March 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Bug in libs/srtp when OpenSSL is used to provide AES-256?
by Richard Chan 06 Mar '17

06 Mar '17

RFC-6188 violation when FreeSWITCH is compiled with libs/srtp to use OpenSSL? The cipher_id_type_t is set to AES_256_ICM, (if OpenSSL is NOT used then the cipher_id_type_t is set to AES_ICM). This means that in srtp.c: srtp_protect_rtcp() and srtp_unprotect_rtcp() the wrong code path will be chosen for the ICM nonce and keystream will be reused on consecutive RTCP packets. srtp_protect_rtcp() also srtp_unprotect_rtcp(): /* * if we're using rindael counter mode, set nonce and seq */ if (stream->rtcp_cipher->type->id == AES_ICM) { v128_t iv; iv.v32[0] = 0; As a result FS 1.6.15 is generating invalid SRTCP packets when AES-256 is being used (and libs/srtp is compiled to use OpenSSL). Note: RTP explicitly checks for AES_ICM and AES_256_ICM so it is not affected. It will be affected if AES-192 is chosen. This is also seems to be in upstream. -- Richard Chan

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200 OK not forwarded
by David Villasmil 05 Mar '17

05 Mar '17

Hello Guys, I'm using FreeSWITCH (Version 1.9.0 git f3b7105 2016-11-15 01:50:51Z 64bit) is ready And i'm seeing the issue where the 200OK is not being forwarded from FS to the client. I'm attaching FS Log and sip trace. 1.1.1.1 = Client (Grandstream GXP1165) 9.9.9.9 = freeSWITCH 10.10.10.10 = Terminating GW (my_gateway) Been scratching my head for some time... Appreciate your help! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Multi tenant dlalplans
by Rick Jarvis 04 Mar '17

04 Mar '17

Here’s one I’ve hit before but never quite decided on. What’s the best way to avoid tromboning in a multi tenanted environment. Is it to include the public dialplans in each of the tenanted dialplans? R

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Kill all calls from an account
by Aqs Younas 04 Mar '17

04 Mar '17

Hi, I want to kill all calls originating from an account. I know I could get their uuid through some ESL program and kill them one by one on demand. But i am looking for better way to achieve this. Like grouping them to some variable and excuting a command on it. Any pointer on this is much appreciated. Thanks

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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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