- FreeSWITCH-users - FreeSWITCH Mailing Lists

Error "Cannot create outgoing channel"
by PICHARD, GUILLAUME 19 May '22

19 May '22

Hello, I'm using MicroSIP with FreeSwitch. I can see presence so I guess my MicroSIP is working properly I can text message but I cannot make a call (audio or video). The error in FreeSwitch seems to be: 2022-05-19 09:22:03.475283 [ERR] mod_conference.c:1590 Cannot create outgoing channel, cause: INCOMPATIBLE_DESTINATION (full log there: https://justpaste.it/2h7si) Do you have any idea what might be wrong ? Thank you, Regards, Guillaume. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

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Is the group active?
by Alexander Perkins 18 May '22

18 May '22

Hi All. I have not seen any messages from this group in a bit... is it still active? Thanks, Alex

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FS on RHEL 8.x (and derivatives)
by Unai Rodriguez 17 May '22

17 May '22

Dear List, When will FreeSWITCH officially support installation RHEL 8.x and derivatives? It seems this is not yet the case: https://freeswitch.org/confluence/display/FREESWITCH/Installation Is there a preferred RHEL derivative? How about AlmaLinux? Thank you, Unai Rodriguez

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Originate for a TLS Trunk Connection (Outgoing Calls)
by Adrian Worutowicz 17 May '22

17 May '22

Hello, The tele2.se SIP trunk provider connects to Freeswitch using TLS. Then he sends OPTIONS, which Freeswitch acknowledges. The incoming calls work perfectly. However I have a big problem with the outgoing calls. I use originate. The trick is to use the existing TLS connection. In the SIP external profile, I've set : <param name="reuse-connections" value="true"/> Still the originate does not work. Whatever I try, I get NORMAL_TEMPORARY_FAILURE. I suspect that Freeswitch tries to create a new connection to the provider, instead using the existing one. Does anyone know a correct syntax of the originate command, so that Freeswitch reuses the existing TLS connection ? Thanks in advance, Adrian.

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log format change breaking fail2ban? percentage added to logs? how to revert?
by Jelle de Jong 06 May '22

06 May '22

Hello everybody, The log format seem to have changed, how can I get it back to what it used to without the percentage thing? A search on Google failed me to provide an answer... # fs_cli -x version FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit (-release-19-883d2cb662 64bit) # grep pref_line /etc/fail2ban/filter.d/freeswitch.conf _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+)? failregex = %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ SIP auth (failure|challenge) $(REGISTER|INVITE)$ on sofia profile \'[^']+\' for \[[^\]]*\] from ip <HOST>$ %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ Can't find user \[[^@]+@[^\]]+\] from <HOST>$ 2021-10-28 12:28:55.736515 99.20% [WARNING] sofia_reg.c:1806 SIP auth failure (REGISTER) on sofia profile 'internal' for [30640(a)82.94.245.xxx] from ip 217.138.220.158 2021-05-20 16:44:54.159814 [WARNING] sofia_reg.c:1740 SIP auth failure (REGISTER) on sofia profile 'internal' for [123456(a)82.94.245.xxx] from ip 193.107.216.182 # fgrep percentage /etc/freeswitch/ -R What is the use case of this new percentage in the log files? Kind regards, Jelle de Jong

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uuid_broadcast hang and start after the call hangup
by Dmitry Mordovin 06 May '22

06 May '22

Hello I have LUA script which start a call (new session generated) Code here new_session = freeswitch.Session(destination); if (new_session:ready()) then new_session_disp = new_session:getVariable("endpoint_disposition"); while(new_session:ready() and new_session_disp ~= "ANSWER") do <------>new_session_disp = new_session:getVariable("endpoint_disposition"); end api = freeswitch.API(); api:executeString("bgapi uuid_park " .. tostring(new_session.uuid)); end On answer event I park the call and script finished. I see single active call. Listing below. freeswitch@freeswitch43> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 3aa3a274-c23d-11eb-9150-e74dc4cdb20a,outbound,2021-05-31 18:22:59,1622485379,sofia/external/1000,CS_PARK,,0000000000,,1000,,,,ACTIVE,Outbound Call,1000,,3aa3a274-c23d-11eb-9150-e74dc4cdb20a,freeswitch43,,,,,,,,,,,,,,,,,,,,,, 1 total. After all I execute command from cli: Command looks like: uuid_broadcast 3aa3a274-c23d-11eb-9a50-e74dc4cdb20a lua::’next-script.lua' both In logs I see, command was sent to FS success, but not executed! Moveover, after call hangup, FS start execution of command but can’t coz call in terminate state. 2021-05-31 18:23:30.312133 [NOTICE] sofia.c:1079 Hangup sofia/external/1000 [CS_PARK] [NORMAL_CLEARING] 2021-05-31 18:23:30.312133 [DEBUG] switch_ivr.c:625 sofia/external/1000 Command Execute lua(next-script.lua) 2021-05-31 18:23:30.312133 [DEBUG] switch_core_session.c:2668 sofia/external/1000 ZOMBIE EXEC lua(next-script.lua) EXECUTE sofia/external/1000 lua(next-script.lua) What/why is ZOMBIE EXEC ? Could someone help me to find reason of strange behavior FS? Looks like the uuid_broadcast command stay in queueu and wait something… and start after call has hangup.

