- FreeSWITCH-users - FreeSWITCH Mailing Lists

300 Multiple Choices
by Eric Liedtke 01 Jun '09

01 Jun '09

Hey all, Was curious if there is any 300 Multiple Choices support in freeswitch. It seems from the sofia logging that the sofia library just kills the call when it receives a 300 and freeswitch the same. Is this the case or am I missing something? I googled around and searched the list but didn't see anything definitive. So I just wanted to ensure I hadn't overlooked anything. -Eric

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Problem about play wav file in conference
by zhaoxxqq 01 Jun '09

01 Jun '09

HI, I use event socket to send command to FS conference. I send " conference testconf play /root/test.wav" in console. It worked ok. I send "api conference testconf play /record/test.wav" by event socket. and the response is"Disconneted, Good bye.See you at ClueCon..". I changed the wav file to www root. the same problem. can you help me? 2009-06-01 zhaoxxqq

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Passthru mode
by FERNANDO VILLARROEL 01 Jun '09

01 Jun '09

Dear all. I have problem with g729 passthru mode. I received traffic from a Asterisk on my FS and forward to other Asterisk, when i use codec ulaw this works very well. But when i try use G729 i received the following messages and SIP Trace: 2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/admin/42452904(a)190.208.xx.yy [f65514e0-4ec7-11de-9b78-150e2985561f] 2009-06-01 12:19:20 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/admin/42452904(a)190.208.xx.yy entering state [received][100] 2009-06-01 12:19:20 [DEBUG] sofia.c:3044 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 25643 25643 IN IP4 190.208.xx.yy s=session c=IN IP4 190.208.xx.yy t=0 0 m=audio 10236 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [G729:18:8000:0]/[PCMU:0:8000:20] 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] 2009-06-01 12:19:20 [NOTICE] sofia.c:3246 sofia_handle_sip_i_state() Hangup sofia/admin/42452904(a)190.208.xx.yy [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/42452904(a)190.208.xx.yy [KILL] 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904(a)190.208.xx.yy [BREAK] send 886 bytes to udp/[190.47.91.83]:60245 at 16:19:20.596633: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.124:2051;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83 From: "102" <sip:102@200.111.XXX.XX>;tag=bmke36jc1v To: "102" <sip:102@200.111.XXX.XX>;tag=rZp0XXrK9NHFD Call-ID: 3c26700b249f-sryanqz0td8u@snom360-00041323143F CSeq: 27056 REGISTER Contact: <sip:102@192.168.1.124:2051;line=dcnm5x2k>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:72e96588-ebe1-476d-8024-75656b4e007d>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30 Date: Mon, 01 Jun 2009 16:19:20 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904(a)190.208.xx.yy) Running State Change CS_HANGUP 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904(a)190.208.xx.yy) State HANGUP 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/admin/42452904(a)190.208.xx.yy hanging up, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to INVITE with: 488 send 634 bytes to udp/[190.208.xx.yy]:5060 at 16:19:20.603208: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060 From: "452904" <sip:42452904@190.208.xx.yy>;tag=as4e2616ae To: <sip:56968482060@200.111.XXX.XX>;tag=S8FSZr9p6y71r Call-ID: 117330d21f3828470f39a95f538be036(a)190.208.xx.yy CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/admin/42452904(a)190.208.xx.yy Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904(a)190.208.xx.yy) State HANGUP going to sleep 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/admin/42452904(a)190.208.xx.yy) State Change CS_HANGUP -> CS_REPORTING 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904(a)190.208.xx.yy [BREAK] 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904(a)190.208.xx.yy) Running State Change CS_REPORTING 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904(a)190.208.xx.yy) State REPORTING recv 408 bytes from udp/[190.208.xx.yy]:5060 at 16:19:20.620338: ------------------------------------------------------------------------ ACK sip:56968482060@200.111.XXX.XX SIP/2.0 Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport From: "452904" <sip:42452904@190.208.xx.yy>;tag=as4e2616ae To: <sip:56968482060@200.111.XXX.XX>;tag=S8FSZr9p6y71r Contact: <sip:42452904@190.208.xx.yy> Call-ID: 117330d21f3828470f39a95f538be036(a)190.208.xx.yy CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/admin/42452904(a)190.208.xx.yy Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904(a)190.208.xx.yy) State REPORTING going to sleep 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/admin/42452904(a)190.208.xx.yy) State Change CS_REPORTING -> CS_DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 7 (sofia/admin/42452904(a)190.208.xx.yy) Locked, Waiting on external entities 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/admin/42452904(a)190.208.xx.yy) Ended 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/admin/42452904(a)190.208.xx.yy [CS_DESTROY] 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904(a)190.208.xx.yy) State DESTROY 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/admin/42452904(a)190.208.xx.yy SOFIA DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/admin/42452904(a)190.208.xx.yy Standard DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904(a)190.208.xx.yy) State DESTROY going to sleep My vars.xml : <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> <X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/> <X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/> I hope your comments for know where is the config problem Fernando.

