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High CPS performance meltdown
by Victor Chukalovskiy 23 Jun '14

23 Jun '14

Good day, I'm running a few stress-testing scenarios with SIPp. On FS side call is dipped for LRN via SIP redirect, LRN saved / checked with mod_memcache, and routing done by mod_lcr with odbc connection to pgsql. Each call is ended by FS as soon as all steps above are complete, so no media is involved and calls are never answered. What I observe is that it can run at a very high CPS initially for a minute or so (100-200 CPS range). After that, it will drop to 50 CPS and will stay there. Suppose I stop all test calls and resume in a few minutes. It will not go back to initial 100-200 CPS values, it will stay at the 50 CPS. Whatever the bottleneck is, it clears with full FS restart. It does not clear with individual components restart. So far I've tried VM and real hardware and don't see much change with the bottleneck above. The initial CPS may differ (real HW better than VM). But, the value of 45-50CPS where it get's stuck is still the same. Tried CentOS 6.5 vs Ubuntu 14.04 no difference. Tried disabling different dial plan steps one-by-one, same pattern. A couple words on the testing scenario: say, sipp is limited to 200 CPS and 200 concurrent calls. So, until the bottle-neck condition occurs, it runs at full 200 CPS and some 10-20 concurrent calls. As soon as condition occurs, CPS drop to 40-50 and concurrent calls rise to max value of 200...which is pretty much a feeding buffer for FS going forth. Would such bottle-neck be expected from FS and what would be the steps to isolate it? Thank you, -Victor

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Re: [Freeswitch-users] Why a call hangs up without warning during recording
by Clive Lansink 23 Jun '14

23 Jun '14

Hello I haven't yet enabled logging at debug level but I did turn on SIP tracing. I found there was no reason for the hangup other than receiving the bye packet from the external public phone system. So Freeswitch just treated it as a normal hangup with "normal clearing" I think it says. But when I realised that the hangup was caused by something external, that was what led me to think it might have something to do with getting no RTP audio packets from Freeswitch, because in all my experiments this seemed to happen around the five minute mark and only while recording. I think the suggestion to set record_waste_resources=true is likely to be the answer. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Michael Jerris <mike(a)jerris.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Why a call hangs up without warning during recording Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Mon, 23 Jun 2014 12:06:32 -0400 do you have a freeswitch debug log of when the call is hung up? On Jun 21, 2014, at 9:09 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > Hello. > > I've noticed in a Freeswitch IVR application I've been working on that when the customer phones in to record a message, the call is hung up without warning after about five minutes. This does not happen when I call in locally. When this happens, the log just shows there was a hangup with normal clearing. But the customer didn't hang up. The customer just notices visually that their phone hangs up so the hangup seems to be caused somewhere between the two systems. > > I figured that during a recording, Freeswitch is not generating any audio at all, and it could be that something in the public phone system times out after absolutely no audio for a period of time. I have come up with a work around by getting Freeswitch to generate a short tone repeatedly during recording, like 50Hz at -30dB so it is basically inaudible, every thirty seconds or so. This seems to have solved the problem and now the customer can record for as long as he wants. > > But I suspect this is really a work-around and not the recommended approach. I did play with the suppress cng parameter in the sofia configuration but didn't find any setting that would solve the problem. > > Thankfully my work-around has got me home in the meantime but I would appreciate any comments on this. Does nobody make long recordings these days? > > Cheers. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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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choppy sound problem
by Babak Yakhchali 23 Jun '14

