- FreeSWITCH-users - FreeSWITCH Mailing Lists

Conference users hear MOH until leader enters?
by j3flight 28 May '09

28 May '09

I'm attempting to replicate the behavior of an Asterisk system with FreeSwitch and I need a feature that, I'm surprised to say, doesn't seem to be supported (easily). Ok, so I've setup my dialplan so that when a specific extension is hit, it calls out to some javascript which acts like an IVR to handle the conference setup. (Similar to this: http://wiki.freeswitch.org/wiki/Examples_confcall_js but with my own improvements.) Anyway, the conferences are stored permanently in a database, but I want them protected by their "owner" so they can only be used when that conference owner dials. If other users have entered the conference prior to the owner, they should hear music-on-hold until the leader enters. This is easy in Asterisk because you can pop someone into MeetMe with different flags. So, in my IVR, I prompt for the "conference number" (known to all) and then the "password" (known only to the owner/leader). If the proper password is entered, the user is sent to conference XYZ with the leader flag set. If no password is entered, the user goes to conference XYZ, without the leader flag. If anyone enters before the leader, they're told by MeetMe that "the conference will begin when the leader arrives" and MeetMe provides MOH until that time. Help! This is an absolute deal-breaker for my install... How can I do this in FreeSwitch? Thanks... -- View this message in context: http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp2371… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Pre8 Release on Digg
by Brian West 28 May '09

28 May '09

Dear FreeSWITCHers, Now I'm gonna take a moment here to guilt each and everyone of you into checking out the story about Pre8 on Digg. We have all worked long and hard to get to 1.0.4 and we still have a little bit to go. So everyone out there that asks "What can I do to help the project?", this is your chance to do so. Help us to promote the project, which brings more people to help in supporting the community, the project and you the end user. Also looking for people to help manage jira, test bugs, ask the right questions and line up the bugs so we can knock them out! Please email me if you're interested. Here is the link for you to help out http://digg.com/search?s=FreeSWITCH+Pre8 Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

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FS PABX experiences?
by Neale Banks 27 May '09

27 May '09

Hi, We're considering deploying FS instead of a traditional PABX/Key-System in a small office environment (i.e. primarily non-technical users, 15-20 handsets). Anyone have any experiences (good/bad/whatever) in this sort of scenario? Thanks, Neale.

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Questions on build 13441
by Lars Zeb 27 May '09

27 May '09

I followed Michael Collin's instruction in an earlier email about building pre8. During the ./bootstrap.h, I encountered the following error: Entering directory /usr/src/freeswitch.trunk/libs/libsndfile Creating aclocal.m4 Running libtoolize... Putting files in AC_CONFIG_AUX_DIR, `Cfg'. Creating configure configure.ac:278: error: possibly undefined macro: AC_OCTAVE_BUILD If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. configure.ac:478: error: possibly undefined macro: AC_GCC_VERSION Do I need to worry about this? Also, in general, do the XML files in config get updated during the 'make install', or are they left as they were from the previous builds? Thanks, Lars

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Double-dtmf detection in IVR when a call is routed through FreeSWITCH
by Drew Ozier 27 May '09

27 May '09

I've got a configuration where I receive inbound calls and dial out to a pre-determined 800-number based on the DNIS of the call. I set <action application="start_dtmf"/> and have everything set up so that DTMF only comes to me via inband. When I'm providing DTMF data to the IVR, it will recognize a single keypress as a double-tap. My FreeSWITCH logs only contain one DTMF entry, but when I listen to the receiving end of the call, I can hear a hiccup in the DTMF tone that is getting played. When I do not use 'start_dtmf', this problem goes away. I need inband DTMF detection, but I can't have it messing up the audio stream. Any thoughts? -Drew Ozier

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Problem: re-invite with 'inactive' SDP and 'bridge' function
by Artem Shiyanov 27 May '09

