October 2021 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Sofia SIP/503 System Busy
by António Silva 07 Oct '21

07 Oct '21

Hi, Recently update a server to Debian 10 with freeswitch git Version 1.10.7-dev git 432bfc0 2021-07-27, and after a few days I start to see lot of expired registers and new invites rejected with “ 503 System Busy”. I restart FS and everything is ok. Also no log print in FS console that could point me to the root cause of the issue. The server before the update was running debian 8 with freeswitch Version 1.10.4-dev git 546bebb 2020-08-03 and never see this issue. The hardware of the server is a ProLiant DL165 G7: 2 x AMD Opteron(tm) Processor 6128, Cores por CPU: 8, Cores totales: 16 26 GB RAM The system is not with high load at the time the issue was occurring, fs status report: freeswitch@srvone> status UP 0 years, 2 days, 21 hours, 50 minutes, 23 seconds, 264 milliseconds, 655 microseconds FreeSWITCH (Version 1.10.7-dev git 432bfc0 2021-07-27 19:25:13Z 64bit) is ready 147010 session(s) since startup 86 session(s) - peak 251, last 5min 91 0 session(s) per Sec out of max 50, peak 26, last 5min 5 2000 session(s) max min idle cpu 5.00/66.57 Current Stack Size/Max 240K/8192K From the source code, Sofia.c:2496 if (switch_queue_size(mod_sofia_globals.msg_queue) > (unsigned int)critical) { nua_respond(nh, 503, "System Busy", SIPTAG_RETRY_AFTER_STR("300"), NUTAG_WITH_THIS(nua), TAG_END()); goto end; } So the trigger here is the msg_queue hitting the critical value, for what I could check in the source code, is the value of 7200. This value is the maximum number of sip dialogs that FS can handle? I’ve some endpoints registered using TCP, can they affect the msg_queue? I can’t think of a route cause for this issue… any hint? Thanks for the help. -- Saludos / Regards / Cumprimentos António Silva

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mod_fifo onhook member uuid not expanding
by Bipin Patel 07 Oct '21

07 Oct '21

hi, im pretty much stuck in this situation where I have users dialing into a DID which FS answers and places callers in a fifo queue, the calls get bridged to on-hook members from the fifo list, they want the ability to press *2 at anytime to record the call, all this works but I need to store the recording filename with the uuid which doesnt seem to expand at all my diaplan is as below: <extension name="support_fifo"> <condition field="destination_number" expression="^0001$"> <action application="answer"/> <action application="set" data="fifo_chime_list=$${base_dir}/sounds/en/us/callie/misc/8000/misc-freeswitch_is_state_of_the_art.wav"/> <action application="set" data="fifo_chime_freq=15"/> <action application="set" data="fifo_orbit_dialplan=XML"/> <action application="set" data="fifo_orbit_context=lanusersdialout"/> <action application="set" data="fifo_orbit_announce=$${base_dir}/sounds/en/us/callie/ivr/8000/ivr-connect_actual_human_being.wav"/> <action application="set" data="fifo_override_announce=$${base_dir}/sounds/en/us/callie/ivr/8000/ivr-one_moment_please.wav"/> <action application="fifo" data="support_fifo@$${domain} in undef $${base_dir}/sounds/music/8000/moh.wav"/> </condition> </extension> my fifo config is as below: <fifo name="support_fifo@$${domain}" importance="0"> <member timeout="15" simo="1" lag="5">{execute_on_answer='bind_meta_app 2 a s record_session::$${base_dir}/recordings/archive/${uuid}.mp3',Caller-Context=lanusersdialout,absolute_codec_string=PCMU,call_timeout=30,fifo_member_wait=nowait}user/88</member> </fifo> anyways to pass any sort of uuid from the dialplan or get the uuid of the originating channel using any variables? any help is very much appreciated -- Regards, Bipin

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Jitter and dropping MOS score
by Christoph Russow 07 Oct '21

07 Oct '21

Hi all, we are currently experiencing jitter and dropping MOS score on an increasing amount of calls on our system. i can't see any increased load on our system (neither CPU nor I/O wise). Current stats: ~ 15Mbit/s network traffic (on a 1Gbit/s network) load average: 1,10, 0,82, 0,91 (16 core machine) memory used: 2,26GB/16GB freeswitch@lc1> status UP 0 years, 12 days, 13 hours, 30 minutes, 24 seconds, 649 milliseconds, 831 microseconds FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready 137634 session(s) since startup 218 session(s) - peak 245, last 5min 222 1 session(s) per Sec out of max 30, peak 8, last 5min 2 1000 session(s) max min idle cpu 0.00/95.37 Current Stack Size/Max 244K/244K System ist not virtualized. the only "strange" thing i see is the freeswitch "main"process that consumes (according to htop) >=100% cpu (see attached screenshot). Can this be the problem? if yes what can i do against it. if not what are possible bottlenecks i should weed out? if you need any additional informations to help me sort out this problem please don't hesitate to ask. Thanks in advance and best regards Christoph -- ________________________________________________________________ EMTEX GmbH Christoph Russow Software Engineer Bischof-Otto-Weg 9 D-91086 Aurachtal Geschäftsführer: Markus Enzinger Tel. +49 9132 7490 0 Sitz der Gesellschaft: 91086 Aurachtal Fax. +49 9132 7490 900 Amtsgericht Fürth: HRB6804 ________________________________________________________________

