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Video not secured on B-Leg
by Asier Benito 09 Jul '13

09 Jul '13

Hi, I'm new to Freeswitch and I'm trying to secure B-leg with SRTP from incoming call with video and audio via RTMP. To secure it, i'm using nolocal:sip_secure_media=true, the problem is that freeswitch only secures audio but not video, I'm using Freeswitch 1.2.11 and Linphone as B leg, here you have the SDP: 2013-07-09 08:35:07.210255 [DEBUG] switch_channel.c:1176 sofia/internal/ 1006(a)5.56.58.79 EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-07-09 08:35:07.210255 [DEBUG] switch_channel.c:1176 sofia/internal/ 1006(a)5.56.58.79 EXPORTING[export_vars] [sdp_secure_savp_only]=[true] to event 2013-07-09 08:35:07.210255 [DEBUG] sofia_glue.c:2678 Local SDP: v=0 o=FreeSWITCH 1373328169 1373328170 IN IP4 XX.XX.XX.XX s=FreeSWITCH c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 30738 RTP/SAVP 99 100 101 13 a=rtpmap:99 SPEEX/16000 a=rtpmap:100 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:u9q805XwSn8pIzKZZbfxxGoOOQU6ztTbr7zCUx+q a=ptime:20 a=sendrecv m=video 24388 RTP/AVP 98 a=rtpmap:98 H264/90000 a=fmtp:98 packetization-mode=0; profile-level-id=42000a If I capture packets with tcpdump I can see how audio is secured with SRTP but not the video. Any idea about what is going wrong? Thanks in advance. Regards.

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effective_caller_id_number / freetdm / libpri - not working
by Peter Waldheim 09 Jul '13

09 Jul '13

Hi, I have seen some posts about caller-id-problems recently, but not many resolutions. Also as this uses a PRI line, I'll post it extra. I'm trying to get user-specific caller-ids (or any caller-id for now really) to work on a PRI line. This has been working with a (very) old version of freeswitch and libpri, so if there is a change in the required configuration, a pointer to that would be very welcome of course. In the dialplan I set effective_caller_id_number (and _name just to be sure) right before bridging (also nowhere else): <action application="export" data="effective_caller_id_number=0xxxxxxx21"/> <action application="export" data="effective_caller_id_name=0xxxxxxx21"/> <action application="info"/> <action application="bridge" data="freetdm/1/a/$1" /> Info displays accordingly: variable_effective_caller_id_number: [0xxxxxxx21] variable_effective_caller_id_name: [0xxxxxxx21] variable_export_vars: [effective_caller_id_number,effective_caller_id_name] variable_current_application: [info] But the other side still sees the base-number (0xxxxxxx20) of the pri-trunk. Any ideas on what is going wrong or required to make this work would be greatly appreciated (or how to further analyze this). Thanks and regards Peter

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Re: [Freeswitch-users] LRN DIP 302 Dialplan
by Andre Demattia 09 Jul '13

09 Jul '13

Hi, I'm checking to see if that is the best way to do the dip. So calling the dialplan then waiting for the on hang up complete then dialing again is the right way? Andre -----Original Message----- From: "Avi Marcus" <avi(a)avimarcus.net> Sent: ‎7/‎8/‎2013 5:07 PM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] LRN DIP 302 Dialplan Sure, do the dip first then do your LCR. Maybe I'm just not getting the question? -Avi On Mon, Jul 8, 2013 at 6:54 PM, <andretodd(a)verizon.net> wrote: HI, I need to look up a number to see if it's ported/ LCR before I bridge the call. What is the best way to go about doing this. Call comes in, I need to look up the LRN data via SIP 302. The results of that I need to hit my database for the LCR then I need to bridge the number. Would it be wise to dial the LRN server then on Hangup complete hit the database then dial the number? 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

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Loss of DTMF in latest GIT with SRTP enabled?
by Yehavi Bourvine 09 Jul '13

09 Jul '13

Hello, I've noticed that DTMF does not work with the recent latest GIT versions when using SRTP. When I disable SRTP it works ok. Has anyone else noticed it? BTW, I've opened a Jira FS-5571 on it. Thanks, __Yehavi:

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Cisco 7970G MWI + External SIP Voicemail Server
by Jeremy Moschner 09 Jul '13

