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SDP with RTP SAVPF is rejected on some SIP phones
by John Smolka 12 May '16

12 May '16

Hello All, I am working with webrtc and I am having problems with incoming audio. I use : - Kamailio as a SIP and Websocket proxy - Freeswitch as a media server. UA/WebRTC <--> Kamailio <--> Freeswitch For outgoing calls (webrtc->freeswitch) everything works well, but for incoming call(freeswitch->webrtc), my webrtc client was complaining about missing ICE candidates. In my dialplan I added a new line with: <action application="export" data="media_webrtc=true" /> since now, RTP/AVP and RTP/SAVP was changed to RTP/SAVPF Now, I can receive calls on my webrtc client but it does not work on some sip phones. On sip phones it is rejected with 488 Not Acceptable Here SDP from freeswitch's INVITE looks like : v=0. o=My-SBC 1462416600 1462416601 IN IP4 MyPublicIP. s=My-SBC. c=IN IP4 MyPublicIP. t=0 0. a=msid-semantic: WMS 21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf. m=audio 32204 RTP/SAVPF 8 0 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fingerprint:sha-256 B8:7C:42:58:26:FE:30:27:DB:F5: 71:21:11:96:AF:0B:DD:6E:79:9A:AC:FF:86:69:89:14:88:70:AA:EE:4A:EB. a=setup:actpass. a=rtcp-mux. a=rtcp:32204 IN IP4 MyPublicIP. a=ssrc:590458548 cname:GLr7dIFWlcfql4sH. a=ssrc:590458548 msid:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf a0. a=ssrc:590458548 mslabel:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf. a=ssrc:590458548 label:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmfa0. a=ice-ufrag:akVRJFgs6QHjvUo2. a=ice-pwd:vG0zewmpyrGKmZhm924Jxa1u. a=candidate:9311553015 1 udp 659136 MyPublicIP 32204 typ host generation 0. a=candidate:9311553015 2 udp 659136 MyPublicIP 32204 typ host generation 0. a=ptime:20. Maybe I don't understand this correctly, but is that possible to add SAVPF, AVP and SAVP into a INVITES's SDP so I can remove them on kamailio based on transport towards client (TLS,UDP,TCP,WSS) ? I have tested it on the latest freeswitch 1.4 and 1.6 versions.

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Call disconnection logic question
by John Nash 12 May '16

12 May '16

A ----------->Freeswitch --------------->B If A party sends BYE and Freeswitch forwards BYE to B party (via bridge) the end_timestamp value for B leg (Freeswitch to B party) will be ... - The moment when BYE is sent by Freeswitch to B or - When 200 OK received for BYE or timeout Is there any variable to control this behavior?

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INFO with picture_fast_update - How often?
by Dmitry Mordovin 12 May '16

12 May '16

Hello I make audio call to PSTN number thru my SIP provider. FreeSWITCH send INFO message with picture_fast_update (and receive OK) every second during call established. Every second - Is it normal behavior? Can I set another interval? FreeSWITCH 1.6.7 Dmitry

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FreeSWITCH Erroneously Reconnects Calls
by kevintestfire 11 May '16

11 May '16

Hello, We have a mobile app that allows users to place video conference calls without a user-facing concept of accounts. A user is able to press a button, and then they are sent a generated username, password, and conference number to send up to our FreeSWITCH server to connect their calls. This setup works fine for us under normal circumstances but fails in a puzzling way under the following circumstances: 1: User starts the mobile app 2: User places a call (Call A) 3: User force kills the app 4: User re-opens the app 5: User places a second call (Call B) Under this scenario, the video is directed correctly to Call B, but the audio is erroneously redirected to Call A which is still alive because the mobile app terminated without sending a BYE request. Additionally, when using a network monitor on the mobile device, we noticed that there are two open connections to FreeSWITCH, one on the local port we expect, and a second incoming connection on the lowest RTP port used by our mobile VoIP library. This only happens on these bad calls, and does not occur under normal circumstances. We considered the possibility that the VoIP library we've been using on the mobile side could be the problem, but this issue persists even if we reinstall the application or reset the phone. Changing networks from our office Wifi network to LTE between Call A and Call B usually fixes the problem for us, so we suspect that FreeSWITCH has some internal logic to reconnect a caller to a call by IP address, ignoring the fact that when the device reconnects, its username, password, and conference number are different than the call FreeSWITCH is trying to reconnect it to. Of course, that's speculation on our part. We've combed through the FreeSWITCH logs and can't find anything unusual on call B that suggests any sort of redirection of audio. We would appreciate any advice you could give us. - Kevin -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-Erroneously-Reconn… Sent from the freeswitch-users mailing list archive at Nabble.com.

