August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Can not make out going call from sangoma A200
by James zhu 13 Aug '13

13 Aug '13

hello:I installed freeswitch and sangoma A200 with 2 FXOs. i set the tone group to CNand opermode=china. the A200 can make accept incoming calls from FreeTDM/2:1 , but I use same port to make out going port, the shows the port is CONGESTION.--------------------------system log-----------------------------------------------2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:640 [2:1] FreeTDM/2:1/13811737925 CHANNEL HANGUP EXIT2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:48 FreeTDM/2:1/13811737925 Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:679 (FreeTDM/2:1/13811737925) State HANGUP going to sleep2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:692 (FreeTDM/2:1/13811737925) Callstate Change DOWN -> HANGUP2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:447 (FreeTDM/2:1/13811737925) State Change CS_HANGUP -> CS_REPORTING2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_REPORTING2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:92 FreeTDM/2:1/13811737925 Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING going to sleep2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:441 (FreeTDM/2:1/13811737925) State Change CS_REPORTING -> CS_DESTROY2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1549 Session 37 (FreeTDM/2:1/13811737925) Locked, Waiting on external entities2013-08-02 14:36:42.023531 [DEBUG] switch_ivr_originate.c:3632 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [INFO] mod_dptools.c:3192 Originate Failed. Cause: NORMAL_CIRCUIT_CONGESTION2013-08-02 14:36:42.023531 [NOTICE] switch_channel.c:4593 Hangup sofia/internal/1008(a)192.168.0.173 [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [DEBUG] switch_channel.c:3135 Send signal sofia/internal/1008(a)192.168.0.173 [KILL]2013-08-02 14:36:42.023531 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal---------------------------------------------the FS is download from git and wanpipe is latest version. please give a help for that issue. Best regards, James.zhu website: www.hiastar.com

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FreeSWITCH Weekly News and Notes - ClueCon 2013 Edition
by Michael Collins 12 Aug '13

12 Aug '13

Hello all! We've survived another ClueCon <http://www.cluecon.com/> season - and ClueCon 2013 was brilliant! We had 29 talks, 5 lightning talks, and a security BoF that featured Philip Zimmermann. Ken Rice has posted some pictures here<https://plus.google.com/115131822208898699412/posts/WjwVZxShGK5>. We are currently gathering all of the presentation files and YouTube links. Once they are collected they will be available from the presentations pages. Just click on the title of a presentation to see the download links. This week Ken and I, along with any other ClueCon attendees, will be doing a recap of some of the things that took place at ClueCon 2013. If you were at ClueCon 2013 and have an experience you'd like to share with the group then please join us on this week's conference call<https://wiki.freeswitch.org/wiki/FS_weekly_2013_08_14> . Thanks to everyone for making this year's event the best one yet. We look forward to 2014! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Weekly FreeSWITCH conference call
by Ken Rice 12 Aug '13

12 Aug '13

Hey Guys, Well Last week, we took a week off for ClueCon 2013 and it was great! If you missed it, theres things popping up all over social media. However we now have about 50 weekly Episodes of the FreeSWITCH Community conference call to fill! Want to come on and talk about your project? Email Me! Have a suggestion for someone to get on the call to talk about their project? Email Me! Do you use FreeSWITCH at your job and you want to show us what cool thing you have done? Email Me! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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How to create a single outbound ESL socket at freeswitch startup.
by julien terrasson 12 Aug '13

12 Aug '13

Hi, I would like to create a single outbound ESL socket that would establish the connection with the ESL server at freeswitch startup. This connexion would remains up and would used to transfer events from calls going through this server. I managed to do this with an inbound ESL but i would prefer doing it with an outbound ESL (it is important for me that freeswitch controls the establishement of the connection). Does anybody know if that's possible to start a common ESL connection (that would push event from all call) at FS startup ? If yes, where should it be provisionned (from the dialplan? in a startup script ?) Hope somebody can help.. Regards, Julien

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Conference for Flash and WebRTC users
by pablo platt 12 Aug '13

