July 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to test a Pull Request! aka how to help us out!
by Brian West 02 Oct '15

02 Oct '15

Checkout freeswitch master first! git clone https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch.git cd freeswitch.git git config --add remote.origin.fetch '+refs/pull-requests/*/from:refs/remotes/origin/pr/*' git config alias.pr '!f() { git checkout pr/$1; git config branch.pr/$1.merge refs/pull-requests/$1/from;} ; f' git pr <PR_NUMBER> build, test, report back. You can view the list of pull requests here: https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests Questions? Comments? Thanks, -- *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

3 4

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

1 1

Fork( ) and Exec ( ) functions
by Кирилл Преслер 15 Aug '15

15 Aug '15

Abdul, have you any success? -- С уважением, Преслер Кирилл Сергеевич Инженер ООО «Мегафит» e-mail: 112(a)megafit.su тел: +7 (383) 311-09-09 (доп.112)

3 2

Virtual DB FreeSWITCH
by Usama Zaidi 14 Aug '15

14 Aug '15

So I found the solution, using HAProxy would be a viable fix. Regards!

2 1

ESL Freeswitch cluster
by NH45 CMI 11 Aug '15

11 Aug '15

Hi Guys, I am using opensips for load balancing and freeswitch for register,IVR,Callcenter ,Using mod_xml_curl for realtime ivr and dialplan but callcenter i have problem because it load only once so i use ESL to add and remove agent ,if i use one FS it's ok but more then one FS how can i add or remove agent to queue all FS using ESL Regard's NH45 CMI

8 13

Display live calls in web pages
by John Nash 10 Aug '15

10 Aug '15

Is there any way of storing "live" calls (Ongoing calls) in DB to a specific table? where we can fetch these records and display in web page? I planned to store calls using some lua script hooks but cli command show calls already shows calls which we can use?

6 9

how to extract messages(SIP) from large pcap file.
by Aqs Younas 05 Aug '15

05 Aug '15

Hi, I know this not a relevant forum for this type of question but hope some of you guys could help me with some pointers. I have a large pcap(dump) file with calls of multiple clients having different IPs. I want to extract messages based on different IPs and dump into separate dump file. How could I achieve this? Any help would be much appreciated. Thanks.

4 5

ZRTP Mandatory for Calls
by Ollie Owens 05 Aug '15

05 Aug '15

Hello, I'm currently struggling to find a way to force Freeswitch to only allow ZRTP based calls without a trusted MiTM. I've tried to use "is_zrtp_secure" from the features.xml to hang the call up with the Dialplan but it isn't getting me anywhere. I've also looked at using LUA scripts to detect when the call has been answered and then check on the variables set in switch_channel.c but this also doesn't work. I can see from switch_channel.c that it prints "ZRTP not negotiated on both sides; disabling ZRTP passthru mode." and sets zrtp_passthru_active to false when this happens. Can someone point me in the right direction for looking up the value of this var once the call has been answered to determine whether ZRTP support is available or if there is a better way to approach this? Cheers, Ollie

2 1