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Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?
by Peter Olsson 01 Jun '09

01 Jun '09

I'll try to do this during this weekend. I've looked through the SNV logs, and I really can't find a good reason for this to happen. And when looking into wireshark I can see RTP audio flowing from FS to my SIP phone, but not in the other direction. So this still makes me wonder if something has happened to sofia (that sets up the media incorrectly)... And also when I hangup the call, it takes about a minute for FS to detect this, and it reports hangup reason unknown. But as I said, I'll look into this a bit deeper during this weekend, and file a jira case when I have some more information. //Peter ________________________________________ Från: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org] för Michael Jerris [mike(a)jerris.com] Skickat: den 29 maj 2009 19:35 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org. Thanks Mike On May 29, 2009, at 9:55 AM, Peter Olsson wrote: I’m on Windows, so I have everything under my fs directory, but I deleted the complete directory and did everything from scratch... /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale Skickat: den 29 maj 2009 15:46 Till: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ? On Fri, May 29, 2009 at 8:33 AM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: Nope – it’s not :) Just to make sure I even deleted the source completely, and checked everything out again. Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Brian West Skickat: den 29 maj 2009 15:26 Till: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Nope its not a sofia issue... its build skew ;) On May 29, 2009, at 8:24 AM, Peter Olsson wrote: I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything. /Peter !DSPAM:4a201fa932931035648682!

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Rotating log files not working
by Lars Zeb 01 Jun '09

01 Jun '09

I am trying to rotate the logs, specifically the cdr ones. But the existing extension and Master csv files are not rotated; they remain untouched. I issue the command 'kill -s HUP pid' (pid of freeswitch). The fs console says 2009-05-31 10:25:58 [NOTICE] mod_logfile_c:157 mod_logfile_rotate() New log started. The conf/autoload_configs/cdr-csv.conf.xml shows: <configuration name="cdr_csv.conf" description="CDR CSV Format"> <settings> <param name="default-template" value="sql"/> <param name="rotate-on-hup" value="true"/> <param name="legs" value="a"/> </settings> . What am I doing wrong here? Thanks, Lars

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g729 support
by FERNANDO VILLARROEL 01 Jun '09

01 Jun '09

Dear All, In this moment i am using Asterisk for my services VoIP; i am testing FS and i am very interest on change Asterisk for FS, my motivation is improve my qualify mainly ACD statics. But i have problem, G729 codec is only supported in passthrough mode. FreeSwitch does not do any transcoding with G729 yet, hence I can't fully move to FS without G729 support. I am receiving traffic in g729 in Asterisk(does very bad) i need a FS to receive this traffic and convert to ulaw and forward to other SIP provider, how i can do or anyone help me? I hope your comments or idea how i can do, Fernando.

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Help with lua and channel variable processing
by Larry Marshall 01 Jun '09

01 Jun '09

I would like to process the ${billsec} variable in the conf/autoload_configs/cdr_csv.conf.xml file below, returning a different value for the INSERT command. I already have written a lua function which takes an integer value from ${billsec} and returns a string. It is stored in freeswitch/scripts as sec2time.lua. Do I change the sql template below to ., ${lua(billsec)}, .? <configuration name="cdr_csv.conf" description="CDR CSV Format"> <settings> <param name="default-template" value="sql"/> <param name="rotate-on-hup" value="true"/> <param name="legs" value="a"/> </settings> <templates> <template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${contex t}" ,"${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec} ",... </templates> </configuration> Thanks, Lars

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JavaScript apiExecute vs session.execute
by jcromes＠gmail.com 31 May '09

31 May '09

From JavaScript, what is the difference between apiExecute and session.execute? It seems they are mostly interchangable, in fact the Wiki gives examples of them doing identical operations. What advantages does one have over the other? Are there some things that cannot or should not be called from one or the other? Thanks!

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Is still Woomera supported again?
by Sadjad Seyed-Ahmadian 31 May '09

31 May '09

Hi dear All, I tried to support H323 by opal thaat failed. I want to know is still Woomera supported by FreeSwitch or not. Sincerely, Sadjad

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Missing Events in mod_event_socket
by Gerry Hull 31 May '09

31 May '09

Hello, I am a Windows developer who has written an application around the event_socket interface. My client piece started with the C# EventSocket client Jonas Gauffin had posted on CodePlex. Well, Jonas did not keep up that code on Codeplex, but after communicating with him, I did get the latest client-side code from the freeaswitch SVN, and it seems to work fine. However, their is a persistent, nasty bug I'm seeing: On an inbound call to FreeSwitch, I get the EventChannelAnswer event, which gives me some of the info I need on the incoming call. Following that event, I should get an EventChannelExecuteComplete event, which gives me important information like call-direction, channel-state, answer-state, caller-destination-number, caller-caller-id-name, etc. The problem I'm seeing is that EventChannelAnswer ALWAYS fires on an inbound call, but EventChannelExecuteComplete does not fire --randomly. I thought this mighrt have something to do with linger, but executing the linger command does not help. Jonas made the following comment on the issue: "It has been a bug in the eventsocket implementation in freeswitch. It can sometimes skip packets if the socket layer in the os gives an error code (internal socket buffer becomes full). A simple send retry usually fixes the problem. I've created a patch for it long time ago (and reported it in FS jira). Mike Jerris have made an own fix for the issue. I do not know if it works, I'm still running my own patch. I've attached it to this email. It's a patch for freeswitch\src\mod\event_handlers\mod_event_socket\ mod_event_socket.c, everything works gr8 for me with it." Well, I have no idea how to apply the patch. I've downloaded the latest code from trunk at files.freeswitch.org, and built FS using Visual Studio 2008. all compiles fine. However, the bug sticks it's nasty head up randomly about every other call. I've never done a patch... I tried downloading GNU Patch for windows, and tried applying it, but it reported errors. Has this issue been fixed in core code? If not, can someone help me patch this? I'm dead in the water on a project until I resolve this. In every other aspect, I've found FS to be flawless. Regards, Gerry

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