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Memory leak when using ESL
by gggyniidt 16 Nov '13

16 Nov '13

Hello Using ESL found a memory leak problem valgrind --tool=memcheck --leak-check=full ./myprog output as follows ==6872== 29,986 (88 direct, 29,898 indirect) bytes in 1 blocks are definitely lost in loss record 401 of 408 ==6872== at 0x4A06AAA: malloc (vg_replace_malloc.c:291) ==6872== by 0x413941: esl_event_create_subclass (esl_event.c:179) ==6872== by 0x410DC3: esl_recv_event (esl.c:1245) ==6872== by 0x411675: esl_send_recv_timed (esl.c:1517) ==6872== by 0x411C1B: esl_execute (esl.c:555) ==6872== by 0x408758: CRBT::Route() (CRBT.cpp:42) ==6872== by 0x40A02F: DPL::route() (DPL.cpp:234) ==6872== by 0x40FCA8: mycallback(int, int, sockaddr_in*) (listener.cpp:47) ==6872== by 0x41032F: client_thread (esl.c:654) ==6872== by 0x413FC7: thread_launch (esl_threadmutex.c:74) ==6872== by 0x34CAA07850: start_thread (in /lib64/libpthread-2.12.so) ==6872== by 0x34CA6E890C: clone (in /lib64/libc-2.12.so) ==6872== LEAK SUMMARY: ==6872== definitely lost: 88 bytes in 1 blocks ==6872== indirectly lost: 29,898 bytes in 76 blocks ==6872== possibly lost: 550,082 bytes in 51 blocks ==6872== still reachable: 832,937 bytes in 840 blocks ==6872== suppressed: 0 bytes in 0 blocks ==6872== Reachable blocks (those to which a pointer was found) are not shown. ==6872== To see them, rerun with: --leak-check=full --show-leak-kinds=all Part of my code is as follows bool CRBT::Route() { string data = "ringback=file_string://"; data.append(FileName); esl_execute(handle, "set", data.c_str(), NULL); return true; } The OS version is Centos x86_64 6.3 The Freeswitch version is 1.2.13 Thanks for your help.

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SRTP issue with latest master + (possible) DTMF change
by Privus 007 15 Nov '13

15 Nov '13

Hi, I've been using FS successfully for some years now. Recently I decided to update from 1.2.12 to latest master via git (running FS on bare metal Debian 7.0 64bit) Obviously I saved my conf directory and tried to apply it to the 1.5.7b+git~20131114 version I just installed and have up and running. I realize that master is not yet stable but I notice that there seems to be some incompatibility issues, namely with SRTP. All my SRTP calls are now failing with "incompatible destination" messages in the logs, and looking through them more closely I see this: parsing [features->is_secure] continue=true Dialplan: sofia/external/1000@mydomain Regex (PASS) [is_secure] ${sip_via_protocol}(tls) =~ /tls/ break=on-false Dialplan: sofia/external/1000@mydomain Regex (FAIL) [is_secure] ${sip_secure_media_confirmed}() =~ /^true$/ break=on-false Dialplan: sofia/external/1000@mydomain ANTI-Action eval(not_secure) 2013-11-14 22:02:22.006273 [NOTICE] switch_core_session.c:2940 Execute eval(not_secure) Notice the FAIL for sip_secure_media_confirmed. This is very strange since I'm sure that SRTP is enabled (both CSipSimple Android client and Groundwire iOS client confirm that indeed the signalling is secured via TLS and the media via SDES SRTP). A few seconds earlier in the logs, FS also sees the crypto taking place and there doesn't seem to be any problem 2013-11-14 22:02:21.986279 [INFO] switch_rtp.c:2830 Activating Audio Secure RTP SEND 2013-11-14 22:02:21.986279 [DEBUG] switch_core_sqldb.c:2354 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 So the problem seems to be with the sip_secure_media_confirmed variable. This same setup worked fine yesterday with 1.2.12, so I'm at a loss as to what changed. Any ideas? To further add some confusion, since my clients are configured for mandatory SRTP, all calls to them fail, and FS routes to VM. So far, pretty normal. Except that the VM message we usually hear sounds super slow like the voice is drunk. This is definetely not normal, but I'm not sure if it's related to the crypto issue. I don't think it's a flite issue since calling into the IVR sounds as normal as ever. Also, I notice a change in FS handling DTMF. My CSipSimple client which worked flawlessly with DTMF before now just doesn't work at all, but my Groundwire client continues to send DTMF without a problem. Has something changed in the latest master regarding DTMF? I'd appreciate any help in debugging these issues. Perhaps the new conf in latest master has different variables or options and by simply copying my old conf directory over the new one wasn't too smart after all. Thanks

