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Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.
by Ashley B 03 Jul '13

03 Jul '13

I was experiencing the same problem as Henry. Ubuntu 12.04 has an earlier version version of OpenSSL and when I upgrade OpenSSL prior to installing FS as per the following snippet: # prerequisites sudo apt-get install git-core build-essential autoconf automake libtool libncurses5 libncurses5-dev gawk libjpeg-dev zlib1g-dev pkg-config libssl-dev -y # requires openssl v1.0.1e for WebRTC dtls wget http://www.openssl.org/source/openssl-1.0.1e.tar.gz tar -xvzf openssl-1.0.1e.tar.gz cd openssl-1.0.1e/ ./config --prefix=/usr make make install ... I get the following error: quiet_libtool: link: gcc -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -I/usr/local/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -pthread -D_REENTRANT -D_GNU_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -o .libs/freeswitch freeswitch-switch.o -lm -lz ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lncurses -ljpeg -pthread -Wl,-rpath -Wl,/usr/local/freeswitch/lib ./.libs/libfreeswitch.so: undefined reference to `SSLv2_client_method' collect2: ld returned 1 exit status make[1]: *** [freeswitch] Error 1 make: *** [install] Error 2 [1]+ Exit 2 make Cannot even build FS now. Ashley. From: Michael Jerris [mailto:mike@jerris.com] Sent: Wednesday, July 3, 2013 9:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA. yes. you can see it working on the demo at http://webrtc.freeswitch.org. On Jul 3, 2013, at 12:53 AM, Henry Huang <red.rain.seven(a)gmail.com <mailto:red.rain.seven@gmail.com> > wrote: Anyone getting audio working bridging call to webrtc client without bypassing media? On Jul 2, 2013 6:45 PM, "João Mesquita" <jmesquita(a)freeswitch.org <mailto:jmesquita@freeswitch.org> > wrote: I think you might need to re bootstrap maybe? On Jul 2, 2013 8:54 PM, "Henry Huang" <red.rain.seven(a)gmail.com <mailto:red.rain.seven@gmail.com> > wrote: I did, and just in case I ran it again. ./configure make make install restart freeswitch This is what I am still seeing. 2013-07-02 16:40:51.135302 [ALERT] switch_core_media.c:1310 Dispatched RTCP event 2013-07-02 16:40:51.175310 [ALERT] switch_rtp.c:3883 Drop audio packet 70 bytes (dtls not ready!) b=4294967168 2013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1510 Setting RTCP src-1 to 10.128.7.103 2013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1515 Setting RTCP src-1 LENGTH to 12 (1280, 10.128.7.103) 2013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1516 Setting msw = -1267375797, lsw = -601115572 2013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1517 now = 0, now lo = 0, now hi = 0 Henry On Tue, Jul 2, 2013 at 3:13 PM, Ken Rice <krice(a)freeswitch.org <mailto:krice@freeswitch.org> > wrote: Did you reconfigure? On 7/2/13 4:29 PM, "Henry Huang" <red.rain.seven(a)gmail.com <http://red.rain.seven@gmail.com/> > wrote: Ok, I have upgraded openssl to 1.0.1e, recompiled FreeSWITCH, but I am still getting the "dtls not ready!" alerts and no audio 2013-07-02 14:26:47.695312 [ALERT] switch_rtp.c:4408 sofia/internal/1004(a)10.128.7.103 <http://sofia/internal/1004@10.128.7.103> Hot Hit 2 2013-07-02 14:26:47.695312 [ALERT] switch_rtp.c:4425 sofia/internal/1004(a)10.128.7.103 <http://sofia/internal/1004@10.128.7.103> timer while HOT 2013-07-02 14:26:47.715313 [ALERT] switch_rtp.c:5443 Skip sending audio packet 172 bytes (dtls not ready!) Here is my openssl version: root@fusionpbx:/usr/src# openssl version OpenSSL 1.0.1e 11 Feb 2013 On Tue, Jul 2, 2013 at 9:35 AM, Michael Collins <msc(a)freeswitch.org <http://msc@freeswitch.org/> > wrote: You'll need to make sure you have OpenSSL 1.0.1e which was released in Feb of this year. I suspect the version in Ubuntu might not be older. -MC On Tue, Jul 2, 2013 at 9:02 AM, Henry Huang <red.rain.seven(a)gmail.com <http://red.rain.seven@gmail.com/> > wrote: Michael, I am just using whatever comes with Ubuntu 12.04 64bit. Package: openssl Version: 1.0.1-4ubuntu5.8 Depends: libc6 (>= 2.15), libssl1.0.0 (>= 1.0.1) Thanks, Henry _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <http://consulting@freeswitch.org/> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <http://FreeSWITCH-users@lists.freeswitch.org/> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org <http://www.freeswitch.org/> -- Ken http://www.FreeSWITCH.org <http://www.freeswitch.org/> http://www.ClueCon.com <http://www.cluecon.com/> http://www.OSTAG.org <http://www.ostag.org/> irc.freenode.net <http://irc.freenode.net/> #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@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 <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

