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Freetdm configuration
by Nikolay Zaytsev 28 Jul '15

28 Jul '15

Good day to you. I have a problem with the setting up of the freetdm module. I have sangoma a102de card. The system recognizes the card correctly. Audiocodes mediant 600 with the same trunk configuration works corrctly. But during the startup freeswitch writes [DEBUG] ftdm_io.c:5469 Module sangoma_isdn does not support configuration. and after that [INFO] ftmod_sangoma_isdn_stack_rcv.c:1098 sng_isdn->s2: Link is down, dropping transmit frame. Config files are attached. Please, help me:) Best regards, Nikolay Zaytsev

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audio Handshake failure 1
by Alex Polischuk 28 Jul '15

28 Jul '15

Hi All, I use FreeSwitch for transcoding SRTP from WebRTC to SIP with RTP and vice versa. When Firefox version 38 add requires for Perfect Forward Secrecy (PFS) I have problem with Handshake in FS when Firefox Browser terminate a call. I found that that problem was fixed in the version 1.4.19 https://freeswitch.org/jira/browse/FS-7425 I have the same problem in the version 1.4.20 eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:2924 Changing audio DTLS state from HANDSHAKE to SETUP eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:2832 audio Fingerprint Verified. eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:3384 Activating Audio Secure RTP SEND eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:3362 Activating Audio Secure RTP RECV eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:2872 Changing audio DTLS state from SETUP to READY eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.604193 [DEBUG] switch_rtp.c:1937 rtcp_stats_init: ssrc[-1374535478] base_seq[6179] f01c46d8-33a0-11e5-ac83-891cb7dea826 2015-07-26 14:16:46.644210 [DEBUG] switch_rtp.c:5884 Correct ip/port confirmed. f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [ERR] switch_rtp.c:2917 audio Handshake failure 1 f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [INFO] switch_rtp.c:2918 Changing audio DTLS state from HANDSHAKE to FAIL f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [NOTICE] switch_rtp.c:2899 Hangup sofia/external/992555500004(a)voip.webrtc.jajah.com [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] what additional definition I should make to resolve this problem? Thanks, Alex Polischuk

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Incorrect Caller-Context in the events
by Dmitri Muntean 28 Jul '15

28 Jul '15

Hey guys, I'm running freeswitch 1.4.20 and 1.4.13 with the number of conferences setup on them and have 3 profiles and contexts defined, none of those are called "default". I read the events via inbound event socket, and most of the events happening inside those conferences have correct Caller-Context set to the profile that the call came through. But sometimes (very rarely) the Caller-Context is set to "default" - I was wondering if anyone seen such behaviour. I don't have "default" profile or context defined anywhere in my configuration. The events I am reading off freeswitch are Event-Name: DTMF and Event-Name: CUSTOM with Event-Subclass: conference::maintenance. When the Caller-Context is "default" it appears so in both type of events. Thanks

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Should vpn address space be defined as part of local network?
by Rajil Saraswat 28 Jul '15

28 Jul '15

Hello all, I am trying to get my head around the nat.auto and localnet.auto acls. I have a VPN server using the 10.8.0.0/24 address space with gateway on 10.8.0.1. The PBX is on the local lan (172.16.5.0/24) with ip 172.16.5.5. When freeswitch starts i see it builds the following acls nat.auto Created ip list nat.auto default (deny) Adding 172.16.5.5/255.255.255.0 (deny) to list nat.auto Adding 10.0.0.0/8 (allow) [] to list nat.auto Adding 172.16.0.0/12 (allow) [] to list nat.auto localnet.auto Created ip list localnet.auto default (deny) Adding 172.16.5.5/255.255.255.0 (allow) to list localnet.auto Do i need to move my vpn address space (10.8.0.0/16) from nat.auto to the localnet.auto so that it not natted? Something like this: nat.auto 172.16.5.5/255.255.255.0 (deny) 10.0.0.0/8 (allow) 172.16.0.0/12 (allow) 10.8.0.0/16 (deny) localnet.auto 172.16.5.5/255.255.255.0 (allow) 10.8.0.0/16 (allow) Thanks

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Fwd:
by Nikolay Zaytsev 27 Jul '15

27 Jul '15

Good day to you. I have a problem with the setting up of the freetdm module. I have sangoma a102de card. The system recognizes the card correctly. Audiocodes mediant 600 with the same trunk configuration works corrctly. But during the startup freeswitch writes [DEBUG] ftdm_io.c:5469 Module sangoma_isdn does not support configuration. and after that [INFO] ftmod_sangoma_isdn_stack_rcv.c:1098 sng_isdn->s2: Link is down, dropping transmit frame. Config files are attached. Please, help me:) Best regards, Nikolay Zaytsev

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Coder Games Monday August 3rd! Remote Participants needed!
by Brian West 27 Jul '15