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Bridge to other FS server has no audio ???
by Avi Marcus 06 Oct '21

06 Oct '21

Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again. - FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works. - FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works. - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS server B. Call looks OK, but no media is flowing (I don't hear anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All the same codecs are set in the json cdrs (PCMU). FS server B is to join a conference if that matters. I was assuming it had to do with codecs, but setting absolute_codec_string to PCMU doesn't make any difference in the logs - it's already always PCMU. I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please? -Avi Marcus 1-718-989-9485 (USA) 02-372-1570 (Israel)

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tls-verify-depth
by Victor Bogatyryev 06 Oct '21

06 Oct '21

Hi! I cannot understand how the directive <param name="tls-verify-depth" value="1" /> works. The verification policy is set like this <param name="tls-verify-policy" value="all|subjects_all" />. The certificate chain has one intermediate CA.But I don't see anything in the tport log except for checking the subject of the user's certificate. Regards. -- V.Bogatyryev

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Re: [Freeswitch-users] FreeSWITCH Stops Responding
by Brian West 06 Oct '21

06 Oct '21

Can you try to create one where this happens without all this mess? On Wed, Sep 1, 2021 at 2:41 PM Cary Shouse <cary(a)ytel.com> wrote: > Here is the LUA code regarding creating sessions and bridging. It is > broken up into 5 sections. > The top part is the common functions, and the 4 sections below are > different call scenarios. > https://pastebin.com/qqGw3tms > > > On Wed, Sep 1, 2021 at 3:53 PM Brian West <brian(a)freeswitch.com> wrote: > >> I you do sessions from your lua, I need to see how you do that. >> >> On Tue, Aug 31, 2021 at 4:20 PM Cary Shouse <cary(a)ytel.com> wrote: >> >>> Yes, it bridges the inbound call to an outbound one. >>> >>> (ProxyOutbound.luac)https://pastebin.com/RDUMy473 >>> >>> >>> On Tue, Aug 31, 2021 at 8:39 PM Brian West <brian(a)freeswitch.com> wrote: >>> >>>> Guessing you create sessions in lua? ProxyOutbound.luac? >>>> >>>> On Tue, Aug 31, 2021 at 3:33 PM Brian West <brian(a)freeswitch.com> >>>> wrote: >>>> >>>>> and to make sure when you uuid kill one it says it didn't exist but >>>>> show channels had it still listed correct? >>>>> >>>>> On Tue, Aug 31, 2021 at 2:58 PM Cary Shouse <cary(a)ytel.com> wrote: >>>>> >>>>>> Hello, >>>>>> >>>>>> We have got the logs for a few calls that were stuck. >>>>>> (Show channels fe8f206c-2bd9-4af2-b329-35f53228a38a) >>>>>> https://pastebin.com/NTnte7J7 >>>>>> >>>>>> (Extracted from logs fe8f206c-2bd9-4af2-b329-35f53228a38a) >>>>>> https://pastebin.com/sAhFiGjY >>>>>> >>>>>> (Show channels 46c6e04b-a58e-4b82-9968-fb0e9c2c0fb3) >>>>>> https://pastebin.com/bmcdjzAr >>>>>> >>>>>> (Extracted from logs 46c6e04b-a58e-4b82-9968-fb0e9c2c0fb3) >>>>>> https://pastebin.com/aGyY3m9F >>>>>> >>>>>> (Show channels 624cac90-0a7b-11ec-b7d0-6f543a588268) >>>>>> https://pastebin.com/0cnQaRhC >>>>>> >>>>>> (Extracted from logs 624cac90-0a7b-11ec-b7d0-6f543a588268) >>>>>> https://pastebin.com/atX0VXdZ >>>>>> >>>>>> Please let us know if there is anything else we can provide that >>>>>> would be useful. Thank you. >>>>>> >>>>>> >>>>>> On Mon, Aug 30, 2021 at 8:20 PM Brian West <brian(a)freeswitch.com> >>>>>> wrote: >>>>>> >>>>>>> Check the UUID of the session in question, and extract its logs, and >>>>>>> let's diagnose it >>>>>>> >>>>>>> On Mon, Aug 30, 2021 at 12:39 PM Cary Shouse <cary(a)ytel.com> wrote: >>>>>>> >>>>>>>> Yes, that is what we are seeing when running uuid_kill. We >>>>>>>> cannot "replicate" the issue, but it happens multiple times a day for us >>>>>>>> across our servers. We have looked through logs across multiple >>>>>>>> systems, taken SIP traces to see if that showed anything. Last we >>>>>>>> got a Core dump of the running process after the failure. Nothing seems to >>>>>>>> stand out as the cause. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Are there any other debug steps you can think of or what other >>>>>>>> information to look at? We could capture all traffic on the server, or >>>>>>>> recompile with special debug flags whatever it is that might help. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Steps we have taken: >>>>>>>> >>>>>>>> sofia loglevel all 9 >>>>>>>> >>>>>>>> sofia global siptrace on >>>>>>>> >>>>>>>> fsctl loglevel 7 >>>>>>>> >>>>>>>> # Set environment variable >>>>>>>> TPORT_LOG=1 >>>>>>>> >>>>>>>> >>>>>>>> On Sat, Aug 28, 2021 at 12:54 AM Brian West <brian(a)freeswitch.com> >>>>>>>> wrote: >>>>>>>> >>>>>>>>> It's an issue we've been searching for, if you uuid_kill it, and >>>>>>>>> it says no session, it only means the record was NOT removed from the >>>>>>>>> sqlite database. >>>>>>>>> >>>>>>>>> Nobody can seem to replicate the steps required to get into that >>>>>>>>> condition, maybe you could look thru the logs and see if you can replicate >>>>>>>>> what ever happened on that lingering UUID >>>>>>>>> >>>>>>>>> /b >>>>>>>>> >>>>>>>>> >>>>>>>>> On Fri, Aug 27, 2021 at 5:40 PM Cary Shouse <cary(a)ytel.com> wrote: >>>>>>>>> >>>>>>>>>> We