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make a call between two sip phones using fs_cli ?
by amani mansour 11 May '16

11 May '16

Hi all , Can i make a call between 2 sip phones from fs_cli interface ? if yes ,and it is using originate haw can i do it please ? 2 extensions 1000 and 1001 . Best regards Amani

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ARM Repo
by Darren Williams 11 May '16

11 May '16

I have some ARM packages built for our own use but a couple of people have expressed an interest so I thought I would share the repo with the wider community. Obviously, not endorsed or approved by the FreeSWITCH team in any way: http://repo.sip247.com/debian/ Regards Darren Williams

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fail2ban does not apply ban, fail2ban-regex works
by Mimiko 11 May '16

11 May '16

Hello. I have this config for freeswitch: jail.local: [ssh] enabled = true port = ssh filter = sshd logpath = /var/log/auth.log maxretry = 5 [freeswitch] enabled = true port = 5060,5061,5080,5081 ignoreip = 127.0.0.1/8 10.10.0.0/16 filter = freeswitch logpath = /var/log/freeswitch/freeswitch.log maxretry = 1 findtime = 600 bantime = 60 action = iptables-ban freeswitch.conf: [Definition] failregex = ^[-: \.\d]+ \[WARNING\] sofia_reg\.c:\d+ Can't find user \[\d+@[^\]]+\] from <HOST>$ ignoreregex = When running fail2ban-regex /var/log/freeswitch/freeswitch.log /etc/fail2ban/filter.d/freeswitch.conf There are matches with failed users: /usr/share/fail2ban/server/filter.py:442: DeprecationWarning: the md5 module is deprecated; use hashlib instead import md5 Running tests ============= Use regex file : /etc/fail2ban/filter.d/freeswitch.conf Use log file : /var/log/freeswitch/freeswitch.log Results ======= Failregex |- Regular expressions: | [1] ^[-: \.\d]+ \[WARNING\] sofia_reg\.c:\d+ Can't find user \[\d+@[^\]]+\] from <HOST>$ | `- Number of matches: [1] 18 match(es) Ignoreregex |- Regular expressions: | `- Number of matches: Summary ======= Addresses found: [1] 163.172.194.73 (Wed May 11 10:16:54 2016) 163.172.194.73 (Wed May 11 10:18:02 2016) 163.172.194.73 (Wed May 11 10:18:30 2016) 163.172.194.73 (Wed May 11 10:18:40 2016) 163.172.194.73 (Wed May 11 10:19:10 2016) 163.172.194.73 (Wed May 11 10:19:29 2016) 163.172.194.73 (Wed May 11 10:20:10 2016) 163.172.194.73 (Wed May 11 10:20:30 2016) 163.172.194.73 (Wed May 11 10:20:37 2016) 163.172.194.73 (Wed May 11 10:21:42 2016) 163.172.194.73 (Wed May 11 10:21:53 2016) 163.172.194.73 (Wed May 11 10:22:00 2016) 163.172.194.73 (Wed May 11 10:22:50 2016) 163.172.194.73 (Wed May 11 10:23:13 2016) 163.172.194.73 (Wed May 11 10:23:28 2016) 163.172.194.73 (Wed May 11 10:23:30 2016) 163.172.194.73 (Wed May 11 10:25:07 2016) 163.172.194.73 (Wed May 11 10:25:29 2016) Date template hits: 0 hit(s): MONTH Day Hour:Minute:Second 0 hit(s): WEEKDAY MONTH Day Hour:Minute:Second Year 0 hit(s): WEEKDAY MONTH Day Hour:Minute:Second 0 hit(s): Year/Month/Day Hour:Minute:Second 0 hit(s): Day/Month/Year Hour:Minute:Second 0 hit(s): Day/Month/Year Hour:Minute:Second 0 hit(s): Day/MONTH/Year:Hour:Minute:Second 0 hit(s): Month/Day/Year:Hour:Minute:Second 5196 hit(s): Year-Month-Day Hour:Minute:Second 0 hit(s): Day-MONTH-Year Hour:Minute:Second[.Millisecond] 0 hit(s): Day-Month-Year Hour:Minute:Second 0 hit(s): TAI64N 0 hit(s): Epoch 0 hit(s): ISO 8601 0 hit(s): Hour:Minute:Second 0 hit(s): <Month/Day/Year@Hour:Minute:Second> Success, the total number of match is 18 However, look at the above section 'Running tests' which could contain important information. However, fail2ban does not ban that ip. Although when ssh login attempt is detected, fail2ban does ban the ip. I've checked the time and time zone and all ok. On server Local time is +3 from UTC. sshd log, freeswitch log and fail2ban log are in same time zone. What could be the culprit?

