * *bufPtr,Queue *q)); ^~~~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1056:20: error: inline function ‘SQueueLast’ declared but never defined [-Werror] EXTERN INLINE S16 SQueueLast ARGS((Buffer *buf,Queue *q)); ^~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1055:20: error: inline function ‘SQueueFirst’ declared but never defined [-Werror] EXTERN INLINE S16 SQueueFirst ARGS((Buffer *buf,Queue *q)); ^~~~~~~~~~~ cc1: all warnings being treated as errors

1 1

Unset variable in dialstring
by Markus Bönke 03 Feb '20

03 Feb '20

Hello NG, I know it is possible to unset a variable in the dial plan by using the „unset“ application like: <application="unset" data="variable_name“> But is it also possible to unset a variable within a bridge command like: bridge({variable_name="_unset_"}sofia/gateway/mygateway/+49123456789) _unset_ does not work with FS 1.10.2 but maybe there is any other „special value“ to unset? Thanks and regards Markus

3 7

Debian 9 or 10 for new installs
by Lloyd Aloysius 03 Feb '20

03 Feb '20

Hi All Looking for a recommendation. Does Debian 10 works reliably for freeswitch v1.10 or stick with Debian 9. Thanks Lloyd

3 2

setting mod_portaudio priority codec call out
by Tamer Higazi 03 Feb '20

03 Feb '20

Hi people, I am looking to configure the priority codec with mod_portaudio. When I make an outbound call, pa uses PCMA as primary codec... When I receive from the same person a call, G722 is used. Any chance enforcing G722 as priority codec with PCMA as fallback by placing an outbound call? Thank you console output: o=root 1240735378 1240735378 IN IP4 21x.xx.xxx.xxx s=Asterisk PBX 11.21.1 c=IN IP4 21x.xx.xxx.xxx t=0 0 m=audio 12878 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2020-02-03 07:56:26.478728 [NOTICE] sofia.c:7314 Pre-Answer sofia/external/xxxxxxxxxx! 2020-02-03 07:56:26.478728 [DEBUG] mod_portaudio.c:809 portaudio/1xxxxxxxxxx CHANNEL KILL 2020-02-03 07:56:26.478728 [DEBUG] switch_channel.c:3565 (sofia/external/xxxxxxxxxx) Callstate Change DOWN -> EARLY 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[L16:100:48000:20:768000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[L16:100:48000:20:768000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[L16:100:48000:20:768000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 101@8000 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:3837 Set Codec sofia/external/xxxxxxxxxx PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2020-02-03 07:56:26.478728 [DEBUG] switch_core_codec.c:111 sofia/external/xxxxxxxxxx Original read codec set to PCMA:8 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5851 Set telephone-event payload to 101@8000 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5909 sofia/external/xxxxxxxxxx Set 2833 dtmf send payload to 101 recv payload to 101 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:8659 AUDIO RTP [sofia/external/xxxxxxxxxx] 84.75.67.75 port 32108 -> 21x.xx.xxx.xxx port 12878 codec: 8 ms: 20 2020-02-03 07:56:26.478728 [DEBUG] switch_rtp.c:4408 Starting timer [soft] 160 bytes per 20ms 2020-02-03 07:56:26.498733 [DEBUG] switch_core_media.c:8973 sofia/external/xxxxxxxxxx Set 2833 dtmf send payload to 101 2020-02-03 07:56:26.498733 [DEBUG] switch_core_media.c:8980 sofia/external/xxxxxxxxxx Set 2833 dtmf receive payload to 101 2020-02-03 07:56:26.498733 [DEBUG] switch_core_media.c:9003 sofia/external/xxxxxxxxxx Set rtp dtmf delay to 40 2020-02-03 07:56:26.498733 [DEBUG] switch_ivr_originate.c:3809 Originate Resulted in Success: [sofia/external/xxxxxxxxxx] 2020-02-03 07:56:26.498733 [DEBUG] mod_portaudio.c:809 portaudio/1xxxxxxxxxx CHANNEL KILL 2020-02-03 07:56:26.498733 [DEBUG] mod_portaudio.c:809 portaudio/1xxxxxxxxxx CHANNEL KILL 2020-02-03 07:56:26.498733 [DEBUG] switch_channel.c:2130 (portaudio/1xxxxxxxxxx) Callstate Change RING_WAIT -> ACTIVE 2020-02-03 07:56:26.498733 [DEBUG] mod_portaudio.c:809 portaudio/1xxxxxxxxxx CHANNEL KILL 2020-02-03 07:56:26.498733 [DEBUG] mod_portaudio.c:809 portaudio/1xxxxxxxxxx CHANNEL KILL 2020-02-03 07:56:26.498733 [DEBUG] switch_ivr_bridge.c:1796 (sofia/external/xxxxxxxxxx) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2020-02-03 07:56:26.498733 [DEBUG] switch_core_state_machine.c:585 (sofia/external/xxxxxxxxxx) Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 2)

