- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch not loading SSL
by Tim Smith 24 Aug '14

24 Aug '14

I've installed new certs, checked config in vars, restarted freeswitch entirely, and yet it is still using some sort of internal certificates ?? cafile and agent contain my certs and not those referred to in the openssl output ?? Where is freeswitch hiding these useless default certs are they hardcoded in the source or something silly ? FreeSWITCH Version 1.4.8+git~20140821T185758Z~1fe89f530f~64bit (git 1fe89f5 2014-08-21 18:57:58Z 64bit) /usr/local/freeswitch/conf/ssl# openssl verify -CAfile cafile.pem agent.pem agent.pem: OK /usr/local/freeswitch/conf# cat vars.xml | grep ssl valid options: sslv2,sslv3,sslv23,tlsv1,tlsv1.1,tlsv1.2 <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/> $ openssl s_client -showcerts -connect my.server:5061 CONNECTED(00000003) depth=0 /C=US/CN=FreeSWITCH verify error:num=18:self signed certificate verify return:1 depth=0 /C=US/CN=FreeSWITCH verify return:1 --- Certificate chain 0 s:/C=US/CN=FreeSWITCH i:/C=US/CN=FreeSWITCH -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- --- Server certificate subject=/C=US/CN=FreeSWITCH issuer=/C=US/CN=FreeSWITCH --- No client certificate CA names sent --- SSL handshake has read 615 bytes and written 328 bytes --- New, TLSv1/SSLv3, Cipher is AES256-SHA Server public key is 1024 bit Secure Renegotiation IS supported Compression: NONE Expansion: NONE SSL-Session: Protocol : TLSv1 Cipher : AES256-SHA Session-ID: Session-ID-ctx: Master-Key: Key-Arg : None Start Time: Timeout : 300 (sec) Verify return code: 18 (self signed certificate) ---

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Re: [Freeswitch-users] WebRTC Calls via Kamailio rejecting with 488
by Luis Azedo 24 Aug '14

24 Aug '14

do you have srtp enabled globally ? in freeswitch.xml <X-PRE-PROCESS cmd="set" data="rtp_secure_media=true"/> in sip profile <param name="NDLB-allow-crypto-in-avp" value="true"/>

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Re: [Freeswitch-users] WebRTC Calls via Kamailio rejecting with 488
by Kamrul Khan 24 Aug '14

