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Freeswitch transcoding
by Gustavo Espeche 14 Jun '11

14 Jun '11

Hi we are trying to do transcoding with freeswitch g711a-->g711u or g711-->speex but we can't do that FS do a trascoding, follow is a dialplan that we use with mod_curl <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="Regex/XML Dialplan"> <context name="public"> <extension name="external"> <condition field="destination_number" expression="^666654351428063"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="progress_timeout=20"/> <action application="set" data="export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key"/> <action application="set" data="et_routing_number=666654351428063"/> <action application="set" data="et_routing_host=10.0.0.12"/> <action application="set" data="et_random_key=029645322327599477"/> <action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=true"/> <action application="set" data="absolute_codec_string=PCMU"/> <action application="export" data="nolocal:absolute_codec_string=PCMA"/> <action application="set" data="et_routed_number=555554351428063"/> <action application="set" data="et_routed_host=10.0.0.14"/> <action application="bridge" data="sofia/external/555554351428063(a)10.0.0.14"/> </condition> </extension> </context> </section> </document> we relay can't understand what are we doing wrong. are we need configure someone special in FS? we are using sip_external_profile with port 5060. Regards. -- Gustavo Espeche EasyIpCall S.R.L. www.easyipcall.com Bv Mitre 517 24° E Cordoba - Argentina Te: +54 - 351 - 4280633

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interesting mod_erlang_event performance problem
by Seven Du 14 Jun '11

14 Jun '11

Hi, First, i'm running on Mac 10.6.7 with FreeSWITCH Version 1.0.head (git-765908f 2011-05-22 19-10-52 -0500) I found erlang receives events really slow so I did some tests. I use two custom erl to receive events, the file is on github: https://gist.github.com/1024808 The only difference of fse.erl and fse2.erl is that the following line in fse.erl was commented (line 16). {blah, 'freeswitch@localhost'} ! {blah}, % comment this line will be slow I run the scripts at about the same time, so they should can get the same events. The problem is that fse2.erl runs fast ( 1000 events in 6s ) while fse.erl is slow (1000 events in 2 min). (s2@localhost)1> fse2:start(). start {{2011,6,14},{20,21,41}} end {{2011,6,14},{20,21,47}} ok (s@localhost)1> fse:start(). start {{2011,6,14},{20,21,36}} end {{2011,6,14},{20,23,24}} ok It's like fse.erl can only get 10 events per sec, so I guess there's some sleep(100) things in code, but sending a message(what ever) to FS breaks the sleep. the {blah} message can be anything, while a valid msg can get ok, an invalid one gets {error, undef} etc. I use the simple shell script to generate many events: #!/bin/bash IP=192.168.7.7 for f in `seq 1 8`; do for f in `seq 1 20`; do fs_cli -x "bgapi originate sofia/internal/load_test@$IP:5080 9664" done sleep 2 done -- Seven Du About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow

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Lua ASR TTS
by educs13 14 Jun '11

14 Jun '11

Hi, I was trying to run the 'Examples directory lua asr tts' (http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts) in Ubuntu 10.04/32 bits, but I had some problems with the 'session:streamFile()'. If I say something when the prompt is still playing, the audio stucks indefinitely. It seems that the 'break' command isn't doing all its job. I guess that maybe the 'break' is just stop feeding the audio player with new samples, but it is not stopping the player. So, this player keeps reading and playing the same samples continuously ... I really don't know if my guess makes sense and as I don't know C very well, I don't know where to look for a solution in the source code. I Hope you could help me. Thanks, Eduardo -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Lua-ASR-TTS-tp6439449p6439449… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Cannot configure ftmod_libpri
by Valery Kalinin 14 Jun '11

14 Jun '11

Now other errors: 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c19][1:20] Channel not opened, proceeding anyway 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c20][1:21] Channel not opened, proceeding anyway 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c21][1:22] Channel not opened, proceeding anyway 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c22][1:23] Channel not opened, proceeding anyway 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c23][1:24] Channel not opened, proceeding anyway 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c24][1:25] Channel not opened, proceeding anyway 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c25][1:26] Channel not opened, proceeding anyway 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c26][1:27] Channel not opened, proceeding anyway 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c27][1:28] Channel not opened, proceeding anyway 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c28][1:29] Channel not opened, proceeding anyway 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c29][1:30] Channel not opened, proceeding anyway 2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c30][1:31] Channel not opened, proceeding anyway 2011-06-14 16:51:32.058312 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:33.588339 [ERR] ftmod_zt.c:1087 [s1c31][1:16] HDLC abort frame received (ZT_EVENT_ABORT) 2011-06-14 16:51:33.598348 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:33.608355 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:36.158405 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:36.178402 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:36.188409 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:36.208410 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:36.258408 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:36.358406 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:36.388410 [ERR] ftmod_zt.c:1087 [s1c31][1:16] HDLC abort frame received (ZT_EVENT_ABORT) 2011-06-14 16:51:36.438412 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:36.498413 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:36.868433 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE) on sofia profile 'internal' for [2002(a)192.168.205.1] from ip 192.168.20 2011-06-14 16:51:36.928427 [WARNING] sofia_glue.c:213 Codec G723 payload 4 added to sdp wanting ptime 30 but it's already 20 (PCMA:8:20), disabling ptime 2011-06-14 16:51:41.031529 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:43.078575 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame checksum (ZT_EVENT_BADFCS) 2011-06-14 16:51:44.961433 [CRIT] lpwrap_pri.c:321 Error = -1 [Invalid or incomplete multibyte or wide character] 2011-06-14 16:52:12.490765 [CRIT] sofia.c:841 No more sessions allowed at this time. 2011-06-14 16:52:12.649384 [ERR] Span:0 Received UA frame in invalid state 2011-06-14 16:52:15.057934 [WARNING] ozmod_zt.c:1072 Unhandled event 8 for 1:16 2011-06-14 16:52:15.157916 [WARNING] ozmod_zt.c:1072 Unhandled event 8 for 1:16 2011-06-14 16:52:15.257898 [WARNING] ozmod_zt.c:1072 Unhandled event 8 for 1:16 2011-06-14 16:52:15.357880 [WARNING] ozmod_zt.c:1072 Unhandled event 8 for 1:16 2011-06-14 16:52:15.457862 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for 1:16 2011-06-14 16:52:15.557844 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for 1:16 2011-06-14 16:52:15.657826 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for 1:16 2011-06-14 16:52:15.757808 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for 1:16 2011-06-14 16:52:15.857790 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for 1:16 2011-06-14 16:52:16.057754 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for 1:16

