November 2012 - freeswitch-dev@lists.freeswitch.org

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-dev] 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-dev] 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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[Freeswitch-dev] 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-dev] 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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[Freeswitch-dev] Gateway Presence Subscription Storage
by Jerry Richards 17 Nov '12

17 Nov '12

I added the XML to the gateway.xml file for one of my sip_profiles so that it subscribes to presence (and it works fine). My question is, I cannot find where in the sqlite3 DB that Freeswitch stores the subscription data (the sip_subscriptions table is empty). Is this subscription only saved/maintained in RAM on the heap? <subscriptions name="x"> <subscription event="presence"> <param name="retry-seconds" value="50"/> <param name="expire-seconds" value="60"/> <param name="content-type" value="application/pidf+xml"/> </subscription> </subscriptions> Thanks, Jerry

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[Freeswitch-dev] Friday Free-for-all!
by Michael Collins 16 Nov '12

16 Nov '12

Come join us! We're having a nice chat in the main conf room -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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[Freeswitch-dev] FreeSWITCH Conf Call Today
by Michael Collins 14 Nov '12

14 Nov '12

Hello All! Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_14 Ken Rice and a few of our FS long time users will be presenting some tips and tricks that you may not be aware of or may have forgotten about. 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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[Freeswitch-dev] Patch adding system call support to mod_sms
by Kurtis Heimerl 12 Nov '12

12 Nov '12

We needed to make system calls from a mod_sms chatplan, here's a patch supporting that feature written by my colleague Shaddi Hassan.

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[Freeswitch-dev] About Rxfax function of Freeswitch
by jun xie 09 Nov '12

09 Nov '12

HI, every body. Recently,i tested a case to use FreeSwitch to receive T38 fax from a As5400 (Cisco voice GW).According with Google and Freeswitch.org useful information,the procession is smooth.Rxfax could receive almost all papers. But the fax images received have some problems,that's the images were all re sized,were shorter and broader. I will show you one,it should be Fax paper type's problem.but i don't know how to revise.Anybody can give me some suggestions? thanks a lot.

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