12 Aug '13

Hi, I'm currently using a RTMP server for audio conference. There are no performance or latency issues because there is no encryption, transcoding or mixing involved. A 1GB VPS server can handle more than 100 users without a problem. I'm trying to evaluate FreeSWITCH so I'll be able to support both Flash and WebRTC users in a conference. My use case is two types of conferences: - A conference with up to 5 participants all speaking. - A conference with 20-30 participants with 2-3 speakers. What part require more CPU? - Encryption (DTLS-SRTP) - Transcoding - Mixing Does FS mix a channel separately for each participant or can it reuse the same mix? For example, if I have 1 speaker and 20 listeners, 10 with speex and 10 with opus, does FS produce 2 streams or 20? Does FS exclude silent participants from the mix to save CPU? Thanks

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How to create a single outbound ESL socket at freeswitch startup.
by Chris Cachor 12 Aug '13

12 Aug '13

Julien, Unfortunately I've run into the same problem. Having an outbound ESL connection to push event data to a ESL server is ideal, especially in a multi-server environment (simple config setup). However, if you read the documentation, the intended behavior of the outbound ESL socket is to give control of the call to the ESL server/application. If that's your implementation, then you could simply add the socket application to your dial plan. Otherwise, an inbound socket connection is what you're looking for. - Chris On Aug 12, 2013, at 9:46 AM, freeswitch-users-request(a)lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. How to create a single outbound ESL socket at freeswitch > startup. (julien terrasson) > 2. uuid_displace: change loudness level of audio > (Melanie Treitinger) > 3. Conference for Flash and WebRTC users (pablo platt) > 4. Re: Error in launching fs_cli (Ashish Mishra) > 5. Re: garbled audio with G726-32, other codecs are fine > (Michael Jerris) > 6. Re: Error in launching fs_cli (Steven Ayre) > 7. Re: Issue with JSSIP + Freeswitch (Michael Jerris) > > From: julien terrasson <julien.terrasson(a)gmail.com> > Subject: [Freeswitch-users] How to create a single outbound ESL socket at freeswitch startup. > Date: August 12, 2013 2:01:37 AM CDT > To: freeswitch-users(a)lists.freeswitch.org > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Hi, > > I would like to create a single outbound ESL socket that would establish the connection with the ESL server at freeswitch startup. > This connexion would remains up and would used to transfer events from calls going through this server. > I managed to do this with an inbound ESL but i would prefer doing it with an outbound ESL (it is important for me that freeswitch controls the establishement of the connection). > Does anybody know if that's possible to start a common ESL connection (that would push event from all call) at FS startup ? If yes, where should it be provisionned (from the dialplan? in a startup script ?) > > Hope somebody can help.. > > Regards, > > Julien > > > > > > > > > From: Melanie Treitinger <treitinger(a)as-infodienste.de> > Subject: [Freeswitch-users] uuid_displace: change loudness level of audio > Date: August 12, 2013 3:33:36 AM CDT > To: freeswitch-users(a)lists.freeswitch.org > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Hi Michael, > > I think I'll try sox, thanks. > > > Melanie > > > > From: pablo platt <pablo.platt(a)gmail.com> > Subject: [Freeswitch-users] Conference for Flash and WebRTC users > Date: August 12, 2013 2:40:17 AM CDT > To: freeswitch-users(a)lists.freeswitch.org > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Hi, > > I'm currently using a RTMP server for audio conference. > There are no performance or latency issues because there is no encryption, transcoding or mixing involved. A 1GB VPS server can handle more than 100 users without a problem. > > I'm trying to evaluate FreeSWITCH so I'll be able to support both Flash and WebRTC users in a conference. > > My use case is two types of conferences: > - A conference with up to 5 participants all speaking. > - A conference with 20-30 participants with 2-3 speakers. > > What part require more CPU? > - Encryption (DTLS-SRTP) > - Transcoding > - Mixing > > Does FS mix a channel separately for each participant or can it reuse the same mix? > For example, if I have 1 speaker and 20 listeners, 10 with speex and 10 with opus, does FS produce 2 streams or 20? > > Does FS exclude silent participants from the mix to save CPU? > > Thanks > > > > > > > From: Ashish Mishra <itsme.kunnu(a)gmail.com> > Subject: Re: [Freeswitch-users] Error in launching fs_cli > Date: August 12, 2013 3:51:45 AM CDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > I reinstalled freeswitch but still i am getting the same error : > fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] mod_socket is also loaded...i have also flushed the firewall rules...pls help > > On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu(a)gmail.com> wrote: > When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i have installed freeswitch) it gives me the following error : > fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] > > > > From: Michael Jerris <mike(a)jerris.com> > Subject: Re: [Freeswitch-users] garbled audio with G726-32, other codecs are fine > Date: August 12, 2013 9:14:45 AM CDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > There is always bit packing, but there are 2 different ways to do the bit packing. A lot of devices get it wrong so its worth looking at that. > > Mike > > On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev(a)c3i.bg> wrote: > >> Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 >> ; so bitpacking shouldn't be used. By the way when I tested with AAL2 to >> the linphone client I only got cracks and whitenoise, I've forgot to >> mention that in my post. >> >> That said I've tried to uncomment and set <param name="bitpacking" >> value="none"/> in internal.xml ("none" is a wild guess - I couldn't find >> any doc on values accepted by this parameter), but that doesn't help. >> >> Speex: the ATAs don't support it. And being stubborn I'd like to >> understand what's wrong with G726 :) >> >> >> On 08/04/2013 