24 Aug '14

Hi, This is the continuation of my previous mail, "Freeswitch srtp rejecting SRTP with 488‏". I have changed the subject line as according to the feedback from this mailing list Im convinced its not a SRTP issue. Let me describe my scenario a bit more detail. I am generating call from a webclient using simpl5. The call then hits a Kamailio server and Kamailio here acts like a webRTC proxy that forwards the call to Freeswitch. Freeswitch receives the call on udp port 5060. (not over websocket) We tested the same thing with asterisk and thats working fine. but when we try it with Freeswitch it rejects the call with 488. And in console log it shows [CS_NEW] [INCOMPATIBLE_DESTINATION] Now, we assume (as per the feedback from this mailing list) the issue is not with SRTP but with codec negotiation. Therefore, for testing purpose we stripped off all the codec attributes from our INVITE leaving only rtpmap:0 PCMU/8000 and tested again. We were hoping to see something new in our console log this time but surprisingly it showed up the exact same log as before. The console log is given below. Please somebody give us some clue whats going wrong? Thanks in advanced. 2014-08-20 05:03:42.183856 [DEBUG] sofia.c:5847 IP 192.168.146.133 Rejected by acl "domains". Falling back to Digest auth.2014-08-20 05:03:42.253550 [DEBUG] sofia.c:5847 IP 192.168.146.133 Rejected by acl "domains". Falling back to Digest auth.2014-08-20 05:03:42.269373 [NOTICE] switch_channel.c:669 New Channel sofia/internal/alice(a)192.168.146.133 [07526f12-2862-11e4-8afd-9de9b4eb98f8]2014-08-20 05:03:42.272360 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_NEW2014-08-20 05:03:42.272360 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/alice(a)192.168.146.133) State NEW2014-08-20 05:03:42.281094 [DEBUG] sofia.c:4153 Channel sofia/internal/alice(a)192.168.146.133 entering state [received][100]2014-08-20 05:03:42.281094 [DEBUG] sofia.c:4164 Remote SDP:v=0o=Mozilla-SIPUA-31.0 20849 1 IN IP4 0.0.0.0s=Doubango Telecom - firefoxt=0 0a=ice-ufrag:ca607972a=ice-pwd:dd5932ce75801e50fae435678cd6755ba=fingerprint:sha-256 8B:B0:38:A0:DA:3D:7C:DE:7F:76:F0:35:AF:1A:DE:E9:25:8C:16:E7:D4:3F:75:B2:64:56:79:21:2D:16:7D:05m=audio 57096 UDP/RTP 0c=IN IP4 184.69.59.132a=ptime:20a=rtpmap:0 PCMU/8000a=fmtp:101 0-15a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-levela=setup:actpassa=candidate:0 1 UDP 2128609535 172.16.1.188 57096 typ hosta=candidate:1 1 UDP 1692467199 184.69.59.132 57096 typ srflx raddr 172.16.1.188 rport 57096a=candidate:5 1 UDP 2128543999 192.168.56.1 57097 typ hosta=candidate:10 1 UDP 2128478463 192.168.232.1 57098 typ hosta=candidate:15 1 UDP 2128412927 192.168.146.1 57099 typ hosta=candidate:0 2 UDP 2128609534 172.16.1.188 57100 typ hosta=candidate:1 2 UDP 1692467198 184.69.59.132 57100 typ srflx raddr 172.16.1.188 rport 57100a=candidate:5 2 UDP 2128543998 192.168.56.1 57101 typ hosta=candidate:10 2 UDP 2128478462 192.168.232.1 57102 typ hosta=candidate:15 2 UDP 2128412926 192.168.146.1 57103 typ hosta=rtcp-mux 2014-08-20 05:03:42.281094 [NOTICE] sofia.c:4354 Hangup sofia/internal/alice(a)192.168.146.133 [CS_NEW] [INCOMPATIBLE_DESTINATION]2014-08-20 05:03:42.281094 [DEBUG] switch_channel.c:2102 Send signal sofia/internal/alice(a)192.168.146.133 [KILL]2014-08-20 05:03:42.282146 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK]2014-08-20 05:03:42.282146 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_HANGUP2014-08-20 05:03:42.282146 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/alice(a)192.168.146.133) State HANGUP2014-08-20 05:03:42.282146 [DEBUG] mod_sofia.c:414 Channel sofia/internal/alice(a)192.168.146.133 hanging up, cause: INCOMPATIBLE_DESTINATION2014-08-20 05:03:42.328550 [DEBUG] mod_sofia.c:476 Responding to INVITE with: 4882014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:46 sofia/internal/alice(a)192.168.146.133 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/alice(a)192.168.146.133) State HANGUP going to sleep2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/alice(a)192.168.146.133) State Change CS_HANGUP -> CS_REPORTING2014-08-20 05:03:42.328550 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK]2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_REPORTING2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/alice(a)192.168.146.133) State REPORTING2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:53 sofia/internal/alice(a)192.168.146.133 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/alice(a)192.168.146.133) State REPORTING going to sleep2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/alice(a)192.168.146.133) State Change CS_REPORTING -> CS_DESTROY2014-08-20 05:03:42.328550 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK]2014-08-20 05:03:42.328550 [DEBUG] switch_core_session.c:1164 Session 18 (sofia/internal/alice(a)192.168.146.133) Locked, Waiting on external entities2014-08-20 05:03:42.328550 [NOTICE] switch_core_session.c:1182 Session 18 (sofia/internal/alice(a)192.168.146.133) Ended2014-08-20 05:03:42.328550 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/alice(a)192.168.146.133 [CS_DESTROY]2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_DESTROY2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/alice(a)192.168.146.133) State DESTROY2014-08-20 05:03:42.328550 [DEBUG] mod_sofia.c:341 sofia/internal/alice(a)192.168.146.133 SOFIA DESTROY2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:60 sofia/internal/alice(a)192.168.146.133 Standard DESTROY2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/alice(a)192.168.146.133) State DESTROY going to sleep From: freeswitch-users-request(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 98, Issue 138 To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 22:18:01 +0400 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --Forwarded Message Attachment-- From: andretodd(a)verizon.net