recently started having a problem with our FreeSWITCH servers >>>>>>>>>> where calls are working well, with about 200 calls operating sometimes for >>>>>>>>>> a couple of days, and then something happens and all call progress stops. >>>>>>>>>> Then any new inbound calls that come into the server get stuck in initial >>>>>>>>>> status, new inbound get stuck at RINGING. This happens until the max >>>>>>>>>> sessions are hit and of course nothing else works from there. Our thought >>>>>>>>>> is something is creating a lock and stopping anything else from progressing. >>>>>>>>>> We are seeing the stuck sessions using "show channels". Doing a >>>>>>>>>> uuid_kill on the calls seems to make no difference to the output of "show >>>>>>>>>> channels", but we have noticed in the logs that some events do happen to >>>>>>>>>> some of the calls. >>>>>>>>>> We had ran into a similar issue when writing a custom C module >>>>>>>>>> and made some mistakes with session_rwlock etc. But in this case, all of >>>>>>>>>> this is using LUA. >>>>>>>>>> Some general information: >>>>>>>>>> FreeSWITCH Versions 1.10.6 and 1.10.1 >>>>>>>>>> Modules used: >>>>>>>>>> LUA, xml_curl(dialplan), event_socket, >>>>>>>>>> >>>>>>>>>> Most of this is just application playing audio files and or >>>>>>>>>> creating a new call to bridge to. >>>>>>>>>> >>>>>>>>>> Really looking for some help on what debugging we can enable to >>>>>>>>>> help identify what is getting stuck to identify where a bug might be on our >>>>>>>>>> usage or possible issue with FreeSWITCH. >>>>>>>>>> >>>>>>>>>> We have captured a core dump of the running process when it got >>>>>>>>>> into the bad state as well, if that is helpful, we can send it in. >>>>>>>>>> >>>>>>>>>> Thank you for your help. >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> >>>>>>>>>> The FreeSWITCH project is sponsored by SignalWire >>>>>>>>>> https://signalwire.com >>>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and >>>>>>>>>> PSTN services. >>>>>>>>>> Build your next product on our scalable cloud platform. >>>>>>>>>> >>>>>>>>>> Join our online community to chat in real time >>>>>>>>>> https://signalwire.community >>>>>>>>>> >>>>>>>>>> Professional FreeSWITCH Services >>>>>>>>>> sales(a)freeswitch.com >>>>>>>>>> https://freeswitch.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> https://freeswitch.com/oss >>>>>>>>>> https://freeswitch.org/confluence >>>>>>>>>> https://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 >>>>>>>>>> https://freeswitch.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> >>>>>>>>> Brian West | Co-founder and Developer >>>>>>>>> >>>>>>>>> Need Commercial support? email sales(a)freeswitch.com >>>>>>>>> >>>>>>>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI >>>>>>>>> 53045 >>>>>>>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> >>>>>>>>> >>>>>>>>> Email: brian(a)freeswitch.com >>>>>>>>> >>>>>>>>> Mobile: 918-424-9378 >>>>>>>>> >>>>>>>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> >>>>>>>>> >>>>>>>>> [image: https://www.facebook.com/signalwireinc?src=email] >>>>>>>>> <https://www.facebook.com/freeswitch> [image: >>>>>>>>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch> >>>>>>>>> >>>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> Brian West | Co-founder and Developer >>>>>>> >>>>>>> Need Commercial support? email sales(a)freeswitch.com >>>>>>> >>>>>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>>>>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> >>>>>>> >>>>>>> Email: brian(a)freeswitch.com >>>>>>> >>>>>>> Mobile: 918-424-9378 >>>>>>> >>>>>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> >>>>>>> >>>>>>> [image: https://www.facebook.com/signalwireinc?src=email] >>>>>>> <https://www.facebook.com/freeswitch> [image: >>>>>>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch> >>>>>>> >>>>>> >>>>> >>>>> -- >>>>> >>>>> Brian West | Co-founder and Developer >>>>> >>>>> Need Commercial support? email sales(a)freeswitch.com >>>>> >>>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> >>>>> >>>>> Email: brian(a)freeswitch.com >>>>> >>>>> Mobile: 918-424-9378 >>>>> >>>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> >>>>> >>>>> [image: https://www.facebook.com/signalwireinc?src=email] >>>>> <https://www.facebook.com/freeswitch> [image: >>>>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch> >>>>> >>>> >>>> >>>> -- >>>> >>>> Brian West | Co-founder and Developer >>>> >>>> Need Commercial support? email sales(a)freeswitch.com >>>> >>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> >>>> >>>> Email: brian(a)freeswitch.com >>>> >>>> Mobile: 918-424-9378 >>>> >>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> >>>> >>>> [image: https://www.facebook.com/signalwireinc?src=email] >>>> <https://www.facebook.com/freeswitch> [image: >>>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch> >>>> >>> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales(a)freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> >> >> Email: brian(a)freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> <https://www.facebook.com/freeswitch> [image: >> https://twitter.com/freeswitch] <https://twitter.com/freeswitch> >> > -- Brian West | Co-founder and Developer Need Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> Email: brian(a)freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>

