February 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

G729.A on ARM
by Lesley Pervis 10 Feb '14

10 Feb '14

Does the project have any plans to sell G729 codecs for AMR as well as for x86_64/i386? My company has been running a FS VoIP PA endpoint on ARM, kind of like Ken's Switch Pi, but simpler, and we'd really like to put G729 on the things, if possible.

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esl, send_dtmf, premature ESL_EVENT_CHANNEL_EXECUTE_COMPLETE
by Marcin S 10 Feb '14

10 Feb '14

Hello, In my simple C application I want to send dtmfs. This is quite easy - you just call esl_execute("send_dtmf", "1234", uuid); and it works. However, ESL returns ESL_EVENT_CHANNEL_EXECUTE_COMPLETE way to fast - even before it starts dtmfs generation! How can I know, when dtmfs were really generated? So far, the only method is to issue some sleep(), but it looks like big workaround.

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Mod_Manage won't build switch_core_port_allocator_new
by Andre 10 Feb '14

10 Feb '14

Hi, I downloaded freeswitch-1.2.19.tar.gz today and Mod_manage won't build. The code that is the problem is line 8794 in freeswitch_wrap.2010.cxx SWIGEXPORT int SWIGSTDCALL CSharp_switch_core_port_allocator_new(unsigned short jarg1, unsigned short jarg2, unsigned long jarg3, void * jarg4) { int jresult ; switch_port_t arg1 ; switch_port_t arg2 ; switch_port_flag_t arg3 ; switch_core_port_allocator_t **arg4 = (switch_core_port_allocator_t **) 0 ; switch_status_t result; arg1 = (switch_port_t)jarg1; arg2 = (switch_port_t)jarg2; arg3 = (switch_port_flag_t)jarg3; arg4 = (switch_core_port_allocator_t **)jarg4; result = (switch_status_t)switch_core_port_allocator_new(arg1,arg2,arg3,arg4); jresult = result; return jresult; } error C2664: 'switch_core_port_allocator_new' : cannot convert parameter 1 from 'switch_port_t' to 'const char *' ..\mod_managed\freeswitch_wrap.2010.cxx IntelliSense: too few arguments in function call ..\mod_managed\freeswitch_wrap.2010.cxx 8806 79 mod_managed Looking at the method switch_core_port_allocator_new this seems to of changed. How do I fix the code? SWITCH_DECLARE(switch_status_t) switch_core_port_allocator_new(_In_ const char *ip, _In_ switch_port_t start, _In_ switch_port_t end, _In_ switch_port_flag_t flags, _Out_ switch_core_port_allocator_t **new_allocator);

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Performance gain for Asterisk 12 to use pjsip at core?
by Henry Huang 09 Feb '14

09 Feb '14

Asterisk 12 has added a new sip driver using pjsip. I remember reading about the discussion why FreeSWITCH chose to use sofia instead of other sip stacks, but I can't find it anymore. So my question is, do you guys think Asterisk will be able to boost their performance by swapping out their sip core to use pjsip and be somewhat on par with FreeSWITCH performance wise? Thanks, Henry

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FreeSwitch Proxy + RTPProxy Media server
by adahary 09 Feb '14

09 Feb '14

I would like to integrate FS with RTPproxy like openSips and kamailio are well integrated with it. FS should handle the SIP signaling and the RTPproxy should relay the RTP stream from A to B: A.sip <=> FS <=> B.sip FS = PASS-THRU A.rtp <=> RTPproxy <=> B.rtp. I understand that FS should ask the RTPproxy to allocate UDP ports for both endpoint and then pass-thru-bridge them to cummunicate directly through the RTPproxy. What I cannot yet figure out is how replace both A and B ip/port sets in SDP with the RTPproxy ip/port-udp. I'v read about switch_r_sdp (Leg.A) and switch_m_sdp (Leg.B !?) but couldn't figure it out. how should I use switch_r_sdp (or/and switch_m_sdp) in my dialplan when when A calls B? Tried the following with no help. <action application="set"> </action> <action application="set"> </action> Any tip/help/advise will be appriciated. Regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Med… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch not putting external IP in SD and From Header when dialing out SIPURI by external profile
by Andrew Paul 09 Feb '14