2 1

Video codec transcoding
by alberto Villa 03 Jul '13

03 Jul '13

Hello, I'm trying to setup a simple video call between two voip phones. VoIP phone A (pjsip VoIP phone) uses H263-1998 video codec, VoIP phone B (3cx VoIP phone) uses H263 video codec. The audio call is properly set up but the video stream isn't understood by the two voip phone, probably because the two VoIP phone do not understand each other. In a default installation, is FS supposed to do any codec transcoding? thanks to everyone -- Dr. Villa Alberto Sw Engineer SeleTech srl via Collodi, 8 20052 Monza (MI) tel: +39 039 5962000 fax: +39 039 9716905 email: a.villa(a)seletech.com web: www.seletech.com www.seletech.eu

2 2

Regarding Freeswicth Nat
by johnthan123 03 Jul '13

03 Jul '13

Hi All, I am really having Hard time with NAT, Can any one give me the steps witch already works for them, its Help for Many people who is having issue with NAT. Thanks in Advance.

7 12

Originating a call, caller/destination confusion
by Rafal Gwizdala 03 Jul '13

03 Jul '13

Apart from the previous 'originate' problem I have another question related to same scenario: I'm calling a mobile phone through a 'freeconet.pl' gateway and i'm connecting this call to local 6668 extension which is an mod_httapi IVR service: originate {call_timeout=60,origination_uuid=a32f6358-cce8-423d-8938-6325e57ba992}sofia/gateway/freeconet/48602415455 6668 When this succeeds mod_httapi makes a request to my application with parameters listed below. But I'm wondering why my destination number is reported as the caller id? Caller-Direction=outbound Caller-Caller-ID-Name=Outbound Call Caller-Caller-ID-Number=48602415455 Caller-Destination-Number=6668 It looks like 48602415455 called the 6668 extension, but why? The situation is quite opposite, 48602415455 is the destination number. Is this the expected behavior, or maybe I should use a different method of initiating this call? Thanks RG 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| Accept=*/* 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| Host=89.75.189.46 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| User-Agent=mod_httapi/1.0 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;session_id=a32f6358-cce8-423d-8938-6325e57ba992 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;url=http://89.75.189.46/simver/FSHttApi/GetScript 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;hostname=T1 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Direction=outbound 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Dialplan=XML 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Caller-ID-Name=Outbound Call 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Caller-ID-Number=48602415455 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Orig-Caller-ID-Number=0000000000 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Network-Addr=213.218.116.66 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Destination-Number=6668 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Unique-ID=a32f6358-cce8-423d-8938-6325e57ba992 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Source=..\..\src\switch_ivr_originate.c 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Transfer-Source=1372843556:b0fed160-54c5-45ba-8b38-99655f494749:bl_xfer:6668/default/XML 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Context=default 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-RDNIS=48602415455 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Channel-Name=sofia/external/48602415455 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Profile-Index=2 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Profile-Created-Time=1372843556726726 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Channel-Created-Time=1372843556626714 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Channel-Answered-Time=1372843556726726 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Channel-Progress-Time=1372843556696722 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Channel-Progress-Media-Time=0 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Channel-Hangup-Time=0 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Channel-Transfer-Time=0 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Channel-Resurrect-Time=0 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Channel-Bridged-Time=0 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Channel-Last-Hold=0 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Channel-Hold-Accum=0 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Screen-Bit=true 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Privacy-Hide-Name=false 2013-07-03 11:25:56.7452|Info|SIMVerification.Controllers.FSHttApiController|T33| ;Caller-Privacy-Hide-Number=false

2 1

external calls auth
by Vincent Xia 03 Jul '13

03 Jul '13

call from an unregistered UA through port:5080 cannot get through and reach a regitered UA, e.g. 123456789 calls 1001, only by uncomment <param name="accept-blind-reg" value="true"/> <param name="accept-blind-auth" value="true"/> in interal.xml in sip_sofia, can i make the external call connect. but isn't port 5080 auth free? should i make some modification in external.xml?

