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Recovery capabilities
by Telium Technical Support 18 Oct '21

18 Oct '21

I've been reading about the recovery capabilities of FreeSwitch. It SOUNDS a bit like a work in progress (possibly waiting for funding) - but maybe the docs are out of date. In terms of current state, can FS: 1. resume calls in progress (on restart) 2. place existing calls back into the dialplan where they where 3. restore queue agents & callers (and their order)

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Dial with SIP credentials without defining gateway?
by Antony Stone 15 Oct '21

15 Oct '21

Hi. I've recently started using FreeSwitch, after being familiar with Asterisk for many years. I'm wondering whether FreeSwitch has an equivalent to Asterisk's facility for dialling out to a SIP server using credentials supplied as part of the dial command, instead of needing a pre-defined gateway? In an Asterisk dialplan, you can say something like Dial(SIP/${Destination}:${SIPpass}::${SIPuser}@${SIPhost}) and that will dial to the Destination number on SIPhost using the SIP credentials SIPuser/SIPpass. Nothing needs to be configured in advance. Can FreeSwitch do something like that too? Thanks, Antony. -- "The problem with television is that the people must sit and keep their eyes glued on a screen; the average American family hasn't time for it." - New York Times, following a demonstration at the 1939 World's Fair. Please reply to the list; please *don't* CC me.

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Passing custom sip headers using re-invite
by Babak Yakhchali 13 Oct '21

13 Oct '21

Hi I need to pass some custom sip headers between 2 fs boxes after the call is established. To do this I'm using "uuid_setvar sip_h_X-myvar=value" on the channel and calling uuid_media_reneg to create a re-invite, fs is sending out the re-invite with new custom headers but the other fs box is not adding these new custom sip headers to the channel. How can I make fs to set new custom headers on the receiving side of the re-invite? fs1 ----invite---> fs2 .... call is established fs1 <---re-invite--- fs2 (re-invite with new custom sip X headers) fs1 ... not adding new headers to channel thanks

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Avoiding DDoS with verto?
by David P 13 Oct '21

13 Oct '21

After reading about recent DDoS attacks on VOIP providers in https://www.rtcsec.com/post/2021/09/massive-ddos-attacks-on-voip-providers-… in which Freeswitch is mentioned, I wondered what current practices are for services that must serve the public Internet. For example, a service that is purely verto-based seems like it could protect itself in this way: 1) Block requests on all ports (except the verto WSS login) unless the request is from an address that's already part of signaling. 2) To protect signaling, put it behind AWS API Gateway, which provides rate-limiting, and add an authorization check. Does this seem like it would defend against DDoS? Can FS be configured to do #1?

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User-to-User RFC 7433
by Adrian Worutowicz 11 Oct '21

11 Oct '21

Hello, I need to add the User-to-User field to the SIP header - for incoming calls. I have not managed to do it (neither for incoming nor for outgoing calls). In ./conf/dialplan/default.xml, <extension> section, I added : <action application="set" data="sip_h_User-to-User=123"/> or <action application="set"><![CDATA[sip_h_User-to-User=${sip_Call_ID};tenant=1101;DID=${sip_to_user}]]></action> No result. In sofia_presence.c, I've seen some code for "User-to-User", so it should be possible. But how to do it ? Best regards, Adrian.

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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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