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Opus error: Invalid SDP for opus. Don't ask.. but it needs a /2
by Donny Hardyanto 11 Feb '14

11 Feb '14

Hi, I just testing the Opus in FS talking with Linphone. There is error in FS log: Invalid SDP for opus. Don't ask.. but it needs a /2 What is that mean? The call is success full connected but one side voice. Donny

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Freeswitch Week in Review(Master Branch) Feb 2nd through Feb 8th
by Kathleen King 11 Feb '14

11 Feb '14

This past week was productive for the Freeswitch master branch with many new changes and bug fixes. On the packaging front, .zip joined tar.xz in the default source release archives. Located here: http://files.freeswitch.org/ In terms of stability there were several use cases that were fixed: --Fixed mod_lua crashes related to build symbols introduced during upgrade from lua 5.1 to lua 5.2 --Fixed a crash in mod_conference related to whisper tones --Fixed a crash in mod_shell_stream --Fixed a crash involving T-38 in mod_sofia when the call in bridged between TDM and SIP Several new features were added: – New command 'gethost' allows for a DNS lookup from the Freeswitch console --Added bypass-media-after-hold to complement the existing parameter resume-media-on-hold with more information located at: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#bypass-media-after-hold --Added capability to record conference with a single participant --Added support for Sofia to configure accepted TLS versions and enabled support for perfect forward secrecy. --Expanded Sofia profile status to include websockets Improvements in cross platform build supports: --Spandsp improvements for BSD --Some initial build fixes for Ubuntu 14.04 --Fixing some compiler optimization snafus Feedback welcome. The referenced commits are in the attached text file. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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ZRTP proxying
by Dahlberg, David 11 Feb '14

11 Feb '14

Hello all, I have some problems with proxying ZRTP streams trough Freeswitch. My setup is as follows: [FS-A]----[FS-B]----[FS-C] For End-to-end security, "A" should initiate a ZRTP session and "C" shall terminate it. What works: * With late negotiation, two ZRTP security associations (A,B) and (B,C) are established. Of course in that case there is no E2E encryption, so no real advantage over the usage of TLS/SAVP . According to the Wiki, it should be possible to use "proxy_media" with "inbound-zrtp-passthru". Only this does not work and the problem is this: ZRTP uses a slightly modified RTP header format with a CRC checksum that is calculated over the whole ZRTP header and payload (excluding the CRC field). Freeswitch ("FS-B" in my use-case) on the other hand exchanges the RTP "sequence numbers" and "syncronization source identifiers" (SSRC) (which in ZRTP become "Source Identifiers"). Which leads to the case that the checksums are valid on call leg (A,B) but invalid on leg (B,C). Accordingly, "FS-C" receives ZRTP hello packets from "FS-A", calculates the checksums, finds a mismatch and discards the packets. I have found a thread on [freeswitch-users] dated January 2013 and with a subject "problems with freeswitch + zrtp in proxy-media mode" which discusses exactly this problem. Within this thread was mentioned, that FS would change the proxy behaviour, if it encounters "zrtp-hash" attribute in the SDP and a quick test with SFLphone and w/o proxy_media seems to confirm this. So is there any possibility to originate this attribute, when FS originates the ZRTP? Regards, David -- David Dahlberg Fraunhofer FKIE, Dept. Communication Systems (KOM) | Tel: +49-228-9435-845 Fraunhoferstr. 20, 53343 Wachtberg, Germany | Fax: +49-228-856277

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Controlling B-Leg on A-Leg hangup
by Alex Lake 11 Feb '14

11 Feb '14

I've got a situation where I think I want to prevent the A-Leg hangup resulting in an immediate B-Leg termination. It's like a form of attended transfer where A calls B, B calls C and then hangs up, with the desired outcome that A is talking to C. Suggestions?

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proxy_media possibly broken? "INCOMPATIBLE_DESTINATION"
by Peter Steinbach 11 Feb '14

