June 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Is MySQL OK to use?
by Alex Lake 06 Apr '16

06 Apr '16

I find these "Error in my_thread_global_end()" messages somewhat annoying in my fs1.2stable on Ubuntu12.04 box. Is there any advice (other than "don't use MySQL") for how to install it better? Might it be something to do with thread-safe libraries (I'm using libmyodbc_r.so at the moment)

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FS, SS7 and Dialogic IMG2020 in the UK
by Gavin Henry 15 Mar '16

15 Mar '16

Hi all, We're looking to get a pair of Dialogic IMG2020 for some BT Ofcom regulated SS7 interconnects: http://www.dialogic.com/en/products/gateways/img/img2020.aspx Any issues known with FS behind these? Thanks, Gavin. -- http://www.surevoip.co.uk

6 7

Random calls failing with WRONG_CALL_STATe when using TLS
by Emrah 26 Jan '16

26 Jan '16

Hi all, I am facing a very frustrating issue. I often have to dial twice when using my Yealink phone with TLS because the first attempt times out. The logs on the Yealink indicate that the first invite is successfully received, to which my FS sends a 100 trying and 407 proxy auth required. It is subsequently when my phone sends back the invite that the connection crashes with the following error: SSL ERROR SYSCALL Is this something common? Why does the SSL connection crashes when the phone attempts to send the second invite? My phone is behind NAT. It is going to be a crazy expedition to collect the logs and Pastebin them, so I am tempting my luck on the list first to see if you have any pointers. As a last piece, my Bria on my iPHone, among other clients, never had this issue. I did experience it from time to time with Blink on Mac OS X. Any help appreciated. Emrah

5 12

Re: [Freeswitch-users] Event socket and Python
by Johan Brannlund 17 Jan '16

17 Jan '16

Sami Montour wrote: > Sorry I misunderstood your post about testing with a closed port. > Anyway, I ran your script (port 8041) but didn't get a connection to > FS, different from what you received. Hmm, that's interesting. I guess either I happened to check out a buggy version of FS, there's something strange with my configuration or there's some ARM-specific issue (I run FS on a Raspberry Pi). I'll have to do some more troubleshooting. Thank you for checking! - Johan

4 4

outbound proxy and port
by Chris B. Ware 05 Jan '16

05 Jan '16

Hi, I need to send registrations and calls from external profile to a si provider, passing through an outbound proxy, port 5080. I I set on sofia: <param name="outbound-proxy" value="10.24.4.151"/> Registrations pass through the OB Proxy, but using, of course, standard 5060 port. I I set on sofia: <param name="outbound-proxy" value="10.24.4.151:5080"/> Param seems to be not understood and registrations go directly to the sip provider. How can I set outbound proxy port? Thanks

7 7

How to enable Screen sharing on Verto Conf?
by Victor Medina 23 Nov '15

23 Nov '15

Hi guys! I been playing around with the verto conference demo on 1.6. How do I enable the screen sharing feature? -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys

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FreeSWITCH Unstable Repo
by Brian West 03 Oct '15

03 Oct '15

FreeSWITCHers, To help speed the testing and release cycle up, Team FreeSWITCH has created a FreeSWITCH Unstable Debian repo, This repo is updated after every commit to master and the debs lag behind about an hour from each commit. WARNING this repo is untested and should NOT be used in production. How to use this repo: 1. New Debian Jessie system 2. Run These Commands: echo "deb http://files.freeswitch.org/repo/deb/debian-unstable/ jessie main" > /etc/apt/sources.list.d/99FreeSWITCH.list echo "deb http://files.freeswitch.org/repo/deb/debian/ jessie main" >> /etc/apt/sources.list.d/99FreeSWITCH.list wget -O - https://files.freeswitch.org/repo/deb/debian/key.gpg |apt-key add - apt-get update apt-get install freeswitch-all 3. To update: apt-get update apt-get dist-upgrade Any commit that goes in while the current build is running will be included in the next build that starts. Questions? Mailing List / IRC / Hipchat Comments? Mailing List / IRC / Hipchat Bugs? https://freeswitch.org/jira/ Thanks, and happy testing! -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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INVITE and hangup in the middle of a call
by Oleg Stolyar 27 Aug '15

27 Aug '15

Hi guys, Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm. The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light. send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: ------------------------------------------------------------------------ INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid >;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825 v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0 ------------------------------------------------------------------------ fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous(a)anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous(a)anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous(a)anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous(a)anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous(a)anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous(a)anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous(a)anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous(a)anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous(a)anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous(a)anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous(a)anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous(a)anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous(a)anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous(a)anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous(a)anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous(a)anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous(a)anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous(a)anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous(a)anonymous.invalid) State HANGUP going to sleep

7 23

freeswitch memory leakage.
by Aqs Younas 24 Aug '15

24 Aug '15

Hi, users We are using 6 freeswitch instances, 5 freeswitch instances for playing streams with mod_vlc. After 3 to 4 for days, we see these (5) freeswitch taking more than 3gb of momory and even though calls are not more than 25. Usually, some calls stay for more than 2 to 3 hours on some freeswitch. Everytime we have to restart the freeswitch to release the captured memory. But is believed, freeswitch must release the momory when there are no calls. But freeswitch still keeps captured memory. I have attactted the top command result along with this email. Willing to perform any test if it helps tackle the problem. Thanks for your help.

