- FreeSWITCH-dev - FreeSWITCH Mailing Lists

rtcp-mux support
by openser 26 Nov '12

26 Nov '12

Hi all, Does freeswitch support rtcp-mux feature ? if it support , freeswitch should send rtcp packet by rtp port but not rtp +1 for given audio/video session, my client support rtcp-mux , but freeswitch send rtcp to port rtp +1 .

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FreeSWITCH Weekly News and Notes
by Michael Collins 26 Nov '12

26 Nov '12

Hello all! We are all back to a full week after many of us enjoyed some well-deserved time off last week. However, even though there was a holiday here in the US, the intrepid FreeSWITCH development team was working hard on your behalf. As Ken Rice previously mentioned, Anthony spotted a potential issue in the recently released 1.2.5 version. Therefore, this past Saturday they made 1.2.5.1 available<http://files.freeswitch.org/freeswitch-1.2.5.1.tar.bz2>for us. Many thanks to those who work so hard to make sure that FreeSWITCH is running smoothly for us all. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_21>we spent some time getting everyone up to speed on how to edit the FreeSWITCH wiki <http://wiki.freeswitch.org>, specifically focusing on channel variables <http://wiki.freeswitch.org/wiki/Channel_Variables> pages. Updating documentation is one of the least glamorous aspects of maintaining an open source project. Many thanks to those who've stepped up over the past weeks and months to help us out. With the end of the year upon us we are slowing down a bit in our speaking schedule for the weekly community conference call. We have a few things in the works but nothing yet scheduled. On this week's call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_28>we will be doing a community scrum. Be sure to bring your questions and topics for discussion. If you have a tip or trick that you'd like to share with the group that would be most welcomed. If time permits we will crowdsource a few selected questions from the mailing list. Have a great week and we'll talk to you on Wednesday. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Re: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 77, Issue 6
by David Rot 26 Nov '12

26 Nov '12

Dears 1. In some occasion I get the caller-id with an extension, and in other occasions I get the caller-id without an extension. How do I make sure that I always get the caller-id with an extension? 2. Soon after I perform a recording, I hear a tick sound (like hanging up) and then the call continues without any problems. This situation is very confusing for our customers as they believe that the call was disconnected (because of the tick sound). How can I avoid the tick sound that causes the confusion? Thanks

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Failed DTMF sanity check.
by Debasish Chandra 26 Nov '12

26 Nov '12

Hi, I am getting "Failed DTMF sanity check" randomly (not on every call, 2 - 3 calls out of 10). Application is running using Plivo REST API, while FS receiving call from Sangoma NSG TDM gateway. I have read on FS wiki that "Failed DTMF sanity check" means FS never received RFC2833 DTMF 'end' packet. However call tcpdump shows that DTMF digits has been sent to FS from Sangoma NSG, FS showing error "Failed DTMF sanity check" and ignores the digit. Please let me know if I need to do any extra settings like editing dtmf duration or type. Best Regards, Debasish Chandra

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Redis Module Implementation
by Rodney 26 Nov '12

26 Nov '12

I'm currently looking at the redis module as a way of implementing limits that are shared across multiple servers and noticed that for every limit I create, two keys are set in redis. One key is what I'm expecting, with the name derived from the values I've supplied, however the other key is the same thing but with the hostname prefixed. Looking at the code, this could be either the hostname or the switchname that is being prefixed (depending on if a switchname is set). My question is basically why is it doing this ? I would have assumed that if a dialplan author wanted the key in question to have the hostname or switch name, couldn't they just include it in the call to limit in the dial plan ? It also looks to me like there could be a bug where the hostname prefixed key only gets incremented if there is a max set but will always get decremented. I'm just trying to work out if this is intended behaviour and I have just misunderstood the purpose of the module, or if it is worth trying to put a patch together. Thanks, Rodney

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Re: [Freeswitch-dev] Return code from ESL Message Sending
by Kurtis Heimerl 26 Nov '12

