- FreeSWITCH-users - FreeSWITCH Mailing Lists

Set BLI when specific Valet Parking slot is full?
by Sean Devoy 04 Mar '17

04 Mar '17

Hi All, My customer says his deli staff cannot handle the concept of picking up a Valet Parking call without the busy lamp indicator flashing. Can anyone explain how to set the busy lamp indicator when there is a call parked in a specific slot? (and if it can be done). I am thinking I will assign one line button on the phone to a special extension and make it an auto-dial to that parking place. Now if I can make it blink, we are set to go. Thanks in advance, Sean

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Bridge - USER_NOT_REGISTERED
by Igor Olhovskiy 03 Mar '17

03 Mar '17

Hi! I’m facing a strange issue I can’t get why happening. I have Freeswitch 1.4.26 server on a public IP. 2 users are registered - 201 and 202. 201 user is registered from address, that is from one subnet to public of Freeswitch. 202 - totally different. All 201 and 202 are behind NAT. So, I’m trying to call from 202 to 201, but getting USER_NOT_REGISTERED EXECUTE sofia/internal/202(a)pbx3.abbyconnect.com bridge(user/201(a)pbx3.abbyconnect.com) ….. 2017-03-01 15:13:32.989992 [NOTICE] switch_ivr_originate.c:2759 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2017-03-01 15:13:32.989992 [DEBUG] switch_ivr_originate.c:3747 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2017-03-01 15:13:32.989992 [NOTICE] switch_ivr_originate.c:2759 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2017-03-01 15:13:32.989992 [DEBUG] switch_ivr_originate.c:3747 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] But. freeswitch@internal> show registrations reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata 201,pbx3.abbyconnect.com,jjrRmyMeA6NyWFDzw-2SgQ..,sofia/internal/sip:201@192.168.3.111:36490;rinstance=fdd02e2e94b8fba0;transport=tcp,1488403310,24.234.110.55,63940,tcp,pbx, 202,pbx3.abbyconnect.com,a707f9609d7fecf432d00074da1ef5cf@0:0:0:0:0:0:0:0,sofia/internal/sip:202@192.168.1.233:5060;transport=udp;registering_acc=pbx3_abbyconnect_com;fs_nat=yes;fs_path=sip%3A202%40176.104.49.75%3A5060%3Btransport%3Dudp%3Bregistering_acc%3Dpbx3_abbyconnect_com,1488400362,176.104.49.75,5060,udp,pbx, So, I see user is registered, but bridge is not trying to reach it. ( I see no INVITEs) But. When I’m moving this server to other IP, to not share same net with 201 extension, all works fine. at 201 I have same router as 202. Internal profile settings to NAT and networks aggressive-nat-detection [true] local-network-acl [localnet.auto] nat-options-ping [true] all-reg-options-ping [true] NDLB-force-rport [true] Thanks. Regards, Igor

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CPU sys load issue
by Antonio Silva 03 Mar '17

03 Mar '17

Hi, i'm having an issue with FS that gets lock every morning with cpu sys hitting 80%, because is limited in fs with min-idle, it could reach the 100%, so the calls are rejected by load, to solve the problem i kill fs and start it again. I discard a load issue, because after restarting freeswitch the system increases in number of sessions/cps and no problems at all until next day.... it's given me crazy... i've enough memory, lots of cores... after restart fs the system stays at less than 30% of cpu usage. Right now i have hpet as clocksource, transcoding is minimal (use inheric_codec, only happens because of different ptimes with some endpoints) . anyone having similar problem? or any tip i can do to debug this issue? For info the Jira ticket can be located at https://freeswitch.org/jira/browse/FS-9957, i've done a gcore, top, htop when the system is with high cpu sys but i can't figure the thread that is causing the issue... initial i thought it was because of module lua (most of my logic is done there), but today only 2 threads where running mod_lua when it happen. Appreciate the help. -- Saludos / Regards / Cumprimentos, António silva

