October 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_valet_parking: auto reports on wrong leg of call
by Brian Stafford 04 Nov '09

04 Nov '09

Hi all I did a 'make update' to 15289 and I found the auto feature added to mod_valet_parking so I figured I'd try it out. Everything works fine except that the extension number for the parked call is reported on the wrong leg of the call. It'd be good to sort this out as this is a useful feature for us. Regards Brian Stafford

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Is there any license G729?
by Edmar Cruz 04 Nov '09

04 Nov '09

Hi, Is there any available license G729 for freeswitch? I need it to match G729 of Asterisks? If any please help me and instructions how to install this thing. Thanks -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24251951.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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"Can not record session. Media not enabled on channel."
by Maciej Aniserowicz 03 Nov '09

03 Nov '09

Hello, I am using the same set of extensions for testing the system during development, they include XLite, Cisco sip phone and several extensions that just play some audio file. Sometimes, very rarely, this message "Can not record session. Media not enabled on channel." appears on FS console. Like I wrote before, I don't change any codec settings and always use the same set of devices/emulators. What can cause this message? Thanks, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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SIP provider with extern rtp server
by Ivan C Myrvold 02 Nov '09

02 Nov '09

I have used a SIP provider for more than a year. A few days ago, he said he was moving to a new server, and asked me to reconfigure. I did, and everything seemed to work fine, until I did an outgoing call to an external telephone. I found out I had no audio, in neither direction. Incoming calls was working fine. My provider said that the rtp is not going through the sip server, as it did earlier, but now through several other IP's. Do I have to do some special configuration to handle that? Ivan

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Freeswitch in signaling path only
by DJB 02 Nov '09

02 Nov '09

I am wondering why I cannot do as condition#2. For Lua in dialplan, when I have the followings: --WORKING-- (Condition#1) . . session:execute("set","bypass_media=true") session:execute("set","hangup_after_bridge=true") session:execute("set","continue_on_fail=true") . . session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") . . --NOT WORKING-- (Condition#2) Note: FS tries to be in media path and send re-invite. . . session:execute("set","bypass_media=true") session:execute("set","hangup_after_bridge=true") session:execute("set","continue_on_fail=true") . . session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1") session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2") . . Thank you, Dorn B.

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CDR CSV variables
by DJB 02 Nov '09

02 Nov '09

I wonder if I don't want to have b-leg in cdr csv, is there any variables that can give me the actual gateway ip address that is actually went out. For instance, if I have this in my dialplan: <action application="bridge" data="sofia/default/$1@192.168.1.4|sofia/default/$1@192.168.1.5"/> the only value that I can think of from cdr csv is to get remote_ip_last_arg, but it would contains the whole line of both ip addresses. Thank you, Dorn B.

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Freeswitch seems to doesn't reknow dial tone after the first call using OpenZAP (analog spans)
by Albano Daniele Salvatore - Lavoro 02 Nov '09