05:22 AM, Jeff Leung wrote: >>> >>> You can turn off G726 AAC bit-packing in spandsp.conf.xml. >>> >>> By the way, there are other codecs out there you can try. SPEEX comes >>> to mind if all your endpoints don’t deal with the PSTN. >>> >>> *From:*freeswitch-users-bounces@lists.freeswitch.org >>> [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of >>> *Brian Foster >>> *Sent:* Saturday, August 3, 2013 2:37 PM >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other >>> codecs are fine >>> >>> AAC bitpacking by any chance? I thought I had a similar issue, >>> happened so long ago I cant remember what I did. >>> >>> Thank you, >>> >>> Brian Foster >>> Project Manager/Owner's Rep. >>> Davri Investments, Inc. >>> O: 317-787-2686 x2102 >>> M: 317-600-9753 >>> E: bdfoster(a)davri.com <mailto:bdfoster@davri.com> >>> Indianapolis, Indiana >>> >>> Sent from a mobile device. >>> >>> On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev(a)c3i.bg >>> <mailto:imitev@c3i.bg>> wrote: >>> >>> Hello >>> >>> I'm migrating an office setup from asterisk to FS and in the process I >>> was considering using G726-32 for some bandwidth starved remote >>> endpoints. However I only get metallic/garbled audio with that codec >>> even when simply playing moh to the endpoint, while other codecs work >>> fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds >>> marginally better but still garbled and really worse than G711. >>> >>> The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest >>> (centos6 64bit host). But please don't shoot ! :) - I know about virtual >>> environment limitations but for these tests the host is only lightly >>> loaded, there aren't any calls to the FS instance except my tests, and >>> the fact that it works with other codecs makes me think that >>> virtualization is not the issue here. I may be wrong though. >>> >>> Is there any guide for debugging that kind of problem before reverting >>> to a fresh install on bare-metal with the latest HEAD ? Until now I've >>> tried: >>> >>> - improving timers ; but the default soft timer (which I guess uses >>> timerd) works best. The time interval between sent packets on a tcpdump >>> trace looks identical to the output of "timer_test", so that doesn't >>> seem to be a network/jitter problem. And there's no problem with other >>> codecs, but maybe G726-XX is specific. For info the guest's clocksource >>> is kvm_clock, while the host uses tsc. >>> >>> - using different endpoints: the production ones are Linksys PAP2 >>> ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the >>> same thing happens with linphone on a fedora 19 laptop. >>> >>> A call with rtp media going through FS without transcoding - G726-32 to >>> G726-32 - works perfectly (I can't hear the difference with G711). The >>> problem is only when there's transcoding to G726 (from wav for moh, or >>> from any other codec when bridging). I've looked at the wiki, posts, >>> changelogs, jira, ..., but am a bit at a loss now. >>> >>> Any pointers ? >>> >>> Except that little problem, FS rocks, and I'm happy I can finally ditch >>> asterisk. Kudos to the core devs and contributors. >>> >>> Ivan >>> > > > > > > > From: Steven Ayre <steveayre(a)gmail.com> > Subject: Re: [Freeswitch-users] Error in launching fs_cli > Date: August 12, 2013 9:44:38 AM CDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Have you checked the logfile and netstat? > > > On 12 August 2013 09:51, Ashish Mishra <itsme.kunnu(a)gmail.com> wrote: > I reinstalled freeswitch but still i am getting the same error : > fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] mod_socket is also loaded...i have also flushed the firewall rules...pls help > > On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu(a)gmail.com> wrote: > When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i have installed freeswitch) it gives me the following error : > fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > From: Michael Jerris <mike(a)jerris.com> > Subject: Re: [Freeswitch-users] Issue with JSSIP + Freeswitch > Date: August 12, 2013 9:45:46 AM CDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > We only use non rfc-1918 ip's by default. If you want to use 1918 ip's you need to tweak acls > > Mike > > > On Aug 5, 2013, at 8:24 AM, Shahrzad A. <shahrzad.aziminia(a)gmail.com> wrote: > >> Hi everyone >> >> I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS with the latest version of Openssl ('OpenSSL 1.0.1e 11 Feb 2013') >> I'm using the default configuration and just uncommentated the ' <param name="ws-binding" value=":5066"/> ' in internal.xml in order to have the support for webrtc. >> As the client I'm having JSSIP, the latest version with the adjustment to have (DtlsSrtpKeyAgreement:true). Both my clients and server located in the same network without having any firewalls therefore I'm not using any STUN server. I can make call but there is no sound! and after setting debig_level 10 and dial from extension 1003 to 5000 I'm having the following Alert in console and no sound: >> >> [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) >> >> its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) >> If I call from extension 1000 which is a SIP client set on my iPhone and call 1003 I'm having the following alerts but voice on both JSSIP client and SIP client: >> >> 2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/1000@10.0.14.16:5060 timer while HOT >> 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/1000@10.0.14.16:5060 Hot Hit 1 >> >> And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts: >> >> 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/sip:1000@10.0.14.182:5065 Hot Hit 4 >> 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/sip:1000@10.0.14.182:5065 timer while HOT >> 2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) >> >> If I use bypass media or proxy media I will have voice on both JSSIP clients but cant ring any SIP clients and again no voice if I call 5000! >> >> Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP? >> >> Thanks in advanced! >> >> Sherry > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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uuid_displace: change loudness level of audio
by Melanie Treitinger 12 Aug '13