23 Jun '14

hi I'm facing choppy sound and I think the reason is virtualization and resource allocation problems (network is ok and there is no delay or packet loss) but I'm not sure. Are these timer tests ok? thanks freeswitch@192.168.54.242@internal> timer_test Avg: 21.184ms Total Time: 1059.241ms 2014-06-21 13:59:42.726513 [CONSOLE] mod_commands.c:827 Timer Test: 1 sleep 20 38614 2014-06-21 13:59:42.726513 [CONSOLE] mod_commands.c:827 Timer Test: 2 sleep 20 9493 2014-06-21 13:59:42.766482 [CONSOLE] mod_commands.c:827 Timer Test: 3 sleep 20 43439 2014-06-21 13:59:42.766482 [CONSOLE] mod_commands.c:827 Timer Test: 4 sleep 20 1096 2014-06-21 13:59:42.785982 [CONSOLE] mod_commands.c:827 Timer Test: 5 sleep 20 6333 2014-06-21 13:59:42.805985 [CONSOLE] mod_commands.c:827 Timer Test: 6 sleep 20 19624 2014-06-21 13:59:42.825978 [CONSOLE] mod_commands.c:827 Timer Test: 7 sleep 20 19799 2014-06-21 13:59:42.846009 [CONSOLE] mod_commands.c:827 Timer Test: 8 sleep 20 19863 2014-06-21 13:59:42.866178 [CONSOLE] mod_commands.c:827 Timer Test: 9 sleep 20 20633 2014-06-21 13:59:42.886020 [CONSOLE] mod_commands.c:827 Timer Test: 10 sleep 20 19772 2014-06-21 13:59:42.905975 [CONSOLE] mod_commands.c:827 Timer Test: 11 sleep 20 21133 2014-06-21 13:59:42.925983 [CONSOLE] mod_commands.c:827 Timer Test: 12 sleep 20 18429 2014-06-21 13:59:42.945990 [CONSOLE] mod_commands.c:827 Timer Test: 13 sleep 20 20090 2014-06-21 13:59:42.966008 [CONSOLE] mod_commands.c:827 Timer Test: 14 sleep 20 20028 2014-06-21 13:59:42.985985 [CONSOLE] mod_commands.c:827 Timer Test: 15 sleep 20 20013 2014-06-21 13:59:43.005997 [CONSOLE] mod_commands.c:827 Timer Test: 16 sleep 20 19880 2014-06-21 13:59:43.025982 [CONSOLE] mod_commands.c:827 Timer Test: 17 sleep 20 20150 2014-06-21 13:59:43.066246 [CONSOLE] mod_commands.c:827 Timer Test: 18 sleep 20 45247 2014-06-21 13:59:43.125991 [CONSOLE] mod_commands.c:827 Timer Test: 19 sleep 20 58078 2014-06-21 13:59:43.145988 [CONSOLE] mod_commands.c:827 Timer Test: 20 sleep 20 18716 2014-06-21 13:59:43.165991 [CONSOLE] mod_commands.c:827 Timer Test: 21 sleep 20 19869 2014-06-21 13:59:43.186026 [CONSOLE] mod_commands.c:827 Timer Test: 22 sleep 20 19886 2014-06-21 13:59:43.205974 [CONSOLE] mod_commands.c:827 Timer Test: 23 sleep 20 20076 2014-06-21 13:59:43.225985 [CONSOLE] mod_commands.c:827 Timer Test: 24 sleep 20 20128 2014-06-21 13:59:43.246028 [CONSOLE] mod_commands.c:827 Timer Test: 25 sleep 20 20221 2014-06-21 13:59:43.265980 [CONSOLE] mod_commands.c:827 Timer Test: 26 sleep 20 18851 2014-06-21 13:59:43.285978 [CONSOLE] mod_commands.c:827 Timer Test: 27 sleep 20 20912 2014-06-21 13:59:43.305975 [CONSOLE] mod_commands.c:827 Timer Test: 28 sleep 20 19728 2014-06-21 13:59:43.325980 [CONSOLE] mod_commands.c:827 Timer Test: 29 sleep 20 20227 2014-06-21 13:59:43.345987 [CONSOLE] mod_commands.c:827 Timer Test: 30 sleep 20 19703 2014-06-21 13:59:43.365982 [CONSOLE] mod_commands.c:827 Timer Test: 31 sleep 20 19172 2014-06-21 13:59:43.386006 [CONSOLE] mod_commands.c:827 Timer Test: 32 sleep 20 20404 2014-06-21 13:59:43.405980 [CONSOLE] mod_commands.c:827 Timer Test: 33 sleep 20 19401 2014-06-21 13:59:43.425977 [CONSOLE] mod_commands.c:827 Timer Test: 34 sleep 20 19799 2014-06-21 13:59:43.445982 [CONSOLE] mod_commands.c:827 Timer Test: 35 sleep 20 20200 2014-06-21 13:59:43.466004 [CONSOLE] mod_commands.c:827 Timer Test: 36 sleep 20 19996 2014-06-21 13:59:43.485990 [CONSOLE] mod_commands.c:827 Timer Test: 37 sleep 20 19668 2014-06-21 13:59:43.505980 [CONSOLE] mod_commands.c:827 Timer Test: 38 sleep 20 20713 2014-06-21 13:59:43.525985 [CONSOLE] mod_commands.c:827 Timer Test: 39 sleep 20 19102 2014-06-21 13:59:43.545982 [CONSOLE] mod_commands.c:827 Timer Test: 40 sleep 20 20325 2014-06-21 13:59:43.565971 [CONSOLE] mod_commands.c:827 Timer Test: 41 sleep 20 21667 2014-06-21 13:59:43.585989 [CONSOLE] mod_commands.c:827 Timer Test: 42 sleep 20 18227 2014-06-21 13:59:43.606590 [CONSOLE] mod_commands.c:827 Timer Test: 43 sleep 20 21513 2014-06-21 13:59:43.625996 [CONSOLE] mod_commands.c:827 Timer Test: 44 sleep 20 19142 2014-06-21 13:59:43.645988 [CONSOLE] mod_commands.c:827 Timer Test: 45 sleep 20 19809 2014-06-21 13:59:43.665986 [CONSOLE] mod_commands.c:827 Timer Test: 46 sleep 20 19913 2014-06-21 13:59:43.686044 [CONSOLE] mod_commands.c:827 Timer Test: 47 sleep 20 20424 2014-06-21 13:59:43.705986 [CONSOLE] mod_commands.c:827 Timer Test: 48 sleep 20 19372 2014-06-21 13:59:43.725996 [CONSOLE] mod_commands.c:827 Timer Test: 49 sleep 20 19884 2014-06-21 13:59:43.746003 [CONSOLE] mod_commands.c:827 Timer Test: 50 sleep 20 20479 freeswitch@192.168.54.242@internal> show timers type,name,ikey timer,soft,CORE_SOFTTIMER_MODULE 1 total. freeswitch@192.168.54.242@internal> time_test 100 5 test 1 sleep 100 1575 test 2 sleep 100 764 test 3 sleep 100 222 test 4 sleep 100 224 test 5 sleep 100 238 avg 604 freeswitch@192.168.54.242@internal> show timer type,name,ikey timer,soft,CORE_SOFTTIMER_MODULE 1 total.