1 0

Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue
by Tom Lynn 03 Feb '20

03 Feb '20

I followed your steps, and yes, pa commands function when freeswitch is run from the command line. Nice! I tried running it as both root and as user tom. Both times I was able to use the pa commands with success. I tried running user tom group user from the systemd startup and was not successful using pa commands. The question becomes, why does this not work using the freeswitch.service configuration file supplied with the packaged freeswitch? I installed freeswitch-meta-all. Tom On Sun, Feb 2, 2020 at 9:02 AM Tamer Higazi via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi <th982a(a)googlemail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Sun, 2 Feb 2020 18:02:09 +0100 > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue > Dear Tom, > > This is not true. > I am running freeswitch as the user I am logged in. > > as root: > chown -R tamer:users /opt/freeswitch > > as user: > /opt/freeswitch/bin/freeswitch -nc > > as user: > fs_cli -r > > and you will see portaudio runs without any problems. > > I have it in combination with pulseaudio running. > > > best, Tamer > > On 2020-02-02 17:01, Tom Lynn wrote: > > I do not have pulse audio installed. > > > > I can access alsamixer and aplay from the shell while logged in as > > either root or freeswitch without issues. > > > > mod_portaudio only works WHEN I'm running freeswitch under group root, > > which I will not be able to do in a production environment. > > > > On Sun, Feb 2, 2020 at 7:24 AM Tamer Higazi via FreeSWITCH-users > > <freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org>> wrote: > > > > > > > > > > ---------- Forwarded message ---------- > > From: Tamer Higazi <th982a(a)googlemail.com > > <mailto:th982a@googlemail.com>> > > To: freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org> > > Cc: > > Bcc: > > Date: Sun, 2 Feb 2020 15:22:31 +0100 > > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > > permissions issue > > Dear Mark, > > > > I tell you exactly why. > > > > That is less a permission issue. > > > > if you open the shell as "root" and type alsamixer you won't be > > capable > > to access the soundcard because "root" is not in the 1st) sound group > > and 2nd) root ha no access to the pulseaudio server > > > > This is why I MUST run freeswitch as the user that is currently > > logged in. > > > > mod_alsa doesn't work at all. I didn't get it running, only > > mod_portaudio.... > > > > > > best, Tamer > > > > > > On 2020-02-02 07:59, Tom Lynn wrote: > > > For what it's worth, I'm running into this very same issue over 4 > > > years after the original poster. I suspect that leaving off the > > group > > > option is leaving the process running as group root, which would > > mimic > > > my attempt at having the service run as user freeswitch/group root, > > > which allows mod_portaudio to function. > > > > > > I've posted here in the mail list, on IRC and in Slack, but the > > only > > > bite was someone hinting about freeswitch user needing to be in > > group > > > audio, which it is. This is broken. I've looked at submitting a > > JIRA > > > on this, but my login doesn't appear to allow it. > > > > > > On Fri, Sep 11, 2015 at 9:45 AM Mark Haun <haunma(a)keteu.org > > <mailto:haunma@keteu.org> > > > <mailto:haunma@keteu.org <mailto:haunma@keteu.org>>> wrote: > > > > > > covici(a)ccs.covici.com <mailto:covici@ccs.covici.com> > > <mailto:covici@ccs.covici.com <mailto:covici@ccs.covici.com>> > > > [covici(a)ccs.covici.com <mailto:covici@ccs.covici.com> > > <mailto:covici@ccs.covici.com <mailto:covici@ccs.covici.com>>] > wrote: > > > > If using alsa, did you check the permission of /dev/snd > > and its > > > > children? > > > > > > Yes. In fact, both alsa operations (aplay) and portaudio > > > enumeration work > > > correctly when running as the freeswitch user. All of this is > > > detailed in > > > my initial post, two or three weeks ago. > > > > > > Mark > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > > <mailto:consulting@freeswitch.org <mailto:consulting@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 > > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > > <mailto:FreeSWITCH-users@lists.freeswitch.org > > <mailto:FreeSWITCH-users@lists.freeswitch.org>> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > _________________________________________________________________________ > > > > > > The FreeSWITCH project is sponsored by SignalWire > > https://signalwire.com > > > Enhance your FreeSWITCH install with disruptive priced SMS and > > PSTN services. > > > Build your next product on our scalable cloud platform. > > > > > > Join our online community to chat in real time > > https://signalwire.community > > > > > > Professional FreeSWITCH Services > > > sales(a)freeswitch.com <mailto:sales@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 > > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > https://freeswitch.com > > > > > > > > > > ---------- Forwarded message ---------- > > From: Tamer Higazi via FreeSWITCH-users > > <freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org>> > > To: freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org> > > Cc: > > Bcc: > > Date: Sun, 02 Feb 2020 07:24:11 -0800 (PST) > > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > > permissions issue > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire > > https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and > > PSTN services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > > https://signalwire.community > > > > Professional FreeSWITCH Services > > sales(a)freeswitch.com <mailto:sales@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 > > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > 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 > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Sun, 02 Feb 2020 09:02:50 -0800 (PST) > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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