02 Nov '09

Hi to all, this is my first message here! I'm trying to setup a freeswitch box using 1.0.4 version compiled from sources, ubuntu 8.04.03 lts with lastest updates and lastest zaptel modules/tools compiled from sources. As hardware i'm using an OpenVox A800P. My problem is the following: if i do a call trought openzap it works, but when i hang down and try a second call it doesn't work! Looking with ztmonitor and hearing recorded audio file it seems that freeswitch do the job on the first outgoing call but, on the second, it does nothing because it doesn't reknow dial tone (but it is there). If i call the freeswitch mananged number (i can answer or not, nothing change) i can do another call, but, in any case, the successive call doesn't start. Looking with ztmonitor 1 -v i see that before the first call i get the special dial tone (a continue 425 frequency to state that the line is free to dial sended by the provider) but after the first call is done, before the second, i didn't see any special dial tone ... that appears for less than a sec when freeswitch starts to compose the number. Here logs for first call ------ 2009-10-28 21:38:39.616112 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-10-28 21:38:39.616112 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DIALING 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_INIT 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT 2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz) State Change CS_INIT -> CS_ROUTING 2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT going to sleep 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_ROUTING 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING 2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz CHANNEL ROUTING 2009-10-28 21:38:39.616112 [DEBUG] switch_ivr_originate.c:63 (OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING going to sleep 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:503 (OpenZAP/1:1/xyz) State CONSUME_MEDIA 2009-10-28 21:38:40.475897 [DEBUG] ozmod_analog.c:655 Detected tone DIAL on 1:1 2009-10-28 21:38:40.475897 [DEBUG] mod_openzap.c:1216 got FXO sig 1:1 [TONE_DETECTED] ------ Here logs for second call ------ 2009-10-28 21:39:07.105916 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-10-28 21:39:07.105916 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DIALING 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_INIT 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT 2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz) State Change CS_INIT -> CS_ROUTING 2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT going to sleep 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_ROUTING 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING 2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz CHANNEL ROUTING 2009-10-28 21:39:07.118644 [DEBUG] switch_ivr_originate.c:63 (OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING going to sleep 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:503 (OpenZAP/1:1/xyz) State CONSUME_MEDIA ------ Following my configs! Really thanks for the help, Best Regards, Daniele --- ZAPTEL.CONF --- # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: OPVXA1200/0 "OpenVox A1200P/A800P Board 1" (MASTER) fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxoks=5 # Global data loadzone = it defaultzone = it --- ZT.CONF --- [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 rxgain => 0.0 txgain => 0.0 --- OPENZAP.CONF --- [span zt] number => xyz fxo-channel => 1 number => xyz fxs-channel => 2 number => xyz fxs-channel => 3 number => xyz fxs-channel => 4 [span zt] fxs-channel => 5 --- TONES.CONF --- [it] generate-dial => %(200, 200, 425);%(600, 1000, 425) detect-dial => 425 generate-ring => %(1000, 4000, 425) detect-ring => 425 generate-busy => %(500, 500, 425) detect-busy => 425 generate-callwaiting-sas => %(400,100,425);%(250,100,425);%(150,0,425); detect-callwaiting-sas => 425 --- OPENZAP.CONF.XML --- <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> <param name="hold-music" value="$${moh_uri}"/> </settings> <analog_spans> <span id="1"> <param name="tonegroup" value="it"/> <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"/> </span> <span id="2"> <param name="tonegroup" value="it"/> <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"/> </span> </analog_spans> </configuration>

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Many CS_REPORTING state Zombie session
by Dome Charoenyost 01 Nov '09

01 Nov '09

Dear All, I'm disable all CDR and use nibble_bill (modify) and odbc_query for create CDR and dedcut balance. but i found many zombie channel and can't kill Like a 51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30 18:11:38,1256951498,sofia/external/66848087977@xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx.191,7050,,,XML,public,,,,, I'm using FreeSWITCH Version 1.0.5pre3 (hacked) How to fix ? Dome C.

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Mod_pjsip
by Meftah Tayeb 01 Nov '09

01 Nov '09

Hi my FreeSWITCH friends Today we talked about FreeSWITCH libraries and we found pjsip, which is a lot better than sofia, it can do everything sofia can do and more, it's thread-safe and performs well, I would love to see a mod_pjsip. The library is GPL but they also offer alternative licensing, don't get me wrong, Sofia is great but I think FreeSWITCH will benefit from a thread-safe SIP library. don't say that i need some stuf for free, but we need to make fs a very also we love freeswitch to run in a Embedded PC like the blackfin CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open Source project With much love, DelphiWorld____ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
by Ahmed Munir 01 Nov '09

