June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_lcr caller id
by Yuriy Ivzhenko 04 Jun '09

04 Jun '09

Hello. I have another problem with setting caller id with cid field. command lcr has parameter [caller_id]. and if i call lcr from the commandline it works fine, but if i call lcr from dialplan it ignores this parameter and use original caller id from caller. my cid field = /^(.*)$/999$1/ If i call lcr from the commandline lcr 444555 default 11111 lcr return origination_caller_id_number=99911111 but if i call lcr from dialplan <action application="lcr" data="444555 default 11111"/> lcr return origination_caller_id_number=9991002 1002 is a original caller id number There is a way to use this parameter from dialplan? Yuriy

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busy tone detect issue
by god.nirvana 04 Jun '09

04 Jun '09

hi all i am new to freeswitch. there are some busy tone detect issues,i hope someone could help me. i installed freeswitch from trunk,openzap,zaptel.... but i found some busy tone isuues my tones.conf: [us] generate-dial => v=-7;%(1000,0,350,440) detect-dial => 350,440 generate-ring => v=-7;%(2000,4000,440,480) detect-ring => 440,480 generate-busy => v=-7;%(500,500,450,340) detect-busy =>450,340 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 776.7 openzap.conf: [span zt FXO1] name => OpenZAP-FXO1 number => 1 fxo-channel => 1 [span zt FXO2] name => OpenZAP-FXO2 number => 2 fxo-channel => 2 [span zt FXO3] name => OpenZAP-FXO3 number => 3 fxo-channel => 3 [span zt FXO4] name => OpenZAP-FXO4 number => 4 fxo-channel => 4 openzap.conf.xml : <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/>    </settings> <pri_spans>  <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </pri_spans>  <analog_spans>    <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/>        <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/>        <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/>        <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/>     </analog_spans> </configuration> when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> in the dialplan.and it works.the channel erased. but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan. <extension name="incoming-fxo-channel-1"> <condition field="source" expression="mod_openzap"> <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> <action application="bridge" data="sofia/maqian/6789(a)njpbx.vicp.net"/> </condition> </extension> restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR M.Q 2009-06-04 god.nirvana

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Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99"
by Bruce McAlister 04 Jun '09

04 Jun '09

Hi All, I am trying to build FS 1.0.4pre8 for Solaris 10 (Update 5), however the build fails with the following error: /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool --silent --mode=compile /usr/bin/cc -g -DHAVE_CONFIG_H -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE -I./include -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix -I./include/arch/unix -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch locks/unix/thread_mutex.lo "/usr/include/sys/feature_tests.h", line 336: #error: "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" cc: acomp failed for locks/unix/thread_mutex.c make[2]: *** [locks/unix/thread_mutex.lo] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make: *** [libs/apr/libapr-1.la] Error 2 I have the JDS-CBE build environment setup as recommended on the wiki with Sun Studio 12 compiler suite installed as well. I have tried building using Sun's compiler and GNU compiler but I get the same error message. I just recently tried "bootstrap.sh" prior to "configure", "make" but the error is still the same. Would someone have any suggestions for me to try to get around this? Thanks Bruce

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Error causing freeswitch to crash
by Andy Ayers 04 Jun '09

04 Jun '09

Hi, Every few days I'm getting this error which is causing Freeswitch to crash. Can anyone tell me what may be causing this or how to prevent it? 2009-05-25 10:37:38 [CRIT] switch_core_state_machine.c:263 handle_fatality() Caught signal 11 for unmapped thread! Many thanks Andy

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(no subject)
by zhaoxxqq 04 Jun '09

04 Jun '09

zhaoxxqq pisze: > 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 Do you use 'auth ClueCon' before sending 'api' command? Szymon I have not use 'auth Cluecon' before sending api command. I send other api have no problem.only play wav have problems 2009-06-04 zhaoxxqq

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Attendant transfer problem with a Cisco phone and Freeswitch.
by Durk de Beer 04 Jun '09