11 Feb '14

We are trying to relay a fax from one T.38 compatible device to another. At an early stage, while we are still in PCMA, we receive a [INCOMPATIBLE_DESTINATION] This wonders, as Freeswitch should not be really involved in Codec negociation with the endpoints. In the logs below I can see, that DTMF SDP lines are handled as soon as early media is received from the target endpoint (Asterisk): 2014-02-06 21:46:26.309091 [DEBUG] switch_core_media.c:3085 Set telephone-event payload to 101 2014-02-06 21:46:26.309091 [DEBUG] switch_core_media.c:3382 Set 2833 dtmf send payload to 101 And then there is the INCOMPATIBLE_DESTINATION nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:529 nua_signal() nua(0x1645df0): sent signal r_respond 2014-02-06 21:46:26.309091 [NOTICE] sofia.c:6051 Hangup sofia/external/49712345(a)30.30.30.30 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] nua_stack.c:573 nua_stack_signal() nua(0x1645df0): recv signal r_respond 488 Not Acceptable Here nua_stack.c:271 nua_stack_event() nua(0x1645df0): event i_error 500 Responding to a Non-Existing Request So my question is: Is proxy-media broken here? Do I need another (updated Freeswitch) where this is already solved? Freeswitch is from GIT mid dec 2013. The dialplan is (in short) as follows <action data="proxy_media=true" application="set" /> <action data="sofia/gateway/provider/49712345(a)30.30.30.30" application="bridge" /> SDP are as follows Sender v=0 o=FreeSWITCH 1391701261 1391701262 IN IP4 10.10.10.10 s=FreeSWITCH c=IN IP4 10.10.10.10 t=0 0 m=audio 12816 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Freeswitch SDP v=0 o=FreeSWITCH 0325869010 0325869011 IN IP4 20.20.20.20 s=FreeSWITCH c=IN IP4 20.20.20.20 t=0 0 m=audio 19068 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Receiver's SDP v=0 o=root 458337462 458337462 IN IP4 30.30.30.31 s=Asterisk PBX 11.0.1 c=IN IP4 30.30.30.31 t=0 0 m=audio 10262 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ================= Freeswitch LOG ==================================== recv 868 bytes from udp/[30.30.30.30]:5060 at 20:46:26.309161: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 20.20.20.20:5080;received=20.20.20.20;rport=5080;branch=z9hG4bKXQeZgU1337FFe Record-Route: <sip:30.30.30.30;lr> From: "Telefaks" <sip:49691234567@20.20.20.20>;tag=myacymU8arjje To: <sip:49712345@30.30.30.30>;tag=as25c99f07 Call-ID: 967f3f6c-0a12-1232-a6a1-439bc1c9be81 CSeq: 55523729 INVITE Server: Asterisk PBX 11.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:49712345@30.30.30.31:5060> Content-Type: application/sdp Content-Length: 264 v=0 o=root 2049304398 2049304398 IN IP4 30.30.30.31 s=Asterisk PBX 11.0.1 c=IN IP4 30.30.30.31 t=0 0 m=audio 18798 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------------------------------------------ tport.c:3028 tport_deliver() tport_deliver(0x16d4960): msg 0x7f5f5802d0f0 (868 bytes) from udp/30.30.30.30:5080/sip next=(nil) nta.c:3221 agent_recv_response() nta: received 183 Session Progress for INVITE (55523729) nta.c:3284 agent_recv_response() nta: 183 Session Progress is going to a transaction tport.c:4227 tport_release() tport_release(0x16d4960): 0x7f5f5001a6e0 by 0x7f5f5811ced0 with 0x7f5f5802d0f0 (preliminary) soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7f5f6020dea0, (nil), 0x7f5f58123d6c, 264) called soa.c:1595 soa_process_answer() soa_process_answer(static::0x7f5f6020dea0) called soa_static.c:1137 offer_answer_step() soa_static_offer_answer_action(0x7f5f6020dea0, soa_process_answer): called soa_static.c:1020 soa_sdp_mode_set() soa_sdp_mode_set(0x7f5f5001adb0, 0x7f5f581285e0, ""): called soa_static.c:1283 offer_answer_step() soa_static(0x7f5f6020dea0, soa_process_answer): upgrade codecs with remote description soa.c:1730 soa_activate() soa_activate(static::0x7f5f6020dea0, (nil)) called nua_session.c:988 nua_session_client_response() nua(0x1645df0): INVITE: processed SDP answer in 183 Session Progress nua_stack.c:271 nua_stack_event() nua(0x1645df0): event r_invite 183 Session Progress nua_session.c:4137 signal_call_state_change() nua(0x1645df0): call state changed: calling -> proceeding, received answer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7f5f6020dea0, [0x7f5f65afb2f8], [0x7f5f65afb2f0], [(nil)]) called soa.c:616 soa_get_params() soa_get_params(static::0x7f5f6020dea0, ...) called nua_stack.c:271 nua_stack_event() nua(0x1645df0): event i_state 183 Session Progress nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2014-02-06 21:46:26.309091 [DEBUG] switch_core_session.c:1048 Send signal sofia/external/49712345(a)30.30.30.30 [BREAK] 2014-02-06 21:46:26.309091 [ALERT] switch_core_state_machine.c:585 sofia/external/49712345(a)30.30.30.30 session thread wake state: CS_CONSUME_MEDIA! nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2014-02-06 21:46:26.309091 [DEBUG] switch_core_session.c:1048 Send signal sofia/external/49712345(a)30.30.30.30 [BREAK] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-02-06 21:46:26.309091 [ALERT] switch_core_media.c:286 Looking for zrtp-hash 2014-02-06 21:46:26.309091 [ALERT] switch_core_media.c:241 Deciding whether to pass zrtp-hash between legs 2014-02-06 21:46:26.309091 [ALERT] switch_core_media.c:243 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2014-02-06 21:46:26.309091 [ALERT] sofia.c:983 sofia/external/49712345(a)30.30.30.30 Same Callee ID "Outbound Call" <49712345> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-02-06 21:46:26.309091 [DEBUG] sofia.c:5889 Channel sofia/external/49712345(a)30.30.30.30 entering state [proceeding][183] 2014-02-06 21:46:26.309091 [DEBUG] sofia.c:5899 Remote SDP: v=0 o=root 2049304398 2049304398 IN IP4 30.30.30.31 s=Asterisk PBX 11.0.1 c=IN IP4 30.30.30.31 t=0 0 m=audio 18798 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2014-02-06 21:46:26.309091 [ALERT] switch_core_media.c:286 Looking for zrtp-hash 2014-02-06 21:46:26.309091 [ALERT] switch_core_media.c:241 Deciding whether to pass zrtp-hash between legs 2014-02-06 21:46:26.309091 [ALERT] switch_core_media.c:243 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2014-02-06 21:46:26.309091 [DEBUG] switch_core_media.c:3085 Set telephone-event payload to 101 2014-02-06 21:46:26.309091 [DEBUG] switch_core_media.c:3382 Set 2833 dtmf send payload to 101 nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:529 nua_signal() nua(0x1645df0): sent signal r_respond 2014-02-06 21:46:26.309091 [NOTICE] sofia.c:6051 Hangup sofia/external/49712345(a)30.30.30.30 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] nua_stack.c:573 nua_stack_signal() nua(0x1645df0): recv signal r_respond 488 Not Acceptable Here nua_stack.c:271 nua_stack_event() nua(0x1645df0): event i_error 500 Responding to a Non-Existing Request nua_stack.c:359 nua_application_event() nua: nua_application_event: entering -- With kind regards Peter Steinbach

