January 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

5 7

ICE/DTLS handshake
by Mirko Brankovic 25 Oct '19

25 Oct '19

HI, Has anyone experienced DTLS handshake takes 5s to get to SETUP state: > 2017-09-21 11:40:40.021178 [INFO] switch_rtp.c:3515 Changing audio DTLS > state from OFF to HANDSHAKE > 2017-09-21 11:40:45.319457 [INFO] switch_rtp.c:3172 Changing audio DTLS > state from HANDSHAKE to SETUP In network dump i see that answering side is not sending STUN for this 5s and then suddenly answers last 5 STUNs from A side. Has anyone encountered this kind of problem ? I have a pcap if necessary... Thanks, -- Regards, Mirko

3 7

[mod avmd][1.6.X vs 1.8.X]
by Bilal Abbasi 22 Mar '19

22 Mar '19

Hi Users, I was using mod_avmd in FreeSWITCH version 1.6.20, it was working really well, but now as i shifted to new version i noticed that VM detection number get dropped. While investigating i noticed that the default parameters in 1.8.2 avmd.xml.conf are unable to detect the attached beep sound while 1.6.20 was able to detect that, i tried to change certain parameters in the avmd.xml.conf file but no luck. I was wondering if somebody has used avmd on 1.8.X and have the correct parameters/values OR can somebody point me the right direction to detect the attached file. Regards Bilal Abbasi

3 6

MOD_CALLCENTER become unload and unknown while freeswitch is still running
by Chhorm Chhatra 20 Mar '19

20 Mar '19

Hi, Currently, I faced a very strange issue that 1) After several days of FreeSWITCH running, I could not connect to its CLI at all. But Freeswitch is still working as it is. I have to tail the log file to debug this issue. 2) When the issue (1) happens, the application *callcenter *is unknown and therefore the callcenter failed. FS version: 1.6 Any help would be appreciated. Thank you in advance. Best regards, Chhorm Chhatra

3 3

Change SIP Contact header on b-leg
by Кирилл Преслер 22 Feb '19

22 Feb '19

Provider not accepting calls because of "bad" Contact header sip:gw+ss@10.50.244.15:5060;transport=udp;gw=ss ITSP needs sip:username@10.50.244.15:5060, where username should be passed from dialplan for every call and there is no params. I've tried following variables with set and export, but nothing seems to work. sip_contact_user sip_outgoing_contact_uri sip_invite_contact_params

3 2

Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION
by Paul Muaddib 13 Feb '19

13 Feb '19

vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,PCMU"/> My phones, all Grandstream GXP2140, accept PCMA and PCMU I don't have trouble with other people calling me Am Mi., 22. Aug. 2018 um 15:21 Uhr schrieb Thorsten Göllner < tg-maillistings(a)level5.de>: > Only PCMA@8000 is offered. Check device if codec is activated. > > Am 21.08.2018 um 15:06 schrieb Paul Muaddib: > > Hi, > > I have a rarely error where I get this failure message in the log file. I > am bit wondering because this is suposed to be a normal voice call and not > a fax. > > Best regards, > Paul. > > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [INFO] > switch_ivr_originate.c:1215 Sending early media > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [ERR] > mod_sofia.c:2343 CODEC NEGOTIATION ERROR. SDP: > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 v=0^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 o=- 20012711 20012711 IN IP4 > 217.0.15.67^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 s=on transit^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 c=IN IP4 X.X.X.X^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 t=0 0^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 m=audio 13274 RTP/AVP 8 118 126 101^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:8 PCMA/8000^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:118 PCMA/8000^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:126 CLEARMODE/8000^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:101 telephone-event/8000^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=fmtp:101 0-15^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=gpmd:118 vbd=yes^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=ptime:20^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 m=image 11232 udptl t38^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxVersion:0^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38MaxBitRate:14400^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxMaxBuffer:3000^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxMaxDatagram:500^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 > a=T38FaxRateManagement:transferredTCF^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxUdpEC:t38UDPRedundancy^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [NOTICE] > switch_channel.c:3515 Hangup sofia/external/+ > 49XXXXXXXXX(a)sip-trunk.telekom.de [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales@freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://clu… > > FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… > > >

6 9

DTMF negotiation
by kaiduan xie 04 Feb '19

04 Feb '19

Hi, We are encountering the following DTMF issue, FreeSwitch adds telephone-event support in INVITE to SBC, SBC replies back without telephone-event support in 200. However FreeSwitch still sends out DTMF in RFC 2833 event. I think FreeSwitch should send out DTMF inband instead of 2833 event, what is the configuration for this? Many thanks for help, /Kaiduan

2 4

Anybody want to go in on some Spanish GM voices recordings?
by Chad Phillips 01 Feb '19

01 Feb '19

My company is just getting ready to do a round of Spanish voice prompts through GM voices. They have excellent talent, and charge reasonable rates per word/phrase, but their setup fees are a bit on the steep side. Anyone else with this same need want to join in our order to defer some of the setup costs? I imagine we can pay the per word/phrase costs individually, and split the setup fee. Also, FreeSWITCH core team: are you interested in having some Spanish prompts done? My company would cover the cost of some, if there weren't too many, as a service to the community... Chad

2 1

Debian 9 FreeSWITCH 1.8 Sangoma Card A102
by Alaadin Abdurrahman 01 Feb '19

01 Feb '19

Dears, I have an issue when i try to compile and make freeswitch 1.8 with mod_freetdm to use sangoma card. please note that i followed every step in the sangoma wiki https://wiki.freepbx.org/display/PC/Telephony+Cards+for+FreeSWITCH and i still get this error message cc1: all warnings being treated as errors Makefile:1209: recipe for target 'ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo' failed make[2]: *** [ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo] Error 1 make[2]: Leaving directory '/usr/src/freeswitch/libs/freetdm' Makefile:15: recipe for target '../libfreetdm.la' failed make[1]: *** [../libfreetdm.la] Error 2 make[1]: Leaving directory '/usr/src/freeswitch/libs/freetdm/mod_freetdm' ../../../build/modmake.rules:89: recipe for target 'all' failed make: *** [all] Error 1 root@debian:/usr/src/freeswitch/libs/freetdm/mod_freetdm# best regards

2 2

Sofia stops responding after a few days
by Bipin Patel 01 Feb '19

01 Feb '19

hi, I have 2 instances of FS running on a single windows box, first instance uses normal sip UDP profiles mainly used for routing calls to carriers and its running as a service since a few months without any issues. The second instance runs a sip TLS profile and accepts inbound registrations and forwards calls in sip UDP ahead, it uses xml_curl for directory users but the problem is every 2-3 days the inbound TLS profile stops responding to registrations, and when that happens i cant even stop and restart the service, have to kill it and start again. FS_CLI works but doesnt show any error, at first i though it could be the xml_curl causing the issue but later realized it never sends any requests when sofia stops responding. i have been banging my head from the past week or so but not able to find the cause, could any1 help in guiding me what to check when this happens so can find the root cause, im using commercial certs for TLS and when its running there r no issues other than sofia stops responding every few days, it happens at random times. -- Regards, Bipin ------------------------------------------------------------------------

4 9