June 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Intercept command does not work in bypass media mode
by happy.neko 27 Jun '16

27 Jun '16

Hi, I need to implement this scenario: inbound SIP call A is waiting indefinitely for inbound call B and then they are bridged together. RTP media should flow directly from A to B and vice versa while FreeSwitch should control SIP signaling. I made some progress but I could not make media bypass mode to work. My SIP profile settings: <param name="inbound-late-negotiation" value="true" /> <param name="inbound-bypass-media" value="true" /> Dialplan for leg A:  <action application="park" /> LUA script for leg B: leg_a_uuid = ... session:execute("intercept", leg_a_uuid) This works but FreeSwitch is still in RTP media path with following error in the log: [ERR] switch_ivr.c:1718 Can't re-establsh media on sofia/myprofile/xyz It worth mentioning that I do not answer or send early media before executing intercept command so there is no prior media on both channel A and B. Any ideas what needs to be done to make this scenario work without RTP proxying? -- Best regards, I. Sokolov

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Disable INFO messages
by Grant Bagdasarian 27 Jun '16

27 Jun '16

Hello, How do I disable INFO messages sent by Freeswitch? I have no idea why these messages are sent and I would like to disable them completely. Regards, Grant

2 1

Freeswitch on AWS
by yann christophe 27 Jun '16

27 Jun '16

Hello everyone, I would like to know if someone has an experience on AWS with Freeswitch. Because i'll deploy 2 Freeswitch servers on EC2 and i would like to use S3 for the directory "Dialplan". I want this Directory is the same for these 2 servers, I currently use rsync, maybe it's better to use S3 for that. What do you suggest me ? I just started to work on AWS Thanks Best regards

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No ringback / ESL channel progress when using originate
by Gary Foreman 27 Jun '16

27 Jun '16

Hi all, Ever since upgrading past 1.6.6 I've had problems when using the originate command. I'm unable to hear ringback on the call and no channel_progress ESL event is emitted when the "183 Session Progress" message comes back from my SIP provider. Once the call is connected I can talk as usual. Here are the console outputs; 1.6.6 (working) https://pastebin.freeswitch.org/view/3998aa81 1.6.9 (no ringback) https://pastebin.freeswitch.org/view/65002d96 Thanks in advance

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How to: drop inband dtmf
by Avi Marcus 27 Jun '16

27 Jun '16

I'm not sure this is posted anywhere, so I wanted to make sure it's here. I have a carrier that sends both inband and RFC2833 DTMF. For FreeSWITCH, this is fine, but if I bridge it on to someone else, then the remote system often processes both for duplicate DTMF. Here's a local fix that works for me (but you can't be in bypass mode) to remove the inband (it always had rfc2833, but not *always *inband.) Set this for the affected leg: <action application="set" data="rtp_manual_rtp_bugs=FLUSH_JB_ON_DTMF"/> > <action application="set" data="jitterbuffer_msec=60"/> > <action application="set" data="sip_jitter_buffer_during_bridge=true"/> I have in my notes: <action application="export" data="nolocal:drop_dtmf=true"/> but I don't see on the wiki that it's for sure that it works on inband, a quick look at the code and I still wasn't sure. -Avi Marcus BestFone

1 0

Wrong hangup cause in CDR
by k.presler＠megafit.su 27 Jun '16

27 Jun '16

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Changing Auto stun / rtp / DTLS
by Agustí Ubalde 26 Jun '16

26 Jun '16

HI all, In the log file I observed the trace *Changing Auto stun / rtp / DTLS* It only happens on connections from a particular IP. Anyone know explain to that is? Regards, *PRESENCE TECHNOLOGY* *Agustí Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 300 Fx: +34 93 10 10 333 *www.presenceco.com* <http://www.presenceco.com/> *Follow us on:* *[image: tw]* <http://twitter.com/presencetech>*[image: yt]* <http://www.youtube.com/PresenceTechnology>*[image: in]* <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]* <http://www.slideshare.net/PresenceTech>*[image: fb]* <http://www.facebook.com/presencetech> For additional information, please visit our website *www.presenceco.com* <http://www.presenceco.com/> -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

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Verto + external database
by Gregor Nanger 26 Jun '16

26 Jun '16

I guess that verto uses db for storing user registrations. I am using external database (MS SQL) instead integrated sqlite, but I cannot se verto users in registrations table. Does verto uses external db?

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group_call>user_call>dial-string - Capture UUID ?
by Jonathan Hunter 26 Jun '16

26 Jun '16

Hi All, I am using a lua script to route to a hunt group of extensions using the group_call function, which then in turn actions a user_call, using dial-string. After the dial-string has completed and FS creates an outgoing call such as; switch_channel.c:1091 New Channel sofia/internal/test1@testdomain [a076f82b-1db8-4385-914c-05dabc280d93] Is it possible to grab the UUID created so I can store it in a database? Many thanks Jon

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Photo
by William Colls 26 Jun '16

26 Jun '16

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