27 May '09

Hi everyone! In my environment I use FreeSwitch as media server and session border controller. SIP routing is mostly done with my private B2BUA. The problem itself is in my "hold" functionality. In details: A is calling to B:  <extension name="Local_Extension_from_SP"> <condition field="${sip_contact_uri}" expression="^.*StarPound_.*_B2BUA.*$"/> <condition field="destination_number" expression="^([0-9]{2,9})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> </condition> <condition field="${sip_to_host}" expression="^([^.]*)\..*$"> <action application="set" data="orgname=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <anti-action application="set" data="ringback=${us-ring}"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}var callgroup)}"/> <anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <anti-action application="answer"/> <anti-action application="export" data="sip_h_X-SPFrom=&quote;${sip_from_user}&quote;<${sip_from_uri}>"/> <anti-action application="export" data="sip_h_X-SPTo=<${sip_to_uri}>"/> <anti-action application="export" data="sip_h_X-SPCallId=${sip_call_id}"/> <anti-action application="bridge" data="sofia/external/${orgname}send2voicemail@ $${starpound_sip_app_server}"/> </condition> </extension> Call being answered. Then B wants to hold a call (A should hear music-on-hold, B should hear nothing but still have a call). To accomplish this I: 1. on behalf of B2BUA make call to FreeSwitch "hold" extension, get SDP for music-on-hold (moh_sdp) <extension name="hold"> <condition field="destination_number" expression="hold"> <action application="answer"/> <action application="playback" data="$${hold_music}"/> </condition> </extension> 2. re-invite A with SDP moh_sdp 3. re-invite B with it's own sdp and attribute 'inactive' (also tried re-invite with A's SDP and 'inactive' and with 0.0.0.0 IP in SDP and 'inactive' - same results) Result: both A and B hear music-on-hold. I suspect that the reason is in 'bridge' function.. it seems, that it still bridges media against the SIP. Here is a snippet from FreeSwitch log related to "strange" (from my point of view) behaviour - when FreeSwitch change media for B to the incorrect one. Here 1004 is B. 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2939 sofia_glue_negotiate_sdp() Audio Codec Compare [GSM:3:8000:20]/[GSM:3:8000:20] 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:1863 sofia_glue_tech_set_codec() Already using GSM 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2899 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2908 sofia_glue_negotiate_sdp() Set comfort noise payload to 13 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2139 sofia_glue_activate_rtp() Audio params changed for sofia/external/6782886802@10.249.194.9:5060 from 75.101.166.29:27426 to 75.101.166.29:22922 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/6782886802@10.249.194.9:5060] 10.249.194.9 port 28594 -> 75.101.166.29 port 22922 codec: 3 ms: 20 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2165 sofia_glue_activate_rtp() AUDIO RTP CHANGING DEST TO: [75.101.166.29:22922] 2009-05-26 13:19:14 [DEBUG] sofia.c:2996 sofia_handle_sip_i_state() Channel sofia/external/1005(a)10.249.194.9 entering state [ready][200] 2009-05-26 13:19:15 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/doublenat5090/sip:1004@172.16.0.6 <sip%3A1004(a)172.16.0.6> ;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 Command Execute playback(local_stream://moh) EXECUTE sofia/doublenat5090/sip:1004@172.16.0.6 <sip%3A1004(a)172.16.0.6>;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 playback(local_stream://moh) 2009-05-26 13:19:15 [DEBUG] mod_local_stream.c:346 local_stream_file_open() Opening Stream [moh/8000] 8000hz 2009-05-26 13:19:15 [DEBUG] switch_ivr_play_say.c:1084 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2009-05-26 13:19:15 [DEBUG] switch_core_io.c:649 switch_core_session_write_frame() sofia/doublenat5090/sip:1004@172.16.0.6 <sip%3A1004(a)172.16.0.6> ;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 receive message [TRANSCODING_NECESSARY] Any thoughts would be really appreciated! Artem

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AutoChanging port problem
by Mariusz Kołodziejczyk WP 27 May '09