1 0

Eavesdrop between 2 FS instaces
by Maciej Aniserowicz 14 Jun '11

14 Jun '11

Hello, Is it possible to eavesdrop between 2 FS instaces? I have CallA parked on FS1 and want to eavesdrop on CallB that is on FS2. I have both channel ids. I tried transfering CallA to FS2, intercept CallB in FS2 dialplan and then eavesdrop in inline dialplan, but this does not work (however it works with bridging). Can I make it work somehow? Best regards, Maciej Aniserowicz -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Eavesdrop-between-2-FS-instac… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Cannot configure ftmod_libpri
by Valery Kalinin 14 Jun '11

14 Jun '11

Hi all! Cannot setup FreeTDM with libpri. I have configured mode network on channel E1, the provider also set this mode. But log write: 2011-06-14 13:42:01.930818 [WARNING] ftmod_libpri.c:1590 -- PRI error event: We think we're the network, but they think they're the network, too. 2011-06-14 13:42:02.934635 [WARNING] ftmod_libpri.c:1590 -- PRI error event: We think we're the network, but they think they're the network, too. 2011-06-14 13:42:03.965449 [WARNING] ftmod_libpri.c:1590 -- PRI error event: We think we're the network, but they think they're the network, too. 2011-06-14 13:42:04.965269 [WARNING] ftmod_libpri.c:1590 -- PRI error event: We think we're the network, but they think they're the network, too. 2011-06-14 13:42:05.905336 [WARNING] ftmod_libpri.c:1590 -- PRI error event: We think we're the network, but they think they're the network, too. 2011-06-14 13:42:06.925361 [WARNING] ftmod_libpri.c:1590 -- PRI error event: We think we're the network, but they think they're the network, too. 2011-06-14 13:42:07.945387 [WARNING] ftmod_libpri.c:1590 -- PRI error event: We think we're the network, but they think they're the network, too. 2011-06-14 13:42:08.565403 [CRIT] lpwrap_pri.c:321 Error = -1 [Invalid or incomplete multibyte or wide character] 2011-06-14 13:43:07.975737 [ERR] Span:0 Received UA frame in invalid state <libpri_spans> <span id="1" name="pri"> <param name="node" value="network"/> <param name="switch" value="4ess"/> <param name="opts" value="omit_redirecting_number"/> <param name="l1" value="alaw"/> <param name="dp" value="unknown"/> <param name="debug" value="all"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </libpri_spans> How to properly set up FreeTDM?

2 1

unload pcmu and pcma
by Gustavo Espeche 14 Jun '11

14 Jun '11

Hi we buy a sangoma d100 and we are having trouble because sangoma support inform to us that we can't do a trascoding g711-->g729 because we can unload the codec g711 from freeswitch. Some one know how we can fix this issue? Best Regards. -- Gustavo Espeche EasyIpCall S.R.L. www.easyipcall.com Bv Mitre 517 24° E Cordoba - Argentina Te: +54 - 351 - 4280633

3 2

Voicemail Message - How t know how many are read & how many are unread in Inbox
by sharad 14 Jun '11

14 Jun '11

Dear All, Just wondering whether there is any way to know how many voice messages are read or unread in New & SAVED box. A help will be highly appriciated. Regards Sharad

3 8

start_dtmf_generate during early media?
by Bill W. 14 Jun '11

14 Jun '11

Hey everyone, I'm trying to terminate a call over a voip carrier to an AT&T conference bridge. AT&T (who, of all people, should know better) are playing the prompts for the conference bridge over early media. And of course, you can't connect to the conf. bridge because no DTMF is transferred (because the call isn't answered yet). So the question is, is there a way to force FreeSWITCH to transcode 2833 DTMF from the A-leg and send inband DTMF on the B-leg while the B-leg is in early-media state? I see there is the start_dtmf_generate application, but will this work if the b-leg is still in the early-media state? Thanks!

2 4

in bypass_media mode, 2833 payload type modified by freeswitch!
by Qingquan Luo 14 Jun '11

14 Jun '11

Hi All, I use freeswitch to bridge one incoming call to other target phone number. I use bypass_media mode. So the rtp is not go through the freeswitch, By I notice that When LegB reply 183 or 200 SDP message. Freeswitch change it 2833 telephone-event payload type. and forwarding the message to caller. This make the caller use wrong dtmf payload type to send dtmf What wrong with that? How can fix it? Any information or help is welcome. Thanks Best Regards Qingquan -- Using Gmail? Please read this important notice: http://www.fsf.org/campaigns/jstrap/gmail?40922.

2 1