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Re: [Freeswitch-users] Unbridge inbound and outbound legs
by Nikolay Zaytsev 03 Mar '17

03 Mar '17

Hello Chris, Thank you for your advise I will try it. I was also thinking about using the plain park() of the inbound call. Best regards, Nikolay

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Change RTP port range for specific call
by Joel Serrano 02 Mar '17

02 Mar '17

Hi all, I know you can set the upper/lower RTP port limit in autoload_configs/switch.conf.xml but, is it possible to have an upper/lower limit set and change it for a specific call? For example: switch.conf.xml has: <param name="rtp-start-port" value="16000"/> <param name="rtp-end-port" value="32000"/> We have certain clients that require a VPN connection to reach our servers, and the VPN blocks all UDP traffic from ports >10000. One solution would be of course just to lower our range, ok... But is there a way that I can set the RTP port range for a specific call? Something like this in diaplan: [...] <action application="set" data="rtp-start-port=3000"/> <action application="set" data="rtp-end-port=4000"/> [...] I don't know if this is doable some way... If FS doesn't have this option, maybe with a Kamailio + rtpengine in front we could potentially achieve this? Thanks! Best regards, Joel.

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Setting codec on the fly on incoming legs
by Oleg Stolyar 02 Mar '17

02 Mar '17

Hi guys, is there a way to set the codec string on the fly in the dialplan for an incoming leg? I tried setting codec_string and absolute_codec_string but it didn't seem to work on incoming calls.

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Unbridge inbound and outbound legs
by Nikolay Zaytsev 02 Mar '17

02 Mar '17

Good day to everyone! Could you please advise me how to unbridge inbound and outbound legs via ESL? Basically, what I need is on certain dtmf on outbound leg, "unbridge" inbound and outbound legs and put inbound call on music_on_hold. Both legs should not affect each other after "unbridge" Please advise me how to do it, I am lost :( Thank you for advise, Best regards Nikolay

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Recording Stopped out of the Blue
by Christoph Russow 02 Mar '17

02 Mar '17

Hi all, i just discovered (at least) two calls where the recording stopped some seconds after it was started without a notice why it stopped. I only see: 2017-02-21 14:28:29.585573 [DEBUG] switch_core_media_bug.c:808 Removing BUG from sofia/default/<number>@<carrieraddress> in the log. The recording was started arround 7 seconds before the above line appeared in the log and an corresponding event was sent to the eventsocket. What are the reasons for freeswitch to stop a running recording? Things i've checked already: - HDD has enough free space for recordings. - permissions are ok (other calls record just fine on same dir) - no command was sent to stop recording (according to the log) - no dtmf tones in the pcap of the calls - no error message in the log for the calls or arround the record end line We do the recordings for our Customes so it is very crucial that they work correct. I apreciate every hint about this. Best Regards Christoph Russow -- ________________________________________________________________ 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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debugging config errors
by Rick Jarvis 02 Mar '17

02 Mar '17

When getting something like: freeswitch@internal> reloadxml +OK [[error near line 7391]: unexpected closing tag </condition>] I’m sure I used to do something along the lines of: cat -n /var/log/freeswitch/freeswitch.xml.fsxml | sed -n '7380,7400p' And there at 7391 would be the offending line, surrounded by some of the lines before and after. Now, however, the line is never at the corresponding number. Am I looking at the wrong file maybe?

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Freeswitch Verto + Video installation
by Gonzalo Gasca Meza 02 Mar '17

02 Mar '17

Hi all, I would like to use Freeswitch Verto + Video. I'm using documentation from Wiki: https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie *Question* a) I will be using this section "*Building From Source. **Compiling Release Branch (production)*" to build my Freeswitch, I found a note at the end: "Quick Start FreeSWITCH Demo With Verto Communicator" Is that part still required for Verto demo? or previous "Building From Source. Compiling Release Branch (production)" already includes some of the steps. Thanks -Gonzalo

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