3 7

(no subject)
by bill vaxevanopoulos 15 Nov '13

15 Nov '13

hello i would like to ask if there is a possible way to change change connnection headers for external sip profiles.. my provider complain with the following Your SIP Contact header must be presented like: sip:330591@my-external-ip but you have it configured as: the following is the defaulft configuration of freeswitc.sip:gw+voipdito@my-external-ip this is wrong. is there any way to format that contact headers in freeswitch? thank you in advanced.

3 2

Memory Leak Windows + Mod_Managed
by Andre 15 Nov '13

15 Nov '13

Hi, I'm using Windows and Mod_Managed (freeswitch-1.2.14 x64). After about 4 min of real production traffic at 30-40 CPS the memory goes from 50mg to 3GB. Freeswitch stops responding about 1.6gb. I've checked with Ants memory profiler and the .Net code is only using about 8mb. In testing I send back a 503 and never hit bridge and things appear normal and the memory returns back to normal. On production I hit the bridge application and this is where the problem is. I've removed Limit and any AppPlugin I've created. I'm not sure what to do now. :( Is this a freeswitch memory issue, mod_managed? The only modules I have running are <load module="mod_console"/> <load module="mod_enum"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_commands"/> <load module="mod_db"/> <load module="mod_dptools"/> <load module="mod_hash"/> <load module="mod_dialplan_xml"/> <load module="mod_managed"/> <load module="mod_speex"/> <load module="mod_h26x"/>

1 0

Friday Free For All and BUG Review!
by Ken Rice 15 Nov '13

15 Nov '13

Hey Guys Joinus on the FreeForAll sip:888@conference.freeswitch.org or fs0.us/call888 for other access methods Today, join us with your jiras! We¹re reviewing them live! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

1 0

Sending messages to the list?
by Don Ireland 15 Nov '13

15 Nov '13

I sent a message to the list last night after confirming my subscription. But I haven't seen the message come through my inbox. I just wanted to test and be sure I can actually send to the list. Thanks. Don Ireland Sent from my Android device with K-9 Mail. Please excuse my brevity.

3 2

Cannot queue any more events.....
by Andre 15 Nov '13

15 Nov '13

2 6

Static and Scratchy Audio
by James Mortensen 15 Nov '13

15 Nov '13

Hello, I have two FreeSWITCH 1.4b servers running on Amazon EC2. Starting yesterday, the call quality suddenly got so bad that callers from the PSTN can't hear the other person talking. The setup is Chrome WebRTC -> FreeSWITCH -> PSTN (Level 3/Bandwidth.com) We've cleared all users off the server and are trying to find out the cause. We're not dropping packets on the server or the client, yet there's problems with the audio. The audio issues occur with both servers, regardless of how many active calls there are. We don't have a problem with Chrome to Chrome peer to peer calls, so it seems like it's a problem inside FreeSWITCH. Here's what we've tried: - Checking for dropped packets - Restarting FreeSWITCH - Moving all users to another server so we test only with 1 active call. - Trying calls with both Level 3 as well as Bandwidth.com trunk providers. Does anyone have some ideas on what we can try next? Thank you! James

3 7

Re: [Freeswitch-users] Issue with audio file playback in early media
by Samir Doshi 15 Nov '13

15 Nov '13

It works. :) Thanks ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-audio-file-playbac… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

New Freeswitch User, Installation Issues.....
by Don Ireland 15 Nov '13

15 Nov '13

I followed the guide at the following link for installing Freeswitch on my Raspberry Pi. http://raspberrypipbx.blogspot.com/2012/12/using-google-voice-as-land-line.… But when I get to the point of actually starting Freeswitch (manually -- still not doing it upon boot up), it just hangs. Ironically it hangs at a line that says it's "Adding Application 'AvoidingDeadlock'". It's been like this for about an hour now. I've rebooted and restarted Freeswitch twice. What can I do? -- Don Ireland

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