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Getting Bill Seconds in Lua
by Alexander Perkins 04 May '22

04 May '22

Hi All. I am trying to access the billmsec from lua, but not being very successful at it. Here's what I've done so far: freeswitch.consoleLog("info", "Billable Duration: " .. session:getVariable("variable_billmsec")); However, when I look at the cli, I get an error because it is nil. I am calling this script from the api_hangup_hook. Any suggestions? Thank you, Alex

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SIP TLS - Certificate Revocation List (CRL checks)
by Weber, Thomas 04 May '22

04 May '22

Hi We have configured FreeSWITCH to use TLS (SIPS/SRTP). This works fine except that no CRL checks seem to be performed on certificates. There does not seem to be a way to turn on CRL checks. Is this correct? If so, is there a specific reason why this has not been implemented? The TLS implementation is based on OpenSSL, so support for CRL checking should be relatively easy. Thanks for your help Thomas Wichtiger Hinweis: Diese E-Mail kann Betriebs- oder Geschäftsgeheimnisse oder sonstige vertrauliche Informationen enthalten. Sollten Sie diese E-Mail irrtümlich erhalten haben, ist Ihnen eine Kenntnisnahme des Inhalts, eine Vervielfältigung oder Weitergabe der E-Mail ausdrücklich untersagt. Bitte benachrichtigen Sie uns und vernichten Sie die empfangene E-Mail. Vielen Dank. Important Note: This e-mail may contain trade secrets or privileged, undisclosed or otherwise confidential information. If you have received this e-mail in error, you are hereby notified that any review, copying or distribution of it is strictly prohibited. Please inform us immediately and destroy the original transmittal. Thank you for your cooperation.

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Java esl-client repository bugs and status
by Yusuf KALAY 02 May '22

02 May '22

Hi all We are using FreeSWITCH for a long time. And there is a known issue with the java client library. But seems the repository is abandoned or not maintained anymore. I opened a PR for the two biggest known issues from Github. And I want to review from someone. Who is authorized for that? How do we progress the fix for the known issues on that repository? Thanks Yusuf

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Genesys Cloud SIP integration with TLS fails with multiple retransmissions when siptrace on
by Unai Rodriguez 01 May '22

01 May '22

Dear List, We are experiencing failures with connection resets and multiple retransmissions when receiving traffic from Genesys Cloud SIP trunk on FreeSWITCH v1.10.5 and v1.10.7 once we run “sofia global siptrace on” on fs_cli as this tcpdump shows: https://www.dropbox.com/s/4ozkv38scykxvmt/fs-genesys-TLS-5061-TRACES_ON.txt ========================================================= $ grep retrans fs-genesys-TLS-5061-TRACES_ON.txt nta.c:7017 _nta_incoming_timer() nta: timer G fired, retransmitting 200 reply nta.c:7017 _nta_incoming_timer() nta: timer G fired, retransmitting 200 reply nta.c:7017 _nta_incoming_timer() nta: timer G fired, retransmitting 200 reply nta.c:7017 _nta_incoming_timer() nta: timer G fired, retransmitting 200 reply ========================================================= If we turn the sip traces off or use TCP or UDP (without TLS) it works fine. Also our setup works with Genesys on prem and pretty much any other contact centres we came across. This is turning into a blocker for us, is there a way to accelerate resolution? We’ve created it here too: https://github.com/freeswitch/sofia-sip/issues/124 Thank you With best wishes, Unai Rodriguez

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