26 Nov '12

Sure. I'm sending a message through event creation, as detailed here on the wiki: http://wiki.freeswitch.org/wiki/Mod_sms#Sending_a_Message_from_a_script The exact script i'm using to do this is here: https://github.com/kheimerl/libvbts/blob/master/libvbts/FreeSwitchMessenger… On Monday, November 26, 2012, Eli Burke wrote: > Kurtis, > > I wouldn't be surprised if it was caused by those patches. Unfortunately > (or perhaps fortunately) I didn't write them. And sending messages via > direct event creation wasn't part of our test matrix. Can you give me some > more detail on exactly what you are doing? > FWIW, we tested from handsets, from ESL scripts, and from fs_cli. > > -ELi > > On Nov 25, 2012, at 4:59 AM, freeswitch-users-request(a)lists.freeswitch.org<javascript:_e({}, 'cvml', > 'freeswitch-users-request(a)lists.freeswitch.org');> wrote: > > *From: *Kurtis Heimerl <kheimerl(a)cs.berkeley.edu <javascript:_e({}, > 'cvml', 'kheimerl(a)cs.berkeley.edu');>> > *Subject: **Re: [Freeswitch-users] Return code from ESL Message Sending* > *Date: *November 25, 2012 1:00:08 AM EST > *To: *FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<javascript:_e({}, 'cvml', 'freeswitch-users(a)lists.freeswitch.org');> > > > *Reply-To: *FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<javascript:_e({}, 'cvml', 'freeswitch-users(a)lists.freeswitch.org');> > > > > > Hi Eli, > > I recently updated my installation, and now my old scripts for sending > messages through event creation aren't working. Basically, FS is sending > tens to hundreds of MESSAGEs a second after I send a SMS::SEND_MESSAGE event. > This was working before your patch (and I'm sure a bunch of other patches), > so I'm hoping you have some intuition as to what might be broken. I'll > start tracking it down myself here in a little bit. > > >

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Freeswitch Hangup
by Jon Lederman 21 Nov '12

21 Nov '12

Hi, I am able to make calls and connect. However after about 30 seconds the call is always disconnected. I am using a public IP address on my server. Everything was working perfectly before. The only thing new is that I upgraded to the new version of linphone. Here is my log file. Any thoughts? Thanks 2012-11-21 00:40:31.418129 [DEBUG] switch_core_session.c:905 Send signalsofia/internal/1000(a)54.245.109.196[BREAK] 2012-11-21 00:40:31.418129 [DEBUG] sofia.c:6285 Channelsofia/internal/1000(a)54.245.109.196 entering state [completed][200] 2012-11-21 00:40:31.418129 [DEBUG] switch_core_session.c:759 Send signalsofia/internal/1000(a)54.245.109.196[BREAK] 2012-11-21 00:40:31.418129 [DEBUG] switch_channel.c:3351 (sofia/internal/1000(a)54.245.109.196) Callstate Change EARLY -> ACTIVE 2012-11-21 00:40:31.418129 [NOTICE] switch_ivr_originate.c:3357 Channel [sofia/internal/1000(a)54.245.109.196] has been answered 2012-11-21 00:40:31.418129 [DEBUG] switch_ivr_originate.c:3414 Originate Resulted in Success: [sofia/internal/sip:1001@75.83.124.194:35948] 2012-11-21 00:40:31.418129 [DEBUG] switch_ivr_originate.c:3414 Originate Resulted in Success: [sofia/internal/sip:1001@75.83.124.194:35948] 2012-11-21 00:40:31.418129 [DEBUG] switch_core_session.c:759 Send signal sofia/internal/sip:1001@75.83.124.194:35948 [BREAK] 2012-11-21 00:40:31.418129 [DEBUG] switch_core_session.c:759 Send signalsofia/internal/1000(a)54.245.109.196[BREAK] 2012-11-21 00:40:31.418129 [DEBUG] switch_ivr_bridge.c:1361 (sofia/internal/sip:1001@75.83.124.194:35948) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2012-11-21 00:40:31.418129 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1001@75.83.124.194:35948) Running State Change CS_EXCHANGE_MEDIA 2012-11-21 00:40:31.418129 [DEBUG] switch_core_session.c:1210 Send signal sofia/internal/sip:1001@75.83.124.194:35948 [BREAK] 2012-11-21 00:40:31.418129 [DEBUG] switch_core_state_machine.c:456 (sofia/internal/sip:1001@75.83.124.194:35948) State EXCHANGE_MEDIA 2012-11-21 00:40:31.418129 [DEBUG] mod_sofia.c:661 SOFIA EXCHANGE_MEDIA 2012-11-21 00:40:31.518129 [DEBUG] switch_core_session.c:905 Send signalsofia/internal/1000(a)54.245.109.196[BREAK] 2012-11-21 00:40:31.538130 [DEBUG] switch_core_session.c:821 Send signal sofia/internal/sip:1001@75.83.124.194:35948 [BREAK] 2012-11-21 00:40:31.538130 [DEBUG] switch_core_session.c:821 Send signalsofia/internal/1000(a)54.245.109.196[BREAK] 2012-11-21 00:40:31.538130 [DEBUG] switch_core_session.c:905 Send signalsofia/internal/1000(a)54.245.109.196[BREAK] 2012-11-21 00:40:31.538130 [DEBUG] switch_core_session.c:905 Send signalsofia/internal/1000(a)54.245.109.196[BREAK] 2012-11-21 00:40:31.558132 [DEBUG] sofia.c:6285 Channelsofia/internal/1000(a)54.245.109.196 entering state [ready][200] 2012-11-21 00:40:33.198130 [INFO] switch_rtp.c:3579 Auto Changing port from 10.0.1.15:7076 to 75.83.124.194:37261 2012-11-21 00:41:07.718130 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/sip:1001@75.83.124.194:35948 [BREAK] 2012-11-21 00:41:07.738130 [DEBUG] switch_channel.c:2950 (sofia/internal/sip:1001@75.83.124.194:35948) Callstate Change ACTIVE -> HANGUP 2012-11-21 00:41:07.738130 [DEBUG] switch_ivr_bridge.c:501 sofia/internal/sip:1001@75.83.124.194:35948 ending bridge by request from write function 2012-11-21 00:41:07.738130 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/internal/1000(a)54.245.109.196] 2012-11-21 00:41:07.738130 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/internal/sip:1001@75.83.124.194:35948 [BREAK] Sent from my iPhone Sent from my iPhone