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Re: [Freeswitch-users] Why a call hangs up without warning during recording
by Michael Jerris 23 Jun '14

23 Jun '14

do you have a freeswitch debug log of when the call is hung up? On Jun 21, 2014, at 9:09 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > Hello. > > I've noticed in a Freeswitch IVR application I've been working on that when the customer phones in to record a message, the call is hung up without warning after about five minutes. This does not happen when I call in locally. When this happens, the log just shows there was a hangup with normal clearing. But the customer didn't hang up. The customer just notices visually that their phone hangs up so the hangup seems to be caused somewhere between the two systems. > > I figured that during a recording, Freeswitch is not generating any audio at all, and it could be that something in the public phone system times out after absolutely no audio for a period of time. I have come up with a work around by getting Freeswitch to generate a short tone repeatedly during recording, like 50Hz at -30dB so it is basically inaudible, every thirty seconds or so. This seems to have solved the problem and now the customer can record for as long as he wants. > > But I suspect this is really a work-around and not the recommended approach. I did play with the suppress cng parameter in the sofia configuration but didn't find any setting that would solve the problem. > > Thankfully my work-around has got me home in the meantime but I would appreciate any comments on this. Does nobody make long recordings these days? > > Cheers. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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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Re: [Freeswitch-users] SIP 500 overlapping requests due to race condition
by Phil Quesinberry 23 Jun '14