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jitters generated from freeswitch to trunk in conference call
by Achintha 06 Oct '21

06 Oct '21

Hi all, I configured 2 freeswitch servers (FreeSWITCH version: 1.10.5-release-17-25569c1631~64bit) on Debian 10.4 as media servers. one opensips server is located in front of the freeswitch servers and opensips act as SIP load balancer. media directly connect with sip trunk and freeswitch.We use dynamic dialplan to generate calls and use g711. Anyway i generate below mentioned scenario 1. registered as extension 2. dial outbound to mobile call through the sip trunk and talk 3. put hold the first call 4. dial another outbound call to another mobile through the same sip trunk and talk. 5. connect both calls (conference). in the 5th step jitter buffers generated from freeswitch to trunk. voice not clear. I have changed codecs and tests (opus,G711a,G711u) but the issue is not fixed. how to solve this problem. Best Regards.. Achintha Udukumbura

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stop channel from entering conference
by David Villasmil 05 Oct '21

05 Oct '21

Hello all, I have a scenario where: A calls B (i.e.: +19995551212) and puts it on hold. A calls C (i.e.: +99911111) and when it answers: The client sends a refer to B to connect to a conference, and then transfers C to the conference. All this works well, but i need to block specific called numbers from entering the conference. I can identify the channels as they come in and i'm trying to hangup the leg i want. I tried with <action application="hangup"/> and also with <action application="set" data="res=${uuid_kill(${uuid})}"/> but all channels are disconnected. Not sure if i'm doing something wrong. Any help would be appreciated Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Multiple api_on_answer?
by Avi Marcus 05 Oct '21

05 Oct '21

Docs show that you can have multiple execute_on, specifically: execute_on_answer_1 Does that also work for api_on_answer? I searched the code for execute_on_answer_1 and couldn't find that or a loop for 1-9. https://freeswitch.org/confluence/pages/viewpage.action?pageId=1048892#id-_… Thanks, -Avi Marcus

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ios 15 issue
by Nathan Stratton 05 Oct '21

05 Oct '21

We recently encountered an issue where all socket connection attempts are repeatedly disconnected and retried upon attempting to connect to FreeSWITCH. Freeswitch does not contain logs of these connection attempts and they seem to occur immediately upon the initial request (likely during the SSL handshake). A look at the Apple developer forums has yielded a number of people experiencing the same issue, pointing to a problem with a transition to a new NSURLSession implementation in ios15. Unfortunately, there is no expected timeline for a fix to address these issues. This is the link to the forum thread relevant to the problem: https://developer.apple.com/forums/thread/685403 ><> nathan stratton

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