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external voicemail storage?
by Avi Marcus 11 May '16

11 May '16

I checked confluence and I don't see any updates... Is it possible to store voicemail messages & greetings somewhere NOT on the disk? e.g. S3, in the same database at the voicemail prefs, etc? I'm rebuilding an old cluster and this is an issue I didn't find a good solution to. It's one of the only things I'd *need *to have a distributed filesystem for and I haven't had much luck with those. Thanks, -Avi Marcus BestFone

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Starting freeswitch at boot with systemd
by William Colls 10 May '16

10 May '16

debian 8.4 64 bit freeswitch 1.6.7 built locally from source According to Debian 8 (jessie) Admin manual the preferred method of starting something at boot time is to use systemd. In the freeswitch/debian directory provided with the sources there is a freeswitch.service example provided. I have copied the provided example to /lib/systemd/system, and tweaked it to reflect my setup, and enabled it. ;;;;; Author: Travis Cross <tc(a)traviscross.com> [Unit] Description=freeswitch After=syslog.target network.target local-fs.target [Service] ; service Type=forking PIDFile=/var/run/freeswitch/freeswitch.pid EnvironmentFile=-/etc/default/freeswitch ExecStart=/usr/local/bin/freeswitch -u freeswitch -g freeswitch -ncwait -nonat TimeoutSec=45s Restart=always ; exec User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 LimitSTACK=240K LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install] WantedBy=multi-user.target ; end of freeswitch.service However, when trying to start freeswitch, it fails with the following errors. ● freeswitch.service - freeswitch Loaded: loaded (/lib/systemd/system/freeswitch.service; enabled) Active: failed (Result: start-limit) since Tue 2016-05-10 16:59:16 EDT; 12s ago Process: 1179 ExecStart=/usr/local/bin/freeswitch -u freeswitch -g freeswitch -ncwait -nonat (code=exited, status=1/FAILURE) May 10 16:59:16 freeswitch systemd[1]: freeswitch.service: control process exited, code=exited status=1 May 10 16:59:16 freeswitch systemd[1]: Failed to start freeswitch. May 10 16:59:16 freeswitch systemd[1]: Unit freeswitch.service entered failed state. May 10 16:59:16 freeswitch systemd[1]: freeswitch.service start request repeated too quickly, refusing to start. May 10 16:59:16 freeswitch systemd[1]: Failed to start freeswitch. May 10 16:59:16 freeswitch systemd[1]: Unit freeswitch.service entered failed state. I have very limited experience with systemd. I have the documentation from the Debian Admin Manual, but not much else. I did successfully add another service to systemd. Initially I had the same error message "xxx.service start request repeated too quickly, refusing to start." and traced the problem to the service file pointing to the wrong place for one of the listed files. I have verified that all of the files do exist, and have suitable permissions. Any as where I should go from here appreciated. Thanks for your time. William

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Fail to adjust PTIME with OPUS
by Luciano Augusto Rezende 10 May '16

10 May '16

Hello, I have an environment with debian 8 and freeswitch 1.6.7 installed from the fs repositories. With this env my call were working fine, using opus codec. But recently I deployed a new env, with the same configurations from the previous one but with freeswitch's new release 1.6.8, and my calls are failing after few seconds when adjusting PTIME as the following log shows: [WARNING] switch_core_media.c:2567 [VBR]: Asynchronous PTIME supported, adjusting JB size. Remote PTIME changed from [20] to [40] [DEBUG] switch_core_media.c:2988 Changing Codec from opus@20ms@48000hz to opus@40ms@48000hz [DEBUG] mod_opus.c:661 Opus decoder stats: Frames[0] PLC[0] FEC[0] [DEBUG] mod_opus.c:661 Opus decoder stats: Frames[0] PLC[0] FEC[0] [WARNING] switch_core_codec.c:721 Codec opus Exists but not at the desired implementation. 48000hz 40ms 2ch [ERR] switch_core_media.c:3025 Can't load codec? [NOTICE] switch_core_media.c:3026 Hangup sofia/<profile>/<user>@<domain> [CS_EXCHANGE_MEDIA] [INCOMPATIBLE_DESTINATION] Sometimes it successfully adjust few times before breaking, or breaks on the first try like this one. It also happens with other PTIME combinations, like [40] to [80] or [20] to [60]. Is that an issue/bug? Maybe related to https://freeswitch.org/jira/browse/FS-9062 that came with the last release. Thanks Luciano Rezende

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