2 2

Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue
by Tom Lynn 02 Feb '20

02 Feb '20

I do not have pulse audio installed. I can access alsamixer and aplay from the shell while logged in as either root or freeswitch without issues. mod_portaudio only works WHEN I'm running freeswitch under group root, which I will not be able to do in a production environment. On Sun, Feb 2, 2020 at 7:24 AM Tamer Higazi via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi <th982a(a)googlemail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Sun, 2 Feb 2020 15:22:31 +0100 > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue > Dear Mark, > > I tell you exactly why. > > That is less a permission issue. > > if you open the shell as "root" and type alsamixer you won't be capable > to access the soundcard because "root" is not in the 1st) sound group > and 2nd) root ha no access to the pulseaudio server > > This is why I MUST run freeswitch as the user that is currently logged in. > > mod_alsa doesn't work at all. I didn't get it running, only > mod_portaudio.... > > > best, Tamer > > > On 2020-02-02 07:59, Tom Lynn wrote: > > For what it's worth, I'm running into this very same issue over 4 > > years after the original poster. I suspect that leaving off the group > > option is leaving the process running as group root, which would mimic > > my attempt at having the service run as user freeswitch/group root, > > which allows mod_portaudio to function. > > > > I've posted here in the mail list, on IRC and in Slack, but the only > > bite was someone hinting about freeswitch user needing to be in group > > audio, which it is. This is broken. I've looked at submitting a JIRA > > on this, but my login doesn't appear to allow it. > > > > On Fri, Sep 11, 2015 at 9:45 AM Mark Haun <haunma(a)keteu.org > > <mailto:haunma@keteu.org>> wrote: > > > > covici(a)ccs.covici.com <mailto:covici@ccs.covici.com> > > [covici(a)ccs.covici.com <mailto:covici@ccs.covici.com>] wrote: > > > If using alsa, did you check the permission of /dev/snd and its > > > children? > > > > Yes. In fact, both alsa operations (aplay) and portaudio > > enumeration work > > correctly when running as the freeswitch user. All of this is > > detailed in > > my initial post, two or three weeks ago. > > > > Mark > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org <mailto:consulting@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 > > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > 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 > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Sun, 02 Feb 2020 07:24:11 -0800 (PST) > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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

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Stumped by mod_portaudio permissions issue
by Mark Haun 02 Feb '20

02 Feb '20

I built freeswitch version 1.7.0+git~20150730T192909Z~ab7f83c654~32bit for my ARM-based home server (an NVIDIA Jetson board), with the intention of using mod_portaudio with a wireless USB headset for my main home phone. It works well when running freeswitch directly as root, but when I configured freeswitch to run at system startup, mod_portaudio could no longer find any audio devices. No problem, I thought, it's a simple permissions issue. I am using the example debian init script from the wiki (this is a Jessie install with SysV init rather than systemd). It runs freeswitch as user freeswitch, group daemon using the -u and -g command-line options. So I added "freeswitch" to the audio group, then used sudo to verify that I could play sound files with "aplay" as the freeswitch user. That worked, but it did not solve the mod_portaudio failure. Thinking there must be an issue with portaudio separate from ALSA, I fetched and built the "pa_devs" utility which enumerates and prints the available audio devices. (This is basically what mod_portaudio is doing on startup.) Running as the freeswitch user, that worked too. Hmmmm. I tried running freeswitch from the command line using sudo rather than the -u and -g options, i.e. $ sudo freeswitch /usr/local/freeswitch/bin/freeswitch -nc and that worked as well! Then I started experimenting with -u and -g and discovered that $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch allows mod_portaudio to load correctly, whereas $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch -g daemon causes mod_portaudio to fail. Can anyone explain what's happening here? In both cases the process UID, GID, EUID, EGID, FUID, and FGID are the same (freeswitch:daemon) as displayed with "ps xao pid,uid,gid,euid,egid,fuid,fgid". Could this be a bug in freeswitch? There are a couple of threads from 2-3 years ago in the archives, where mod_portaudio permissions issues were never resolved and the solution remained a mystery. For the time being I guess this (not using -g) is a workaround, but I'd sure like to understand what is going on. Mark

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