04 Jun '09

I've come across a problem when using Cisco phones as sip-clients on a Freeswitch system. The problem that arises has to do whit which RFC the Cisco phones are following. Best thing to do at this moment is to point to the Tech-invite website http://www.tech-invite.com/. The examples on that site are more explanatory to the problem than I am ever are able to provide. Ok if you take a look at the link SIP Service Examples there are 19 examples of how RFC 5359 describes how call signalling should flow. If you take a careful look at example 5 (attendant transfer) you will discover that before there is the transfer the station the transfer will go to is put on hold. Bob is transfering to Carol so she is being put on hold by Bob, signals 12 to 14. Now if you take a look at the RFC involved in transfering calls, to be found on the main site link SIP-Topics (to the left) and then following the link Call Transfer (middle window second line in yellow section) you'll find three RFC regarding to transfering calls. In none of these am I able to find this putting on hold as is scribed in RFC 5359 so it seems reasonably to assume that this 'putting on hold' in not mandatory. Here now arises the source of my problem, the Cisco phones are using RFC 5359 when attempting a attendant transfer. When now this signalling is to flow through Freeswitch it puts the station where the transfer is going to on hold as in the prior example is happening to Carol. Freeswitch the connects the MOH stream to this station. Seems a logical thing to do if your assume the putting on hold is not mandatory for the transfer. When now the original station (prior example Bob) is sending the refer thins go bad. Freeswitch is not sending new invites or other signaling to the stations. It is only processing the byes from the station that preforms the transfer (prior example Bob). If I now break the MOH stream on the freeswitch cli all goes well meaning that all invites and other signalling is flowing correctly through Freeswitch. Did anyone out there have the same problem or better yet have a fix for it?. Kind regards, Durk

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Problem about displayname of a routing call
by Brad Tuan 04 Jun '09

04 Jun '09

Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is "Extension 97730002"?? EVENT DUMP: Channel-State: [CS_ROUTING] Channel-State-Number: [2] Channel-Name: [sofia/internal/sip:97730001@210.68.184.192:62101 ;rinstance=16b8076934af7da9] Unique-ID: [342618e3-84cd-494b-b745-760b60639924] Call-Direction: [outbound] Answer-State: [ringing] Caller-Username: [97719006] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Extension 97730002] Caller-Caller-ID-Number: [97730002] Caller-Network-Addr: [163.28.32.51] Caller-Destination-Number: [sip:97730001@210.68.184.192:62101 ;rinstance=16b80769 34af7da9]

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Re: [Freeswitch-users] Prolems with DTMF
by Rudolf Denert 04 Jun '09

04 Jun '09

Ok, another question. Is there a possibility to "detecte" which DTMF mode is sent by the calling CPE so that I can establish logical interrogation in my configuration? ----- Ursprüngliche Mail ----- Von: "Brian West" <brian(a)freeswitch.org> An: freeswitch-users(a)lists.freeswitch.org Gesendet: Mittwoch, 3. Juni 2009 17:39:42 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Prolems with DTMF NO. You should only have one at a time. Its impossible to handle the scenario where you receive say on 2833 and then end up sending an info packet, 2833 packet and the inband of the DTMF... triple DTMF. :) /b On Jun 3, 2009, at 10:35 AM, Rudolf Denert wrote: Is there a possibility to activate all DTMF detection modes (in-band, SIP INFO & RFC 2388) in the same dialplan or maybe in the same extension of the dialplan? Greetz Brian West brian(a)freeswitch.org -- Meet us at ClueCon! 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 -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert(a)tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enthält vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet.

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Considering jumping ship from Asterisk
by Joseph L. Casale 04 Jun '09

04 Jun '09

I have been lurking for a while here looking for an alternative to asterisk that will still work with my tdm card. Although my sip providers are written out in my hosts file and I have a caching dns server, when our internet connection dies, slowly the internal sip phones loose reg and the "redundant" dahdi line we have for this very scenario doesn't help as asterisk gets its sh#t in a knot and tanks? How does freeswitch handle this scenario? Is it prone to this kind of failure as well? Thanks for any advice! jlc

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Freeswitch taking too long to start up
by Muhammad Shahzad 04 Jun '09

04 Jun '09

Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, "make current" command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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