27 Jul '15

We are playing a version of Hollywood Squares, but it will be VoIP related questions, If you would like to participate, you'll need a camera, headset and Chrome to participate, Please RSVP to kathleen(a)freeswitch.org your details, and we'll do a tech check this week sometime. https://en.wikipedia.org/wiki/Hollywood_Squares If you get a chance watch some videos on Youtube. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Video conferencing
by Alexandru Covalschi 27 Jul '15

27 Jul '15

Hello! Started to figure out the theme in topic. Didn't find any descriptions of what is "video-floor" and etc. Can you please provide a link to the theory I need to read to understand all definitions? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

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Re: [Freeswitch-users] FS 1.6 video dependency build issue
by Ken Rice 27 Jul '15

27 Jul '15

Unless you are using sangoma transcoding hardware you don’t need libsngtc And why 32bit? From: <freeswitch-users-bounces(a)lists.freeswitch.org> on behalf of Peter Stevens Reply-To: FreeSWITCH Users Help Date: Friday, July 24, 2015 at 10:55 AM To: "freeswitch-users(a)lists.freeswitch.org" Subject: [Freeswitch-users] FS 1.6 video dependency build issue Hi, I have a clean install of Debian 8 on a 32 bit machine. I'm following information on FS 1.6 video wiki (https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video) to build this version, but the installation of the dependencies part doesn't install all of the required dependencies. After running: DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yesfreeswitch-video-deps-most I had to then: apt-get install libavcodec-extra apt-get install portaudio19-dev Which were both ok, but running: apt-get install libsngtc-dev I received following: Package libsngtc-dev is not available, but is referred to by another package. This may mean that the package is missing, has been obsoleted, or is only available from another source E: Package 'libsngtc-dev' has no installation candidate Re-ran: DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yes freeswitch-video-deps-most Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: freeswitch-video-deps-most : Depends: libsngtc-dev but it is not installable E: Unable to correct problems, you have held broken packages. I've had a look around amongst the http://files.freeswitch.org/repo/deb/debian/dists/jessie/main directory structure and have discovered that the libsngtc-dev is listed in both the binary-amd64/packages.gz and the binary-amd64/packages.gz files, but is only listed in the Contents-amd64.gz file and not in the Contents-i386.gz file. Don't know whether this is significant or not, but how/where can I obtain the libsngtc-dev file to complete the dependency installation for freeswitch-video-deps-most? Any help appreciated. Peter ---------------------------- http://www.bbc.co.uk This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. --------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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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Can I have a dialplan action that runs when a call hangs up?
by D D 24 Jul '15

24 Jul '15

Hi, I need to have an action take place when a call hangs up. The call will have been in a conference, and I need to take extra actions after it hangs up.How can I do this using the dialplan? Thanks!David

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mod_xml_radius starts accounting twice for incoming calls
by Artur Mega 24 Jul '15

24 Jul '15

When new incoming call comes to tr2.xxxxxxx.ru, new session is being created and accounting begins. Further, call forwarding to another server, to fs2.xxxxxxx.ru, and new session is being created again. 2015-07-23 11:22:42.514095 [NOTICE] switch_channel.c:1054 *New Channel* sofia/external/73472460000@tr2.xxxxxxx.ru:5080 [ 26777290306690-93831426613199(a)192.168.217.156] ... 2015-07-23 11:22:42.574084 [INFO] mod_xml_radius.c:1123 mod_xml_radius: *Accounting Start success* 2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:164 sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Standard ROUTING 2015-07-23 11:22:42.574084 [INFO] mod_dialplan_xml.c:558 Processing 73472460000 <73472460000>->79373057071 in context public Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 parsing [public->originate_leg] continue=true Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Absolute Condition [originate_leg] Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Action export(nolocal:h323-call-origin=originate) Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 parsing [public->public_mult] continue=false Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Regex (PASS) [public_mult] destination_number(79373057071) =~ /^(79373057071)$/ break=on-false Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Action set(hangup_after_bridge=true) Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Action bridge(sofia/internal/79373057071@fs2.xxxxxxx.ru:5070) 2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:214 (sofia/external/73472460000@tr2.xxxxxxx.ru:5080) State Change CS_ROUTING -> CS_EXECUTE ... 2015-07-23 11:22:42.574084 [NOTICE] switch_channel.c:1054 *New Channel* sofia/internal/79373057071@fs2.xxxxxxx.ru:5070 [e059ad7f-b2f8-4beb-b1f3-249bc431cfc4] ... 2015-07-23 11:22:42.634059 [INFO] mod_xml_radius.c:1123 mod_xml_radius: * Accounting Start success* Thus, one call is being counted twice (for incoming to tr2, and then for outgoing from tr2 to fs2). But we need to make mod_xml_radius counts this call only once, how can we handle it? Thanks -- Arthur

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