To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 13:35:01 -0400 Subject: Re: [Freeswitch-users] high pdd Hi, I’m using the master. Should I try 1.4.7? I’ve done a dump after things started to go bad/ backed up. I see 8193 threads in _thread_cond_timedwait. Looks like its on line 81 of thread_cond.c res = WaitForSingleObject(cond->semaphore, timeout_ms); The value of timeout_ms is 2016850392. Not sure if this means anything. Thanks for the support guys. Andre From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, August 20, 2014 11:57 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] high pdd 1.2.X is EOL, 1.4.7 is the current release. We have releases and tags in git that no longer carry the 'stable' moniker starting with 1.4. On Sun, Aug 17, 2014 at 1:39 PM, Andre Demattia <andretodd(a)verizon.net> wrote:Thanks, I'm using the newest master and the last stable 1.2.23 We are willing to pay to get this fixed. Andre From: Anthony Minessale Sent: ‎8/‎17/‎2014 12:30 PM To: Freeswitch-users Subject: Re: [Freeswitch-users] high pddWe dont have a lot of tools to troubleshoot windows. You should mention windows in your subject to have a better chance to get the attention of our windows users. You should consider installing debian 7 on the same hardware and perform similar tests to help determine if your issue is o/s specific or not. You also have not indicated the revision of FS. If you have not worked with latest master, you should be also to rule out improvements. On Aug 17, 2014 9:54 AM, "Andre Demattia" <andretodd(a)verizon.net> wrote:Yes that's off and I believe all the databases are off too. No authentication or registration is enabled. The sizes of the database are 0 until I call the shutdown command then its about 90k. I have the DBS in Ra drive too. Windows seems pretty good until the 3000 sessions per min mark. Then I just get long pdd and the threads get backed up then no more sessions can be created. Funny thing is I has 10 profiles and the same results. I also tried two consoles with the same results. FreeSWITCH is great. I'd be grateful for help. I've worked on this for 2 years and hate to give up. If any one from consulting(a)freeswitch.org is willing to help I'd appreciate that. Thanks again, Andre From: Anthony Minessale Sent: ‎8/‎17/‎2014 10:13 AM To: Freeswitch-users Subject: Re: [Freeswitch-users] high pddDid you comment out manage-presence ? Also we have no real idea how well FS performs in windows under extreme load. On Aug 17, 2014 5:42 AM, "Andre Demattia" <andretodd(a)verizon.net> wrote:Thanks Ken, I'll try the consulting again. Andre From: Ken Rice Sent: ‎8/‎17/‎2014 1:19 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] high pddif the last post does not work contact consulting(a)freeswitch.org for pro help from one of the developers... thats about as far as the list will take youKenSent from my iPad On Aug 16, 2014, at 15:23, "Andre" <andretodd(a)verizon.net> wrote:Going with the idea it’s IO related I did a db_cache status. I just disabled limit in my code by removing it but limit should be HASH not a database however I stills how a database. Below is the results of the db cache status. I’m not real sure what its saying. I should have all registration and limit disabled and should not be using any sql database from my understanding. Anyone know what the results mean? I’ve only tested 2 calls on this status. [The entire original message is not included.] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [The entire original message is not included.] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest --Forwarded Message Attachment-- From: stuart.mills3(a)btopenworld.com To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 19:17:34 +0100 Subject: Re: [Freeswitch-users] Media after hang-up??? Hi Mike, Thanks for the reply. I know for sure this isn’t FreeSWITCH causing the issue, I get the CDR posted at the right time and durations are correct, it’s almost as if the far end PBX hasn’t noticed the hang-up, even though the BYE does get sent. My carrier mentioned today (following a full trace from their end) that the PBX isn’t sending an ACK to the BYE, does that make any sense given the issue I am seeing? My main issue here is if it’s a PBX configuration issue, I am unable to look at and diagnose myself, so some sort of starting point would be really helpful . Cheers, Stuart From: Michael Jerris Sent: Wednesday, August 20, 2014 5:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Media after hang-up??? I have never seen an issue like this, a pcap might tell you more, but i seriously doubt we are still sending media after the call hangs up. I would suggest taking a look at a pcap and the debug logs and seeing that the call is actually being hung up on the leg facing the pbx and we really stop sending media. Mike On Aug 19, 2014, at 8:19 PM, Stuart Mills <stuart.mills3(a)btopenworld.com> wrote: Hi, I’m experiencing a strange issue calling a destination within the UK, whereby media seems to still be connected after hang up. I only know this as the destination is a PBX, supplied by Unify (formerly Siemens), and that PBX on occasions records a voicemail, when a message is taken and after the call is hung up, there is a continuous tone heard on the end of the recording. The call logs show the right duration and the carrier has also confirmed that the call ended as the logs suggest. So, I’m wondering if anyone else has come across the same sort of issue using FreeSWITCH? The end user has told me they never used to experience this issue using a different provider, and I can’t reproduce this issue at all using other types of PBX, all calls end as they should and voicemail recordings end when the call does. This issue doesn’t seem to be a FreeSWITCH one, just thought it worth an ask to the group. Cheers, Stuart _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Faxing Large Files
by afshin afzali 24 Aug '14