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gateway outbound call failover
by Jason Holden 11 Feb '14

11 Feb '14

All, Quick question. Can anyone recommend an example of the best dialplan for having gateway failover that does not use the LCR? Thanks.

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Split string and play charachters
by Grant Bagdasarian 11 Feb '14

11 Feb '14

Hello, Is there an application which plays the characters of a string individually? Or have some sort of split function which splits a string into characters and then manually map the 'character_name' to an existing audio file? For instance: //String: 1234567890 //Split the string Playback: 1.wav Playback: 2.wav Playback: 3.wav Etc Can this be done in the XML dialplan or do I need to use some scripting language to control the dialplan?

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execute_on_xxx too persistent
by Alex Lake 11 Feb '14

11 Feb '14

FreeSWITCH Version 1.2.10 Got a call where the A-Leg triggers a bridge to a B-Leg where I want to have certain things happen on ring/answer. So I based my dial plan on the example code <action application="export" data="nolocal:execute_on_answer=lua incrInUse.lua ${uuid}"/> Although mine is in Lua session:execute("export","nolocal:execute_on_ring=lua cc_tell.lua ring "..CCNumIPN.." "..getSessVar("uuid")) And it works fine. However, it APPEARS that I have a problem when the B-Leg initiates a C-Leg, that the execute_on_xxx seems to be executed by the C-Leg too - not what I want! I think there are ways around this (i.e. put some more intelligence into the lua code) but I can't help thinking that I've misunderstood something about export and/or the execute_on_ family.

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Heart beat set during call and deduct balance of user
by Abdullah 11 Feb '14

11 Feb '14

hi all, current i am using mode nibble for billing , this module is osum, kindly help me how this (module nibble ) to get or set heart beat on call and detect balance from caller account with out using this module.. Thanks in advance,, Regard iffi -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Heart-beat-set-during-call-an… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Customized CDR
by Abdullah 11 Feb '14

11 Feb '14

Hi all genius Developer's, How i get free-switch , Answer-time ,billsec and duration variables in Perl script with out using API hangup hook and using other Module of Freeswitch (CVS,XML-CDR etc). i have try to get Global and local variable like this but not getting any value after hangup call. $billsec = $session->getVariable("billsec"); but when i get Dial status variable value return like SUCCESS But why i cant get Billsec using this ?? Suggestion are Welcome.... Thanks Abdullah -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Customized-CDR-tp7596058.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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