23 Jun '14

Digging into this further, I found the following document detailing best current practices for handling this condition. Here is a clip of the relevant section from RFC 5407 - Example Call Flows of Race Conditions in the Session Initiation Protocol (SIP) http://tools.ietf.org/html/rfc5407 Hasebe, et al. Best Current Practice [Page 16] RFC 5407 Example Call Flows of Race Conditions December 2008 3.1.4. Callee Receives re-INVITE (Established State) While in the Moratorium State (Case 1) State Alice Bob State | | | ini-INVITE w/offer1 F1 | |------------------------------->| Pre | 180 F2 | Pre |<-------------------------------| Ear | | Ear | 200(ini-INV) w/answer1 F3 | |<-------------------------------| Mora | ACK F4(packet loss) | Mora |-------------------->x | Est | | | re-INVITE F6 200 F5(=F3) | | w/offer2 w/answer1 | |------------- --------------| | \ / | | X | | / \ | |<------------ ------------->| *race* | 200(re-INV) F8| | ACK F7(=F4) w/answer2 | |------------- --------------| | \ / | | X | | / \ | |<------------ ------------->| | ACK (re-INV) F9 | Est |------------------------------->| | | | | This scenario illustrates the race condition that occurs when a UAS in the Moratorium state receives a re-INVITE sent by a UAC in the Established state. The UAS receives a re-INVITE (with offer2) before receiving an ACK for the ini-INVITE (with offer1). The UAS sends a 200 OK (with answer2) to the re-INVITE (F8) because it has sent a 200 OK (with answer1) to the ini-INVITE (F3, F5) and the dialog has already been established. (Because F5 is a retransmission of F3, SDP negotiation is not performed here.) As can be seen in Section 3.3.2 below, the 491 response seems to be closely related to session establishment, even in cases other than INVITE crossover. This example recommends that 200 be sent instead Hasebe, et al. Best Current Practice [Page 17] RFC 5407 Example Call Flows of Race Conditions December 2008 of 491 because it does not have an influence on the session. However, a 491 response can also lead to the same outcome, so either response can be used. Moreover, if the UAS doesn't receive an ACK for a long time, it should send a BYE and terminate the dialog. Note that ACK F7 has the same CSeq number as ini-INVITE F1 (see Section 13.2.2.4 of RFC 3261 [1]). The UA should not reject or drop the ACK on grounds of the CSeq number. _____________________________________________ From: Phil Quesinberry Sent: Friday, June 20, 2014 3:49 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: SIP 500 overlapping requests Correction: I meant to say "if FreeSWITCH could be configured to either wait up to x milliseconds for an ACKNOWLEDGEMENT to arrive." ___________________________________ From: Phil Quesinberry Sent: Friday, June 20, 2014 3:36 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: SIP 500 overlapping requests As it turns out, my assumption was correct. We were not receiving an acknowledge before the second invite from the carrier. One of their support engineers determined that they had recently enabled acknowledgement logging on their proxy/SBC, which apparently delayed transmission of the acknowledgement by a millisecond or two, which was enough to cause the invite from the server handling the media/call to arrive just slightly before the acknowledgement, so FreeSWITCH would tear down the call. The problem was intermittent since the acknowledgement would sometimes arrive first. They disabled acknowledgement logging and the problem went away. With this and UDP in mind, it would likely reduce the incidence of dropped calls if FreeSWITCH could be configured to either wait up to x milliseconds for an invite to arrive, or allow x number of overlapping requests before throwing an error. I saw a post where Tony mentioned that a patch had already been crafted to allow overlapping requests in the past, so it seems that most of the work has been done. __________________________________ From: Phil Quesinberry Sent: Thursday, June 19, 2014 11:08 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: SIP 500 overlappting requests We're having intermittent problems receiving calls from a particular DID. The basic scenario is as follows: Carrier sends invite w/SDP We respond w/OK and SDP when call is answered Carrier sends another invite with incremented sequence number and updated SDP (aren't they supposed to ACK/respond before doing this?) We send SIP 500 - overlapping requests Carrier sends BYE - terminating the call Sometimes (rarely) the call completes, most of the time FS sends a 500 and doesn't get the call. This just started happening recently and does not occur with any other DIDs, including other DIDs from the same carrier. I updated to the latest stable GIT to see if that might solve the problem but I'm pretty sure this is on the carrier's end. They're trying to blame it on us since we're sending the 500 and suggested that we allow overlapping requests. I'm assuming that their end needs to acknowledge our answer OK/SDP before sending their own but if that's the problem then it raises another question in my mind: since the SIP transport is generally UDP, what if we don't receive their ACK before receiving their subsequent invite/SDP? This could certainly happen and would cause the call to be rejected if the above is true. I updated to the latest stable GIT last night to rule that out as a possible cause of the problem. Is there a way to get FS to allow a limited number of overlapping requests as a possible workaround? If someone thinks this looks like it could be an FS bug, I'll file a Jira. Here is a trace which shows this happening: http://pastebin.freeswitch.org/22769 Thanks, Phil Quesinberry Q Systems Engineering, Inc. Embedded Hardware/Software Development (410) 969-8002 http://www.qsystemsengineering.com