09 Jul '13

Hi Everyone I am running an embedded FreeSWITCH installation (on OpenWRT, opkg version 1.0.7-2) and have a single Cisco 7970G that registers fine and can make and recieve calls (most of the time). However, one of my SIP providers (Internode) also offers a voicemail system (useful if I have my phone turned off and means I don't have to run a voicemail module on my installation). What I would like to find out is if it is possible for the 7970G (or indeed any other endpoint) to query/receive notification from Internode's voicemail system and display a MWI (or some sort of notification on the phone) when a message is there. Now, from my reading, I believe that this is accomplished through either SIP NOTIFY or SUBSCRIBE messages, but I am still quite confused about how it all works. I am unsure if FreeSWITCH needs to "relay" this information to the endpoint, or if is configurable via the MWI-Account parameter (ie. the directory user registers to receive these updates directly from the SIP server). I imagine I'll need to provide a bit more detail, so please let me know where to start.

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LRN DIP 302 Dialplan
by andretodd＠verizon.net 08 Jul '13

08 Jul '13

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FreeSWITCH Weekly News and Notes
by Michael Collins 08 Jul '13

08 Jul '13

Greetings FreeSWITCH users! WebRTC continues to be a hot topic. We've had a lot of interest in the subject since the announcement <http://freeswitch.org/node/455> of FreeSWITCH 1.4 beta. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_03>we had Ken Rice show us just how easy it is to get a WebRTC-enable page up and running very easily using sipML5 <http://www.sipml5.org> or JsSIP<http://www.jssip.net>. FreeSWITCH users are also welcome to view our WebRTC page's<https://webrtc.freeswitch.org/webrtc/portal.html>underlying code for inspiration. This week we will be doing a community discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_10>highlighting some recent mailing list topics. We will also be entertaining questions from the audience, so if you have have a FreeSWITCH question please join us on Wednesday. In ClueCon news we are happy to let you know that everyone who registers by Friday, July 12th will triple their chances of winning some great prizes, including the 10 Android tablets that will be given away this year. Simply click the "Register Now" button on the ClueCon site <http://www.cluecon.com>to complete the registration process. Also, plan on being in Chicago on Monday August 5th so that you can hang out with the team and meet other ClueCon attendees. Can you believe that ClueCon 2013 is less than a month away? Have a great week and we will talk to you on Wednesday. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Connect FreeSWITCH calls through another box.
by Matías Sanguinetti 08 Jul '13

08 Jul '13

Hi everyone, I got the following scenario, I got two FreeSWITCH boxes, lets call them A and B, and an external gateway (GW). The idea is to bridge two channels through the GW. B has 286ms ping with GW so it suffers from a little delay in the calls, A has 15ms ping with GW but its CPU capacity makes the call stutter. B has 32ms ping with A so I was planning to use A as a proxy and process the call on B, I read that something can be done with proxy_media and transfer, mixing internal profiles on A to transfer calls to B. Any idea how to do this? Thanks -- Matías Sanguinetti

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Can't compile Freeswitch 1.2.11 on debian wheezy
by mbo 08 Jul '13

08 Jul '13

I can't compile freeswitch 1.2.11 on debian wheezy (7.1). I get the following error: gcc -I/usr/src/freeswitch-1.2.11/libs/curl/include -I/usr/src/freeswitch-1.2.11/src/include -I/usr/src/freeswitch-1.2.11/src/include -I/usr/src/freeswitch-1.2.11/libs/libteletone/src -I/usr/src/freeswitch-1.2.11/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 -o .libs/freeswitch freeswitch-switch.o -Wl,-lodbc -lm ./.libs/libfreeswitch.so -L/usr/src/freeswitch-1.2.11/libs/apr-util/xml/expat/lib /usr/src/freeswitch-1.2.11/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/src/freeswitch-1.2.11/libs/apr/.libs/libapr-1.a -L/usr/src/freeswitch-1.2.11/libs/srtp -lz libs/apr/.libs/libapr-1.a -luuid -lrt -ldl -lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lncurses -ljpeg -lodbc -Wl,--rpath -Wl,/usr/local/freeswitch/lib /.libs/libfreeswitch.so: undefined reference to `SSLv2_client_method' Is there a solution for that? Thanks Markus

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Strip allow header
by Mino Haluz 08 Jul '13

08 Jul '13

Hi, now I have this in Allow header: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY My FS should not implement these features in my scenario: MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Could you please point me to modules that are responsible for these methods so that I could disable them? Thanks, M

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