09 Feb '14

Hi all, ------------------FREESWITCH--------------(bridge=sofia/external/sip:outbound-opensip.com:5060 )-----------OPENSIP---------------------- I am facing NAT problem when i am dialing SIP URI via sip profile. When i am dialing the SIP uri with Gateway trunk NAT is working properly. Dialing SIP URI <action application="bridge" data="sofia/external/sip: outbound-opensip.com:5060" In external profile setting I made the setting for ext-rtp-ip and ext-sip-ip. In freeswitch cli "sofia status profile external" i am able to see the external rtp and sip ip correctly. But when freeswitch sending INVITE external IPs are not set in SDP and FROMHeader, instead it is using local IP. When i tried to debug the problem in switch_core_media.c freeswitch failed to check the NAT settings. switch_core_media.c SWITCH_DECLARE(switch_status_t) switch_core_media_choose_port(switch_core_session_t *session, switch_media_type_t type, int force) /* Check if NAT is detected */ if (!zstr(smh->mparams->remote_ip) && switch_core_media_check_nat(smh, smh->mparams->remote_ip)) { /* Yes, map the port through switch_nat */ /* Code */ } else { /* No NAT traversal required, use the profile's rtp ip */ use_ip = smh->mparams->rtpip; } Here remote IP is NULL always when i am dialing URI with external profile. When I tried to make call by "sofia/gateway/trunk/sip: outbound-opensip.com:5060" i can see the remote ip as "oubound-opensip.com" and its able to external IP in SDP. Can any body tell how to solve this problem ? Is there any configuration to set relam in dialplan ? I am using freeswitch version 1.5.6. Hopping the earliest response. Regards Andrew

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Re: [Freeswitch-users] Hold Music not Playing when pressing Hold on Polycom Phone (Lyle Pratt)
by Lyle Pratt 08 Feb '14