27 May '09

Hello I have about 100 phones in lan. Every phone have set rtp port to 20000 (very simple phone 8Level you can set one rtp port not range like linsys can) my logs in freeswithc looks: 2009-05-27 17:51:10 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 101->166 in context default 2009-05-27 17:51:10 [NOTICE] mod_dptools.c:1274 system_session_function() Executing command: mkdir -p /records/2009-05-27 2009-05-27 17:51:10 [INFO] switch_core_session.c:1391 switch_core_session_execute_application() Sending early media 2009-05-27 17:51:10 [INFO] mod_sofia.c:1448 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1243416430 1243416431 IN IP4 192.168.1.1 s=FreeSWITCH c=IN IP4 192.168.1.1 t=0 0 m=audio 23040 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv 2009-05-27 17:51:10 [NOTICE] mod_sofia.c:1451 sofia_receive_message() Pre-Answer sofia/internal/101(a)192.168.1.1! 2009-05-27 17:51:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/sip:166@192.168.1.166:5060;fs_nat=yes;fs_path=sip%3A166%40192.168.1.166%3A5060 [32e3b290-4ad6-11de-981e-f9f8384b4b12] 2009-05-27 17:51:10 [NOTICE] sofia.c:3101 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:166@192.168.1.166:5060;fs_nat=yes;fs_path=sip%3A166%40192.168.1.166%3A5060! 2009-05-27 17:51:10 [INFO] switch_rtp.c:1800 rtp_common_read() Auto Changing port from 192.168.1.101:16384 to 192.168.1.101:1045 Phone has set rtp port to 16384 but freeswitch changes to 1045 (rtp range on freeswitch is default from 16384 to 32768). Why has it change to 1045 There isn't any firewall on this server Can be a problem if every phone in net has set in his config the same rtp port ? freeSwitch ver: FreeSWITCH Version 1.0.trunk (13430) linux: Linux pbx 2.6.24-23-server #1 SMP Mon Jan 26 00:55:21 UTC 2009 i686 GNU/Linux Thanks

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Transcoding question
by Greg Thoen 27 May '09

27 May '09

Is transcoding a wav file something that I should try to avoid? I have my files encoded as 16kbit, 8000hz, mono files, yet still in the logs I see: 2009-05-27 11:55:18 [DEBUG] switch_ivr_play_say.c:1084 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2009-05-27 11:55:18 [DEBUG] switch_core_io.c:649 switch_core_session_write_frame() sofia/internal/ 5853297721(a)64.24.35.78 receive message [TRANSCODING_NECESSARY] I have to think that not transcoding would be easier on the server, yet I don't know why it needs to transcode the audio file if it is, I think, already in the correct format for that codec... -- Greg Thoen, Vice President CGI Communications, Inc. 1-585-427-0020 x260 http://www.cgicommunications.com/ http://www.elocallink.com/

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XML config error
by Diego Viola 27 May '09

27 May '09

Hi, I have downloaded the latest freeswitch trunk, and when I do reloadxml I get this. Error [unterminated ${var}] in line /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml line 12 Any ideas? I haven't edited that file myself. Thanks, Diego

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Origionate a call via event_socket. relation between job_uuid and uuid
by Peter P GMX 27 May '09

27 May '09

I want to do the following: Originate a call via event_socket, I get back a job_uuid. Then I want to control the call when it's established (2 call legs). Scanning the variables of the 2 call legs I currentyl cannot see any relation between the job_uuid and the uuid of the resulting call legs. I may set a variable with my own unique id while originating a call, but finding the calls later on and dumping the variables fo all channels is very time consuming in terms of CPU. A workaround I tried, is to set caller-id or caller-id-number with a unique id. This works, but has the known side effects of not having a valid caller-id or caller-id-number. So my question is: Has anybody an idea how to build a relationship between job_uuid and the resulting call legs which does not require dumping the variables of all channels? Best regards Peter

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