2 2

"bridge" command return on answered state, previously when call hung up
by BNML 03 Jul '13

03 Jul '13

Hi all, The "bridge" command executed from a python script was previously returning after the call had hung up. After upgrading, the bridge command will return once "answered" state is reached. This is creating me a trouble as I need (from the python script) to wait for the end of the call. Has this new behaviour any parameters to tune ? Do I need to open a jira on that ? Thanks you, BNML

1 1

char *exec_on_max_fail | char *exec_on_max_timeout;
by Cesar Bermudez 03 Jul '13

03 Jul '13

Hi List. How can use the exec_on_max_fail and the exec_on_max_timeout in a IVR ? Best regards.

3 4

(no subject)
by mang jose 03 Jul '13

03 Jul '13

Has anyone tried calling from freeswitch to the rtmp client? I can make outgoing calls but can't receive incoming calls, it says the number is unavailable

1 0

Re: [Freeswitch-users] ReINVITE failure with ACK+SDP scenario
by BNML 02 Jul '13

02 Jul '13

Sorry I meant : "I am _now_ able". Thanks you Le 02/07/2013 19:10, BNML a écrit : > I am not able to propagate the ACK+SDP (and the INVITE without SDP) to > the carrier. Thanks ! > > Le mar. 02 juil. 2013 18:13:40 CEST, Shona McNeill a écrit : >> Try adding >> >> <param name="enable-3pcc" value="proxy"/> >> >> to the sip profile. >> >> The horror that is "bypass media with proxy-mode 3pcc" should handle ACK+SDP after a no-SDP invite. :-) >> At the least, it may point the way to what else needs to be added to make it all work. >> >> >> Shona >> >> >> >>> -----Original Message----- >>> From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch- >>> users-bounces(a)lists.freeswitch.org] On Behalf Of BNML >>> Sent: 02 July 2013 16:49 >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] ReINVITE failure with ACK+SDP scenario >>> >>> I am suspecting a patch on mod_sofia migth be needed in order to >>> resolve the Late SDP Negotiation in an RE-INVITE scenario to avoir the >>> one-way audio. I can get involved in writing the patch, if needed. >>> >>> Here is the FS console logs along with sip trace >>> http://pastebin.freeswitch.org/21141 >>> >>> Le 01/07/2013 14:40, Muhammad Shahzad a écrit : >>> >>> >>> Can you send FS console logs along with sip trace enabled? Just >>> curious how FS responds to this situation. >>> >>> Thank you. >>> >>> >>> On Mon, Jul 1, 2013 at 7:21 PM, BNML <bnml(a)andrexen.com> wrote: >>> >>> >>> Hi, >>> >>> The problem is that FreeSWITCH doesn't forward Late SDP >>> negotiation in a RE-INVITE scenario which end up to one way audio. >>> >>> After testing the option "enable-soa" didn't resolved the >>> issue. >>> >>> I want to bridge two calls that are already established (I >>> am not dealing with INVITE but with Re-INVITE) . >>> That's why inbound-late-negotiation (which is within the >>> dialplan) is not in the scope of this (I have it activated already). >>> >>> >>> > When Call B is answered, you should use SDP in 200 OK as >>> basis for SDP of ACK in Call A. Otherwise you will get one way audio as >>> expect. >>> >>> >>> Yes. This is the scenario I described at state 4) of the >>> schema. >>> >>> >>> Thanks >>> >>> BNML >>> >>> Le 06/28/2013 07:00 PM, Muhammad Shahzad a écrit : >>> >>> >>> I am not sure what you are trying to do here. You >>> said we want to originate two calls from your PABX to carrier through >>> FS, i.e. >>> >>> Call A: PABX -> FS -> Carrier -> User A >>> Call B: PABX -> FS -> Carrier -> User B >>> >>> And then bridge them on FS, such that, >>> >>> User A <- Carrier <- FS -> Carrier -> User B >>> >>> I think there are easier ways to achieve this. But >>> anyway in this scenario, >>> >>> Since Call A does not have SDP, which means you are >>> triggering Late SDP Negotiation. Make sure you have enabled in sofia >>> profile, >>> >>> <param name="inbound-late-negotiation" value="true"/> >>> >>> So, that FS does not Answer it with local SDP. >>> Additionally you may need to disable SOA, >>> >>> <param name="enable-soa" value="false"/> >>> >>> When 200 OK for Call A arrives (from carrier, through >>> FS to PABX), you use it as basis for SDP of INVITE in Call B. That's >>> interesting.. >>> >>> When Call B is answered, you should use SDP in 200 OK >>> as basis for SDP of ACK in Call A. Otherwise you will get one way audio >>> as expect. >>> >>> >>> Thank you. >>> >>> >>> >>> >>> On Fri, Jun 28, 2013 at 6:56 PM, BNML >>> <bnml(a)andrexen.com> wrote: >>> >>> >>> Hi all, >>> >>> I am having trouble with a transfert failure in >>> a special scenario which >>> i wasn't able to find documentation or >>> configuration option solving it. >>> >>> I have a PBX creating two outbound call, it >>> want to bridge them on the >>> FreeSWITCH. >>> It is sending a first INVITE without SDP and it >>> will send the SDP in the >>> ACK. >>> >>> (This is not breaking RFC3261 but I admit >>> rarely seen scenario. >>> http://tools.ietf.org/html/rfc3261#section- >>> 13.2.1 : "The UAC MAY choose >>> to add a message body to the INVITE.". Looking >>> at this scenario there >>> is, indeed, at least one interesting scenario >>> where it can helps >>> avoiding creating another useless Re-INVITE >>> transaction, see below) >>> >>> For the second call it is sending a re-invite >>> with SDP. >>> FreeSWITCH will understand and forward (because >>> of "bypass_media") the >>> second Re-INVITE but does never forward the >>> first Re-INVITE (The one >>> without SDP in the INVITE but in the ACK) to >>> carrier, so this will end >>> up in one-way-audio. >>> >>> A little schema to help : >>> >>> PBX FREESWITCH COMMENT >>> 1) INVITE (a-leg) [THERE >>> IS NO SDP] >>> 2) 200 OK (fs-a- >>> leg)[FreeSWITCH answer with the SDP >>> information before any media update occurs, but >>> my PBX is now aware of >>> media informations] >>> 3) INVITE+SDP (b-leg) [This >>> is b-leg updated media >>> information with previous 200 OK (fs-a-leg)] >>> 4) 200 OK (fs-b-leg)[My PBX >>> updated media and has >>> b-leg freeswitch media information that it will >>> send in the ACK+SDP to >>> finalize the both-legs Re-INVITE] >>> 5) ACK (b-leg) >>> [Acknoledgement of b-leg >>> Re-INVITE transaction] >>> 6) ACK+SDP (a-leg) [ *** >>> problem *** it assume >>> FreeSWITCH will update media information and >>> forward to the carrier >>> which it won't] >>> >>> The solution i imagine is FreeSWITCH SHOULD >>> trigger a Re-INVITE either >>> for the a-leg at INVITE (without SDP in "1") >>> receiption (and update >>> media information when receiving the ACK+SDP) >>> _OR_ create a Re-INVITE >>> transaction at "6" when receiving ACK+SDP. One >>> way or another, I am >>> supposing FreeSWITCH refuse to forward the >>> INVITE without SDP at first >>> maybe because of carrier compatibility ? If >>> not, this seems the way to go. >>> >>> Has anyone faced such a scenario ? Am I rigth >>> assuming a patch is needed >>> ? If yes, any suggestions are welcome. >>> >>> Thanks you >>> >>> BNML >>> >>> >>> >>> _________________________________________________________________ >>> ________ >>> 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/freeswitc >>> h-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >>> -- >>> >>> Mit freundlichen Grüßen >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk(a)hotmail.com >>> Email: shaheryarkh(a)googlemail.com >>> >>> >>> >>> >>> _________________________________________________________________ >>> ________ >>> 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/freeswitc >>> h-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/freeswitc >>> h-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >>> >>> -- >>> >>> Mit freundlichen Grüßen >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk(a)hotmail.com >>> Email: shaheryarkh(a)googlemail.com >>> >>> >>> >>> _________________________________________________________________ >>> ________ >>> 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/freeswitc >>> h-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 >

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adding G729 licences
by Mino Haluz 02 Jul '13

02 Jul '13

Hi, how can I add additional g729 licences in FS. I already have some installed, but after copying new licence file to /etc/freeswitch, it does not make sum, it takes only the last licence file. M

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