24 Aug '14

Hi, I don't know if there is a limit on TIFF file size when being faxed! My file is about 16M in size. My error is "*41 TIFF/F file cannot be opened*". Also I checked the permission issue and think it is OK. I do not have any problem on small file size. BEST, Afshin

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Remote audio not heard
by Michael 24 Aug '14

24 Aug '14

While calling certain destination phone numbers (via freeswitch-1.4.7, and flowroute), often, on the calling devices (SPA942, SPA942, GrandStream, old phone on SPA3102 ATA), no ringing is heard and, though the destination has answered, no audio returns from the destination (although destination can hear audio from caller). The same destination numbers sometimes work correctly and sometimes fail as above. Logs show normal activity, meaning no difference between normal and fail. Exact same thing happened on freeswitch-1.2.something, 1.4.6, and now 1.4.7. Exact same thing happens across wildly different calling phones, across many different destination numbers. Just about the only thing that has not been changed, updated, etc. is flowroute as the carrier. Could it be a problem at the flowroute end? Anyone else having similar experiences? Any ideas on where to begin? TIA, Michael

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[ANN] RoCKSolidPBX 0.99 ALPHA
by Richard Mace 23 Aug '14

23 Aug '14

Hi All, Just thought I'd release one last Alpha before a Beta just in case there are any last minute silly bugs (which I'm sure there will be) Any questions, lease me know. www.rocksolidpbx.co.uk PS Please note, you'll need a fresh install of Debian 7 and a Windows client to test this software. Many thanks Richard

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FS uses private rtp address in the SDP
by afshin afzali 23 Aug '14

23 Aug '14

Hi FreeSWITCHers: I configured a sip profile as bellow: ================================================================================================= Name sipinterface_1 Domain Name N/A Auto-NAT false DBName sofia_reg_sipinterface_1 Pres Hosts Dialplan XML Context context_1 Challenge Realm auto_to RTP-IP 192.168.10.59 Ext-RTP-IP X.Y.Z.W SIP-IP 192.168.10.59 Ext-SIP-IP X.Y.Z.W URL sip:mod_sofia@X.Y.Z.W:5060 BIND-URL sip:mod_sofia@X.Y.Z.W :5060;maddr=192.168.10.59;transport=udp,tcp HOLD-MUSIC N/A OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM CODECS OUT G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT true CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1 for every INVITE that comes in via this profile, FS replys with this SDP: v=0 o=FreeSWITCH 1407905843 1407905844 IN IP4 192.168.10.59 s=FreeSWITCH c=IN IP4 192.168.10.59 t=0 0 m=audio 25142 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Although the Contact header of 200 OK is correct : Contact: <sip:4000@X.Y.Z.W:5060;transport=udp> Appreciate all comments, -- Afshin

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Set sip_from_user with channel variable?
by Bruce Lefko 23 Aug '14

23 Aug '14

My SIP provider requires me to pass the caller ID number using the SIP from header and they ignore the value in the Remote Party ID field. Is there a way to set a channel variable that will impact the SIP from header? I've tried setting sip_from_user, sip_auth_username, user_name to the caller ID, but FreeSWITCH keeps defaulting to "FreeSWITCH" as the username. My sip profile simply looks like this: <include> <gateway name="myprovider"> <param name="proxy" value="THE_IP"/> <param name="context" value="public"/> <param name="register" value="false"/> </gateway> </include> Thanks!

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mod_skypopen new documentation is ready !
by Giovanni Maruzzelli 22 Aug '14

22 Aug '14

Dear FreeSWITCHers, new documentation on how to build and operate, using latest Skype client, the Skype FreeSWITCH endpoint mod_skypopen is out in Confluence. Please find it at: https://confluence.freeswitch.org/display/FREESWITCH/mod_skypopen Let us know if there is something to clarify, add, or correct. Ciao, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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GSMopen incoming call behavior
by Mervyn Yeo 22 Aug '14

22 Aug '14

Hi all, I've gotten the mod working and while I was testing I realized that incoming calls are automatically answered by FS immediately and ringing tone is sent to the caller. Does anyone know if this a hardcoded behavior and would there be plans to make it such that FS doesn't pick up the call and ring the SIP endpoints and only pick up when the SIP endpoint picks up? Thanks, Mervyn

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