01 Nov '09

Hi, Thanks for reply, it really helped me. One more thing to ask, how can we make decision against >,<, >=, <= in condition header? Like we use == for action and != for anti-action. Kindly highlight it. > > ---------- Forwarded message ---------- > From: Ahmed Munir <ahmedmunir007(a)gmail.com> > To: FreeSwitch <freeswitch-users(a)lists.freeswitch.org> > Date: Wed, 21 Oct 2009 15:37:15 +0500 > Subject: [Freeswitch-users] Call custom variable in condition > Hi, > > I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I > want to use it in condition, as I'm listing down the configuration below; > > <context name="SIP_incoming"> > <extension name="call-sip-extensions"> > <condition field="destination_number" expression="^(\d+)$"> > <action application="set" data="AUTHENTICATION_STATUS=0"/> > <action application="transfer" data="${AUTHENTICATION_STATUS} > XML Authen_Status"/> > </condition> > </extension> > </context> > > <context name="Authen_Status"> > <extension name="exten-auth-status"> > <condition field="AUTHENTICATION_STATUS" expression="^0$"> > <action application="answer"/> > <action application="playback" data="play.wav"/> > </condition> > </extension> > </context> > > > > > But unfortunately it is not working. Kindly advise me how to do implement > it(Note: I don't want to call script). And one more thing to ask how can I > transfer the values within the same context? > > -- > Regards, > > Ahmed Munir > > > > > ---------- Forwarded message ---------- > From: Ghulam Mustafa <mustafa.pk(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 21 Oct 2009 15:52:24 +0500 > Subject: Re: [Freeswitch-users] Call custom variable in condition > Ahmed, > > you can't use variables set by "set" application within a condition, though > it doesn't make sense. wondering if there is any logic behind this or it's > just a simple missing feature. anyone? > > -m > > Ahmed Munir wrote: > >> Hi, >> >> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I >> want to use it in condition, as I'm listing down the configuration below; >> >> <context name="SIP_incoming"> >> <extension name="call-sip-extensions"> >> <condition field="destination_number" expression="^(\d+)$"> >> <action application="set" data="AUTHENTICATION_STATUS=0"/> >> <action application="transfer" data="${AUTHENTICATION_STATUS} >> XML Authen_Status"/> >> </condition> >> </extension> >> </context> >> >> <context name="Authen_Status"> >> <extension name="exten-auth-status"> >> <condition field="AUTHENTICATION_STATUS" expression="^0$"> >> <action application="answer"/> >> <action application="playback" data="play.wav"/> >> </condition> >> </extension> >> </context> >> >> >> >> >> But unfortunately it is not working. Kindly advise me how to do implement >> it(Note: I don't want to call script). And one more thing to ask how can I >> transfer the values within the same context? >> >> -- >> Regards, >> >> Ahmed Munir >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> 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 >> >> > > > > > > ---------- Forwarded message ---------- > From: Tihomir Culjaga <tculjaga(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 21 Oct 2009 13:13:13 +0200 > Subject: Re: [Freeswitch-users] Call custom variable in condition > consider this: > > > > <context name="SIP_incoming"> > <extension name="call-sip-extensions"> > <condition field="destination_number" expression="^(\d+)$"> > <action application="set" data="AUTHENTICATION_STATUS=0"/> > <action application="transfer" data="${AUTHENTICATION_STATUS} > XML Authen_Status"/> > </condition> > </extension> > </context> > > > > <context name="Authen_Status"> > <extension name="exten-auth-status"> > <condition field="${AUTHENTICATION_STATUS}" expression="^0$"> > <action application="answer"/> > <action application="playback" data="play.wav"/> > </condition> > </extension> > </context> > > > > > > here is one of my dialplan. I'm using execute_extension but it is quite the > same... > > > > <extension name="ServiceLookup"> > <condition field="destination_number" expression="(^300030)(.*)"> > <action application="lookup_service_destination" data="in > ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $ > 1, in ${network_addr}:5060, out red_contact, out authResult"/> > <action application="log" data="INFO ######################## > ServiceLookup ########################\n"/> > <action application="log" data="INFO ######################## > contact = '${red_contact}' ##############\n"/> > <action application="log" data="INFO ######################## > CallerNum = '${caller_id_number:6:16}' ##########\n"/> > <action application="log" data="INFO ######################## > RADIUS auth = '${authResult}' ##########\n"/> > > <action application="execute_extension" data="doRedirect XML > public"/> > </condition> > </extension> > > > <extension name="doRedirect"> > <condition field="destination_number" expression="^doRedirect$"/> > <condition field="${authResult}" expression="^0$|^60$"> > <action application="log" data="INFO ######################## > RADIUS auth OK!!!' ##########\n"/> > <action application="redirect" data="${red_contact}"/> > <anti-action application="log" data="INFO ######################## > RADIUS auth NOK!! ##########\n"/> > <anti-action application="respond" data="403 Forbidden"/> > </condition> > > </extension> > > > > > On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir <ahmedmunir007(a)gmail.com>wrote: > >> Hi, >> >> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I >> want to use it in condition, as I'm listing down the configuration below; >> >> <context name="SIP_incoming"> >> <extension name="call-sip-extensions"> >> <condition field="destination_number" expression="^(\d+)$"> >> <action application="set" data="AUTHENTICATION_STATUS=0"/> >> <action application="transfer" data="${AUTHENTICATION_STATUS} >> XML Authen_Status"/> >> </condition> >> </extension> >> </context> >> >> <context name="Authen_Status"> >> <extension name="exten-auth-status"> >> <condition field="AUTHENTICATION_STATUS" expression="^0$"> >> <action application="answer"/> >> <action application="playback" data="play.wav"/> >> </condition> >> </extension> >> </context> >> >> >> >> >> But unfortunately it is not working. Kindly advise me how to do implement >> it(Note: I don't want to call script). And one more thing to ask how can I >> transfer the values within the same context? >> >> -- >> Regards, >> >> Ahmed Munir >> >> >> >> _______________________________________________ >> 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 >> >> > > _______________________________________________ > 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 > > -- Regards, Ahmed Munir

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