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Cluscon Schedule is Filling Up! You don¹t want to miss this one!
by Ken Rice 23 Jun '14

23 Jun '14

ClueCon is rapidly approaching! The schedule is rapidly filling and this year will be better then ever! Join us in Chicago August 4th 7th! Here¹s just a few of the people you¹ll see! Simon Woodward, CEO of Simwood ( http://simwood.com ) and noted VoIP Fraud Researcher John Riordan, Cofounder and CTO of OnSip ( http://onsip.com ) Bogdan-Andrei Iancu, Founder and Lead Developer of OpenSIPS ( http://opensips.org ) James Body, Head of R&D at Truphone ( http://truphone.com ) Emil Ivov, Project Leader, Jitsi.org Ben Klang, Technology Strategist at Mojo Lingo and Project Lead for Adhearsion ( http://www.adhearsion.com/ )` Darren Schreiber, CEO 2600Hz ( http://2600hz.org/ ) This is just a small sample of the Speakers this year! Register at ClueCon.com now and get extra entries into the door prize drawings while you still can! The year we have a pile of goodies from our friends at ThinkGeek ( http://www.thinkgeek.com ) including a Laser-Guided Tactical Tie, Gunnar Intercept Computer Glasses, the Thing Geek Annoy-A-Tron and others! See https://www.cluecon.com/thinkgeek-sponsors-cluecon-giveaways/ for more information!. Don¹t forget ClueCon starts Monday Afternoon this year! Monday will be the Hack-A-Thon will all the developers around! And Stay later in the Week for FreeSWITCH Training! Sign up for the training now as space is limit! See https://www.cluecon.com/freeswitch-training-is-coming-to-cluecon/ for more information! And for the hacker/maker in you, Truphone will be sponsoring Dangerous Demos! Show off your latest creation in a 3 to 4 minute Demo and the best demos will win a prize! Audience Live Voting will determine the winners! Visit https://www.cluecon.com/dangerous-demos-coming-to-cluecon/ for more information and entry details! ClueCon 2014 is shaping up to be the absolute best ClueCon Ever! Don¹t miss out on the Open Source VoIP Conference that is For Developers and organized by developers! Follow us on Twitter @ClueCon for updates as they happen! See you at ClueCon! Https://www.cluecon.com/ -- Ken http://www.ClueCon.com http://www.FreeSWITCH.org http://www.OSTAG.org Twitter: @ClueCon http://www.FreeSWITCH.org

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secure websocket connection refused
by ik 23 Jun '14