08 Feb '14

An update on why this was happening so others will know: I disabled the Sip ALG on my router and everything now seems to be working fine. So, hold music will not work on some Polycom phones behind NAT with ALG connecting to remote Freeswitch servers. Thanks, Lyle Pratt On Thu, Feb 6, 2014 at 7:04 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. Re: routing incoming SMS? (Privus P) > 2. Re: Filler Music in Call Recordings (Michael Jerris) > 3. Re: Hold Music not Playing when pressing Hold on Polycom > Phone (Michael Jerris) > > > ---------- Forwarded message ---------- > From: Privus P <privus007(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 6 Feb 2014 12:58:35 +0000 > Subject: Re: [Freeswitch-users] routing incoming SMS? > Hi Donny, > > Perhaps you could share your javascript and lua script so that we could > try to figure out together what isn't working. > Judging by the apparent lack of response to this thread, it seems to be > the best way forward to try and resolve this. > > > On Thu, Feb 6, 2014 at 6:50 AM, Donny Hardyanto <hardyanto.donny(a)gmail.com > > wrote: > >> Hi, >> >> I have similar problem. I use gateway to GSM provider that support SIP >> simple . So far i can send sms from fs to gateway and receive sms in fs >> from gateway. But I was lost how to send the incoming sms from gateway to >> client. >> >> Also my client support sip simple also. When the sms from client arrive >> at fs, i dont know how to forward them to gateway. >> >> I use Javascript heavily and tried to produce the same thing as lua >> script in javascript but it always fail. The event fire command does not >> produce any thing. >> >> Can any one show us or point us how to forward sms? >> >> Dinny >> On Feb 4, 2014 8:06 PM, "Peter Villeneuve" <petervnv1(a)gmail.com> wrote: >> >>> Hi, >>> >>> Thanks for helping out. Indeed I would love to try out a lua script to >>> handle SMS. >>> Can anyone point to a sample script that I can study and play with? >>> >>> Any help is much appreciated. I'm still stuck with SMS hitting FS but >>> not being forwarded to the peer.... >>> >>> Thanks, >>> Peter >>> >>> >>> On Sun, Feb 2, 2014 at 5:46 PM, Privus P <privus007(a)gmail.com> wrote: >>> >>>> Hi Peter, >>>> >>>> I'm not really sure why your SMS isn't being routed correctly. >>>> I'm sure others in this list have a lot more experience and can help >>>> you out, but have you thought of using a lua script to handle SMS instead >>>> of relying on the send action? >>>> >>>> >>>> >>>> >>>> On Sat, Feb 1, 2014 at 2:00 PM, Peter Villeneuve <petervnv1(a)gmail.com>wrote: >>>> >>>>> Hi, >>>>> >>>>> I'm experimenting with GSMopen and I'm having trouble doing something >>>>> which is likely pretty simple. >>>>> I have enabled mod_sms and GSMopen in FS. I can send SIP simple >>>>> messages between 2 registered peers OK. >>>>> I can see that incoming GSM SMS messages arrive in FS as expected, but >>>>> I can't seem to get them converted into SIP SIMPLE format and routed to the >>>>> peer (1000). >>>>> Once they arrive in FS, I can see in the log: >>>>> >>>>> 2014-01-30 20:00:54.840999 [INFO] mod_sms.c:336 Processing text >>>>> message +4412398746->gsm01 in context default >>>>> Chatplan: gsm01 parsing [default->basic p2p] continue=true >>>>> Chatplan: gsm01 Regex (PASS) [basic p2p] to(gsm01) =~ /^(.*)$/ >>>>> break=on-false >>>>> Chatplan: gsm01 Action send() >>>>> >>>>> >>>>> My chatplan has: >>>>> >>>>> <?xml version="1.0" encoding="utf-8"?> >>>>> <include> >>>>> <context name="default"> >>>>> >>>>> <extension name="basic p2p" continue="true"> >>>>> <condition field="to" expression="^(.*)$"> >>>>>  >>>>> <action application="send"/> >>>>> </condition> >>>>> </extension> >>>>> >>>>> </context> >>>>> </include> >>>>> >>>>> And in my gsmopen.conf.xml: >>>>> >>>>> <configuration name="gsmopen.conf" description="GSMopen Configuration"> >>>>> <global_settings> >>>>> <param name="debug" value="8"/> >>>>> <param name="dialplan" value="XML"/> >>>>> <param name="context" value="default"/> >>>>> <param name="hold-music" value="$${moh_uri}"/> >>>>> <param name="destination" value="1000"/> >>>>> </global_settings> >>>>> >>>>> >>>>> I believe I'm doing something wrong in the chatplan. Is there any >>>>> transfer action like in the regular XML dialplan? >>>>> I tried creating in the default dialplan the following, hoping that it >>>>> would be similar to sofia calls and transfer the incoming SMS to peer 1000, >>>>> but no dice. >>>>> >>>>> <include> >>>>> <extension name="sms_inbound"> >>>>> <condition field="destination_number" expression="^(gsm01)$"> >>>>> <action application="transfer" data="1000 XML default"/> >>>>> </condition> >>>>> </extension> >>>>> </include> >>>>> >>>>> So, basically, how can I route the incoming SMS to peer 1000? >>>>> >>>>> Thanks >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >> _________________________________________________________________________ >> 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 >> >> > > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 6 Feb 2014 08:02:03 -0500 > Subject: Re: [Freeswitch-users] Filler Music in Call Recordings > You would have to make a media bug similar to the one for record session, > that has additional api commands to target the session by uuid that changes > a flag in a private struct of weather the recording is paused or not, and > then inside the callback for the recording, trigger different behavior when > its paused. As there is no audio being muxed, there is no reason at all to > use mod_conference. > > On Feb 5, 2014, at 7:42 PM, JP <jaykris(a)gmail.com> wrote: > > I'm trying to implement a feature that allows a user to selectively pause > and resume a call recording via FreeSWITCH's event socket interface. The > audio file should contain filler music for the duration of each pause > operation. Is there any simple straight forward way to accomplish this? I > would prefer avoiding the overhead incurred by using the conferencing > module for this if possible. > > > > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 6 Feb 2014 08:04:04 -0500 > Subject: Re: [Freeswitch-users] Hold Music not Playing when pressing Hold > on Polycom Phone > I don't see the phone actually sending the hold signal there, try turning > on sip trace and confirming what the phone is actually sending. Compare > that to your dev environment where it works. > > Mike > > On Feb 6, 2014, at 12:11 AM, Lyle Pratt <lylepratt(a)gmail.com> wrote: > > Hi Guys, > > When I press the HOLD button (or start to transfer someone, join a > conference, etc) the configured hold music will not play. > > I previously had this working in production, but all of the sudden it is > no longer working I can't figure out why. It works fine in my development > environment on my local network, but not when I'm registering to Public > IPs. I first thought it was a Polycom specific NAT issue, but then I tried > it with a different phone (Cisco) from a different network with no success > either. > > Here is a Pastebin of the Freeswitch console logs when I press the hold > button on my device and no hold music occurs. > http://pastebin.com/p6xtwyz7 > > Does anyone have any ideas on how on earth I messed this up? Its the last > piece of the puzzle I have in completely getting this working! Any help > appreciated > > > _______________________________________________ > 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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FSClient Compile
by Nyamul Hassan 08 Feb '14