12 Aug '13

Hi, I want to use uuid_displace to play announcements while someone is listening to music on hold. This works fine, except that the music on hold is too loud - it should be reduced when the uuid_displace takes place. Is there a way to achieve this?

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WebRTC sipml5 client hangs up after 120 seconds on bridged calls to sip phone
by Karsten Horsmann 11 Aug '13

11 Aug '13

Hello, i play around with freeswitch master on centos 6.4 and webrtc to sip-phones. if i call from the browser (chrome) to a snom, the call ends after 120 seconds. I see that the sipml5 client is sending then a bye message. The other direction works fine. From snom to the webrtc phone, no hangup after 120 seconds. And from webrtc to the moh extension works fine too. Any special settings to resolve that? I made an sofia trace, both phones are on the same sip-profile. http://pastebin.freeswitch.org/21291 my git version is d1268e81036ce8ce00de8ee22f387cdbf43a7203 -- Kind Regards Mit freundlichen Grüßen *Karsten Horsmann*

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Tech Prefix with Gateway
by Andre 11 Aug '13

11 Aug '13

Hi, I'm trying to pass a Tech Prefix while using a gateway. When I add the prefix in front of the phone number I get a routing error back. sofia/gateway/myGateway/prefix + Phone Number sofia/gateway/myGateway/211713015551212 So, what is the proper way to pass the Tech Prefix? Thanks Andre

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How to check who is attacking my fs server?
by Jessica Wang 11 Aug '13

11 Aug '13

Dear all, My FS server got attacked by someone. Unknown external user 107 is frequently calling strange number on my FS server. From the log, I can not find the real IP of this external user, the IP logged is my server's IP! How can check who is doing this and block any similar action? Part of the log is as below. Thanks. 2013-08-11 09:48:12.401286 [NOTICE] switch_channel.c:951 New Channel sofia/external/107@(my fs server ip addr) [c1499822-0262-11e3-af23-2d4b950c464a] 2013-08-11 09:48:12.421101 [INFO] mod_dialplan_xml.c:485 Processing 107 <107>->+970598610587 in context public

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