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FreeSWITCH Community Conference Call Today
by Michael Collins 21 Nov '12

21 Nov '12

Hello all, Today's agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_21 We are going to do a wiki tutorial, specifically how to add a channel variable to the wiki. We'll also go over script I wrote that produces an HTML page that lists all the channel variables. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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H.264 / RTSP
by Jan Berger 20 Nov '12

20 Nov '12

hi, Have anyone considered adding support for Video streams in FreeSWITCH? Particulary RTP+RTSP supporting Mpeg4/H.264. /J

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FreeSWITCH Weekly News and Notes
by Michael Collins 19 Nov '12

19 Nov '12

Happy short week to those of you in North America! The weekly FreeSWITCH news and notes took a hiatus while I was out on a medical leave. I am happy to report that I am back to work and recovering nicely. Many thanks to those who sent their well-wishes and happy thoughts. We have a great community and I am glad to be a part of it! On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_14> we covered some Linux/FreeSWITCH install and configuration tips. A special thanks to Ken Rice for giving us some practical information on many of the useful files and utility items that are available in the FreeSWITCH source tree and how to implement them, including FreeSWITCH init scripts<http://fisheye.freeswitch.org/browse/freeswitch.git/build/freeswitch.init.r…>, Anthony's .emacs file<http://fisheye.freeswitch.org/browse/freeswitch.git/support-d/.emacs?hb=true>, and even a monit <http://mmonit.com/monit/> configuration example<http://fisheye.freeswitch.org/browse/freeswitch.git/build/freeswitch.monitr…>. I hope you found these items as useful as I did. We recently released FreeSWITCH 1.2.4 and Ken Rice tells me that more updates are in the works. More information will be available on this week's conference call <http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_21>. This week I will be presenting a Wiki how-to: adding a channel variable page<http://wiki.freeswitch.org/wiki/Variable_skeleton>. This will be especially useful because it illustrates a number of Mediawiki<http://www.mediawiki.org/> concepts. Also, we have a lot of missing channel variables so if everyone picks one or two to add we'll be able to expand the wiki coverage. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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