December 2018 - 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

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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

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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

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INCOMPATIBLE_DESTINATION
by Paul Muaddib 13 Feb '19

13 Feb '19

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]

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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

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 ------------------------------------------------------------------------

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zRTP + OPUS one-way audio issue
by Andrea Mazzeo 30 Jan '19

30 Jan '19

Hi, I'm facing an one-way audio issue with zRTP while using OPUS. zRTP is set in passthru mode. Called party can hear Calling party but not vice versa. Full log: https://pastebin.freeswitch.org/view/639adfe9 This issue is happening *only* with dynamic payloads codecs. Everything is working fine if I use G722 or G729. Scenario Leg-A (207) -> FS -> Leg-B (213) Checking FS logs, I see Leg-A's SDP: a=rtpmap:103 opus/48000/2 Leg-B's SDP: a=rtpmap:102 opus/48000/2 Actually having different payloads should not be an issue. I opened a case to Acrobits, they said the issue is in the actual RTP traffic sent to Leg-A from FS: >From the client log Leg-A Sending RTP packet #200 192.168.128.158:59176 > 31.102.111.134:28560, len=116, really=116, data=80679E55DF67E34A291167D899830CC03C124F353DA3C069574410EC0333125F This is using 103, correct as agreed on this side of the call Received RTP packet #200 31.102.111.134:28560 > 192.168.128.158:59176, len=90, data=80663D3B22CC8CB414490864A230F49B6EFF42E7BCA9EB64B864374B1377965C This is using 102, which is wrong for this side of the call. Seems that after negotiated payload 102 with Leg-B, FS is trying to use it on Leg-A, where it should be used 102 instead. Any idea? FreeSWITCH Version 1.8.1-2-4f54cff~64bit (-2-4f54cff 64bit) OS.: Debian 8.11 Linux pbx-186 3.16.0-4-amd64 #1 SMP Debian 3.16.51-3 (2017-12-13) x86_64 GNU/Linux Thank you, Andrea Mazzeo

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Shared presence not working with callcenter application of Freeswitch
by sagar malam 18 Jan '19

18 Jan '19

I am facing issue related to presence/ shared presence for agents in callcenter(mod_callcenter). Suppose we are dialling a user 1001(a)example.com and it is a shared user in two phones. Shared presence works perfectly if we directly dial extension 1001. But if same extension is configured as agent of callcenter and call is bridged to agent(extension) through callcenter, shared presence does not work. Freeswitch does not generate notify for agent state change. Is this expected behaviour ? If not then what can fix this ? Thanks in advance

3 6

3CX-Sip-Trunk / Freeswitch cannot reach 3CX
by Thorsten Göllner 08 Jan '19

08 Jan '19

Hi, has anybody experience with 3CX-SIP-Trunks? I have installed Freeswitch and added an extension (trunk-1). On 3CX-Side I added a SIP-Trunk (generic sip provider) and I can see a succesfull registration. Now I make a call to Freeswitch and want to pass it to 3CX-Sip-Trunk. But when bridging to user/trunk-1 I get a "Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]" error (with no delay). 3CX is installed as a VM in a Azure-Cloud behind NAT. I disabled local iptables and the Firewall-Test on 3CX is all fine. Freeswitch is on bare metal with a public ip. When making a call, wireshark on 3CX-machine shows the INVITE followed by "Destination unreachable (Port unreachable)". But I am not sure, if this is a firewall-issue or something else. Any idea? Thanks in advance, Thorsten

3 3

Presence not working on resubscribe
by mittali jangid 07 Jan '19

07 Jan '19

Hi, I am working on presence in polycom phones. I have BLF key for fifo-orbits. Presence is working fine when call is parked in orbit. But if phone has lower subscription-expiry and phone resubscribes, then freeswitch responds with "Terminated" state notify. Same is happening for user-directory extension number set as BLF in polycom phones. Also if i add BLF for either fifo-orbit or extension when any call is parked or extension is attending call respectively, presence is not working as per expectation. I verified this in latest freeswitch version 1.8.2 and also Master pull. I would like to know why it is behaving this way. Below link also faced similar issue, but it is still not resolved : http://lists.freeswitch.org/pipermail/freeswitch-users/2013-October/100598.… Your help is much appreciated. Let me know what information is required from my end. Thanks and regards, Mittali Jangid

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