08 Feb '14

Hi, Is this the right forum to ask about FSClient compile problems? If not, can you please guide me to the forum which is? If yes, I would be grateful if you could help me out here. I downloaded FreeSwitch and the build on VS2012 completed successfully (without mod_managed, mod_cephstral, mod_v8, mod_v8_skel & libv8). Whenever I start compiling FSClient, I get the following error: Error 1 The command "if not exist ..\..\"external items\Release\" md ..\..\"external items\Release\" if exist "%FREESWITCH_SRC_LOCATION%" copy "%FREESWITCH_SRC_LOCATION%"\Win32\Release\mod\FreeSWITCH.Managed.dll ..\..\"external items\Release\"" exited with code 1. FSClient Could you please help with what I am doing wrong? I am following the instructions on http://wiki.freeswitch.org/wiki/FSClient. I must be misunderstanding / missing something. Regards HASSAN

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Filler Music in Call Recordings
by JP 08 Feb '14

08 Feb '14

I'm trying to implement a feature that allows a user to selectively pause and resume a call recording via FreeSWITCH's event socket interface. The audio file should contain filler music for the duration of each pause operation. Is there any simple straight forward way to accomplish this? I would prefer avoiding the overhead incurred by using the conferencing module for this if possible. Thanks -JP

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jitter issue with FreeSWITCH Version 1.2.10
by Yuriy Nasida 08 Feb '14

08 Feb '14

Hello, We have noted jitter issue with this version of FS. The only ERR i see in logs now is:debce20e-eb3e-4388-89df-07e412000777 2014-02-06 14:25:21.397009 [ERR] switch_core_session.c:2762 Error sofia/external/dest_num@IP:5060 too many stacked extensionsNot sure that this means exactly Problem can be temporary solved by means of restarting of FS but after some time this happens again. I switch all calls from FS node before restating and also noted that FS eats ~10% CPU without any calls. FreeSWITCH Version 1.2.10+git~20130521T182905Z~b73e2c51bb (git b73e2c5 2013-05-21 18:29:05Z) In my opinion looks like that issue connected with this version of FS. Please advice. Thanks.

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