- FreeSWITCH-users - FreeSWITCH Mailing Lists

Setting user to user information (UUI)
by Varsha Agarwal 07 Mar '17

07 Mar '17

Hi, In Freeswitch, is it possible to set SIP UUI (user to user information) before transferring the call another application? We want to add some information to the call so the application on the other end can retrieve it? Thanks, Varsha

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Packet Loss and Round Trip Time
by Oleg Stolyar 07 Mar '17

07 Mar '17

Hi guys, is there a way to get Packet Loss and Round Trip Time in FS CDR without using a 3rd party tool like VoIP Monitor?

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how to debug call (audio/video) quality
by Mundkowsky, Robert 07 Mar '17

07 Mar '17

We use FreeSWITCH for Verto video conference calls. Sometimes some users experience noise audio and video. How do I go about debugging call quality problems? >From what I have read, problems are likely due to packet loss and jitters, but I don' have a way to determine if this is the problem or a system issue. >From the recent FreeSWITCH Book, seems like using the HOMER might help. I have watched the Chrome webrtc charts before and I don't seem to find them useful, granted maybe I do not understand them well. Also if to packet loss and jitters is the problem, is there any way to try to fix such problems? Robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

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playback PLS, M3U or XSPF file formats
by Anonim Stefan 07 Mar '17

07 Mar '17

Hi, Is there a way to playback playback PLS, M3U or XSPF shoutcast streams (e.g. [1])? I tried both mod_vlc[2] and mod_shout[3] without success. I tried playback a local pls file without success. [1] http://yp.shoutcast.com/sbin/tunein-station.pls?id=403432 [2] vlc://http://yp.shoutcast.com/sbin/tunein-station.pls?id=403432 <http://yp.shoutcast.com/sbin/tunein-station.pls?id=403432>[3] shout:// yp.shoutcast.com/sbin/tunein-station.pls?id=403432 Thank you, Stefan

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set_profile_var question
by Abaci B 07 Mar '17

07 Mar '17

Just wondering if there is something like set_profile_var to get a profile variable , there doesn't seem to be a get_profile_var or at least a way I can get the value of Caller-Orig-Caller-ID-Name Thanks

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FS account got hacked **urgent**
by Siju Nair 06 Mar '17

06 Mar '17

Hi Team my account got hacked and attacked using my DID number as caller id and making calls via my FS server. in logs i could notice this sofia/external/'hi'or'x'='x' ... what does this mean and how can they set my did as caller id and make calls... Urgent help needed. Thanks, Siju Nair

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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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