23 Jun '14

Hello, I'm using Freeswitch 1.4.6 on CentOS 6 . I'ved configured Freeswitch to work with webrtc, and created certification based on http://wiki.freeswitch.org/wiki/SIP_TLS. I have also enabled the ws-binding and wss-binding at internal.xml. When doing netstat, I can see both 5066 and 7433 binded: netstat -lnpt | grep freeswitch tcp 0 0 37.139.XXX.XXX:7443 0.0.0.0:* LISTEN 27937/./freeswitch When I try to access the insecure ws (5066), I have a connection: tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7f095c00b530): events IN tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7f095c00b530): new secondary tport 0x7f095c09b6a0 tport.c:2620 tport_accept() tport_accept(0x7f095c00b530): incoming secondary on ws/37.139.XXX.XXX:5066/sip failed. reason = WS_INIT But when I try to use the wss connection, I'm getting connection refused. I'm testing it on the same machine with the following command: openssl s_client -connect 37.139.XXX.XXX:7433 My cert folder has the following certifications: drwxr-x--- 2 root root 4096 Jun 23 08:42 CA -rw-r----- 1 root root 1270 Jun 23 08:42 agent.pem -rw-r----- 1 root root 729 Jun 23 08:42 cafile.pem -rw-r--r-- 1 root root 1270 Jun 23 08:46 dtls-srtp.pem -rw-r--r-- 1 root root 1270 Jun 23 08:46 wss.pem When using wireshark, I can see that the ACK packet get a reset flag after sending a SYN request. What am I missing here or doing wrong ? Thanks, Ido

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Shorewall QoS/EF Packet Marking
by Jason Moran 23 Jun '14

23 Jun '14

I have found the wiki entry for QoS to mark the packets with "EF" such that my ISP prioritizes the voice traffic accordingly: https://wiki.freeswitch.org/wiki/QoS However, we are using Shorewall (Shoreline Firewall) to manage IPTables on our CentOS 5.x server, and frankly this entry on Packet Marking in Shorewall is confusing: http://shorewall.net/PacketMarking.html Has anybody managed to implement this (from the FS QOS page) in Shorewall? : # Mark RTP packets with EF: iptables -t mangle -A OUTPUT -p udp -m udp --sport 16384:32768 -j DSCP --set-dscp-class ef Thanks, Jason

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setup presence for multiple FS
by Assaf Dahary 23 Jun '14

23 Jun '14

Hi, I have several FS servers in different locations and they are all managed from a central backend server DB. Currently each FS maintains its own local MySql DB tables which were created on installation. Routing calls between agents on different FS server are done with my own custom scripts. *I'm not using central sip proxy server like OpenSips/Kamailio. Now I would like to share presence between all agents that are registered on different FS servers, BUT I cannot make it because each FS server manages its own internal MySql DB with local presence. How can I share presence between all agents from all FS servers? Should I force all FS servers to share a single MySql DB on the backend server (set each DSN to point on the same remote DB)? Please advise Regards Assaf

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NDLB-connectile-dysfunction and 1.5.13b
by Iskren Hadzhinedev 23 Jun '14

23 Jun '14

Greetings! I'm looking at migrating my old FreeSWITCH 1.2.23 to 1.5.13b from git. I have a Yealink T42G phone that has lots of issues, one of them being call drops after 32 seconds. With 1.2.23 I was able to solve the problem using the following in my dialplan: <extension name="Yealink Detection" continue="true"> <condition field="${sip_user_agent}" expression="^Yealink"> <action application="set" data="sip-force-contact=NDLB-connectile-dysfunction"/> </condition> </extension> After the 1.5.13b upgrade this no longer works. The full version I'm currently using is: FreeSWITCH Version 1.5.13b+git~20140621T003222Z~47891d5caa~64bit (git 47891d5 2014-06-21 00:32:22Z 64bit) Is this intended or I'm missing something obvious here :) Thanks! Kind regards, -- Iskren Hadzhinedev

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