6 13

Crypto Problems with NDLB-allow-crypto-in-avp
by Nicola von Thadden 22 Aug '15

22 Aug '15

Hi, I have a connection to a provider which sadly requires NDLB-allow-crypto-in-avp=true to funktion. The setup is like this: FS -- HG -- POTS | GS FS is my freeswitch server (FreeSWITCH Version 1.4.19-10-1~64bit (-10-1 64bit) on debian, from the repos). HG is the SIP server from eventphone (some might know them from the CCC events in Germany). They provide inbound DDI. I also have a phone GS (Grandstream GXP2160) registered to HG. Freeswitch is set to bridge incoming calls to my number to 3200. When calling from GS to FS, everything works. HG sends an RTP/SAVP in the INVITE and the call is working fine: INVITE sip:gw+eventphone-5849@hg:5080;transport=udp;gw=eventphone-5849 SIP/2.0 Max-Forwards: 19 Via: SIP/2.0/UDP 92.222.104.42:5060;rport;branch=z9hG4bK2140280193 From: "nicoduck" <sip:3008@fs>;tag=218198923 To: <sip:gw+eventphone-5849@fs:5080;transport=udp;gw=eventphone-5849> Call-ID: 1962077623(a)92.222.104.42 CSeq: 4570 INVITE User-Agent: EVENTPHONE PBX TrollEdition v1337 - Problem? Contact: <sip:3008@hg:5060> Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, REGISTER, REFER, OPTIONS, INFO Privacy: none Content-Type: application/sdp Content-Length: 317 v=0 o=yate 1434143858 1434143858 IN IP4 hg s=SIP Call c=IN IP4 hg t=0 0 m=audio 18910 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:917UNiViuv6i4TB08AS1XI6nzAH9OPKQupyne0kK a=encryption:optional And freeswitch starts the call with: (The 100 and 183 are stripped here, I don't think that there is something important in them) SIP/2.0 200 OK Via: SIP/2.0/UDP 92.222.104.42:5060;rport=5060;branch=z9hG4bK2140280193 From: "nicoduck" <sip:3008@hg>;tag=218198923 To: <sip:gw+eventphone-5849@fs:5080;transport=udp;gw=eventphone-5849>;tag=81UHaFemF50FN Call-ID: 1962077623@hg CSeq: 4570 INVITE Contact: <sip:gw+eventphone-5849@fs:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.19-10-1~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 307 Remote-Party-ID: "5849" <sip:5849@fs>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1434111633 1434111634 IN IP4 fs s=FreeSWITCH c=IN IP4 fs t=0 0 m=audio 32224 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:YyL9+W/F2EHV9BD5afj3o6T61X7Z116BSIT+Zlzd FS initiates the crypto and also provides a port for the audio. Audio is working in both directions. The problem starts when I get an incoming POTS call trhough HG. I don't know why but their yate sends a=crypto in a RTP/AVP package. I tried to discuss that problem with the yate developers but they did not seem to be interested in that at all and also don't think that they do something wrong. I added NDLB-allow-crypto-in-avp to my public profile to make freeswitch accept the call. But freeswitch somewhow messes up the crypto when trying to establish a connection: INVITE sip:gw+eventphone-5849@fs:5080;transport=udp;gw=eventphone-5849 SIP/2.0 Max-Forwards: 19 Via: SIP/2.0/UDP hg:5060;rport;branch=z9hG4bK715484803 From: "004917xxxx" <sip:004917xxxx@hg>;tag=776909967 To: <sip:gw+eventphone-5849@fs:5080;transport=udp;gw=eventphone-5849> Call-ID: 555992785@hg CSeq: 4550 INVITE User-Agent: EVENTPHONE PBX TrollEdition v1337 - Problem? Contact: <sip:004917xxxx@hg:5060> Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, REGISTER, REFER, OPTIONS, INFO Content-Type: application/sdp Content-Length: 316 v=0 o=yate 1434139910 1434139910 IN IP4 hg s=SIP Call c=IN IP4 hg t=0 0 m=audio 17678 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:KyzcjFG0m80I8CjQzjQGQCTctNyJo7NZgAyBPoW8 a=encryption:optional The Invite looks okay, besides the a=encryption in the RTP/AVP. This is the answer from the freeswitch: SIP/2.0 200 OK Via: SIP/2.0/UDP hg:5060;rport=5060;branch=z9hG4bK715484803 From: "004917xxxx" <sip:004917xxxx@hg>;tag=776909967 To: <sip:gw+eventphone-5849@fs:5080;transport=udp;gw=eventphone-5849>;tag=FgQ1HF20BNrQp Call-ID: 555992785@hg CSeq: 4550 INVITE Contact: <sip:gw+eventphone-5849@fs:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.19-10-1~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 129 Remote-Party-ID: "5849" <sip:5849@fs>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1434120229 1434120230 IN IP4 fs s=FreeSWITCH c=IN IP4 fs t=0 0 m=audio 0 RTP/AVP 19 Freeswitch has a problem establishing the autio connection. It does not send a proper SDP packet back to HG. It does not include a port to send audio to (this is the 0 after m=audio) and also no protocol. Wireshark translates the 19 after RTP/AVP to"Media Format: Comfort noise (old)". The only way to make a proper call between those two is to forbid crypto for incoming POTS calls via HG, which is also not quite good. I think is is a bug in freeswitch, paired with improper behaviour of HGs yate. Normal deskphones seem to work properly when registered to HG and receiving an inbound POTS call, at least the grandstream and snom I have access to. Eventphone does not seem to get complaints from other people registering with them (although most of them use phones and not a PBX). Does anyone know how to debug this further? Thanks Nico

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