- FreeSWITCH-users - FreeSWITCH Mailing Lists

Strange FROM in SIP header
by Dmitry Mordovin 11 Sep '17

11 Sep '17

Hello, I set ANI number like this: [origination_caller_id_name="1234567",origination_caller_id_number=1234567] in bridge. Then in trace I see Invite message where From looks like /From: "1234567" <sip:FreeSWITCH@x.x.x.x>;tag=ypt0Kt3Q0em7j/ Anybody knows why *sip:FreeSWITCH* instead *sip:1234567* ? BR Dmitry

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use of information with query 'sofia status' and locks in internal interface
by Miguel Jesús López Valverde 11 Sep '17

11 Sep '17

Hello, good afternoon. I send this email to see if anyone can give me information about the meaning of the values (2) after the RUNNING state obtained on the query sofia status. I searched for forums and books, but I find nothing explanatory. freeswitch@ip-xxx-xxx-xxx-xxx> sofia status Name Type Data State ============================================================================ ===================== external profile sip:mod_sofia@xxx.xxx.xxx.xxx:5080 RUNNING (0) external profile sip:mod_sofia@xxx.xxx.xxx.xxx:5081 RUNNING (0) (TLS) internal profile sip:mod_sofia@xxx.xxx.xxx.xxx:5060 RUNNING (2) internal profile sip:mod_sofia@xxx.xxx.xxx.xxx:5061 RUNNING (2) (TLS) ============================================================================ ===================== 2 profiles 0 aliases It happens sometimes that this value rises above 2 in the internal interface, I do not know if it is casual, but it blocks that interface and does not respond to sip requests. To identify this problem I have traversed the log files and nothing appears that indicates what may be happening and it becomes necessary to pull and lift the freeswitch service to re-handle the sip requests correctly. Also, from console, if the sofia status profile internal command is executed, no response is obtained and the query must be canceled. Thank you very much. <http://www.smartic.es/> Miguel J. López Valverde Dpto. Técnico. _____ Pº de la Castellana, 135 7ª pl 28046 Madrid España Tel: 900 900 368 | Móvil: (+34) 667 772 911 E-mail: <mailto:mjlopez@smartic.es> mjlopez(a)smartic.es | Web: <http://www.smartic.es/> http://www.smartic.es _____ <https://www.linkedin.com/company/consulting-smartic-solutions-s.l.> <https://twitter.com/smartic_spain> <https://www.facebook.com/cosutiesmatisolutio?ref=ts> Sus datos de carácter personal (nombre, apellidos, dirección postal y de correo electrónico, etc.) son tratados para la gestión de su relación con la Entidad, así como para el envío de información sobre nuestra actividad y la de terceros relacionadas con la actividad de Consulting Smartic Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación, cancelación y oposición dirigiéndose por escrito, con copia de un documento que acredite su identidad, a la dirección info (arroba) smartic.es. Este mensaje puede contener información confidencial. Si usted no es su destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la información que contiene. En este caso, por favor, llámenos o comuníquenoslo por escrito y borre este mensaje de su sistema --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus

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feature access code for call transfer
by Ketan Kothari 11 Sep '17

11 Sep '17

Hello there, In Asterisk i'm using *2 as call transfer and its working fine. *Is there any feature access code for call transfer for manually?* I have tested call transfer using button which available in soft-phone and its working fine. But i want to transfer manually without using button. Any suggestion would be helpful.

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xmlrpc originate failure processing
by Bayani Carbone 11 Sep '17

11 Sep '17

Hi all, I'm using xmlrpc with php to send the originate command to freeswitch to initiate a call to leg-a. This command also specifies the dialplan in which I use the bridge command to make a call to leg-b and bridge both legs. xmlrpc command: originate {originate_timeout=120,origination_caller_id_name='+xxxxx',origination_caller_id_number=+xxxxxx,call_ref='xxxxxxx',bCallerIDName='+xxxxxxx',bCallerID='+xxxxxx',bnumber='+xxxxxxx'}sofia/gateway/siplb-01/+xxxxx|sofia/gateway/siplb-02/+xxxxxx call XML public dialplan: <extension name="public_extensions"> <condition field="destination_number" expression="^call$"> <action application="set" data="api_hangup_hook=lua eoc.lua ${call_ref}"/> <action application="bridge" data="{origination_caller_id_name=${bCallerIDName},origination_caller_id_number=${bCallerID}}sofia/gateway/${default_provider}/${bnumber}|sofia/gateway/${secondary_provider}/${bnumber}"/> </condition> </extension> My problem is that if the originate command results in a call setup failure, I don't enter the dialplan and I can't retrieve the failure in php either. I tried to add the following to the originate command: transfer_on_fail='auto_cause error XML public' With corresponding dialplan entry but it did not work either. I need to be able to execute eoc.lua even on failure of leg-A. Any help is welcome! Thanks

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Re: [Freeswitch-users] TDM-to-TDM Pass-through Media
by Mehdi Shirazi 10 Sep '17

10 Sep '17

Hi Regarding Freeswitch shares same DAHDI and libpri with Asterisk, if there was no reason to call comes inside server ( like codec conversion ,inband DTMF processing , call waiting)... Theoretically this two case should be possible: 1-In DAHDI-driven cards supports direct timeslot bridging (interchange)(E1/T1s in same span cards) the audio won't even pass across the system bus at all. http://lists.digium.com/pipermail/asterisk-dev/2010-March/043056.html 2-With another interesting service = 2BCT(Two B-Channel Transfer)(ITU version ECT = Explicit Call Transfer) even it is possible to free both B channels and transfer Call to Local switch( if both sides was on same E1/T1 span) http://support.digium.com/Answers?id=kA080000000GwmUCAS Regards M.Shirazi >bypass and proxy media is only useful for rtp based calls (read sip to sip) you still have yo copy frames >from on t1/e1 port to the other port. >Sent from my iPhone >> On Sep 9, 2017, at 08:35, Nandy Dagondon <nandy1925 at gmail.com> wrote: >> >> Hello folks, >> >> I have a setup wherein I inserted FS in-between a PSTN exchange and a PABX linked via E1 R2. My TDM card >>has dual E1 ports - port A is connected to the exchange and port B to the PABX. If there's a call that pass->>through FS (EXCH > PABX and vice-versa), I want incoming media from one port to immediately sent to the >>other port. What mode should I set in the dialplan? proxy_media or bypass_media? I have read the Wiki on >>proxy_media. It's very clear to me re handling of TDM media. >> >> Thanks in advance for the inputs. >> >> /Nandy

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PortAudio and Windows devices
by Clive Lansink 10 Sep '17

10 Sep '17

Hi again everyone. OK I appreciate that nobody has yet answered this but now I have more info that may help. I succeeded in building the Freeswitch executable for Windows from the source. I git cloned version 1.6.19 which I understand is the most recent stable release. I am using Visual Studio 2015 Express. I first built the debug version 64-bit. There were three errors but at least there was an executable so I ran that. Now the pa devlist command lists my sound devices so that works. Also the debug information indicates that it is using DirectSound. I actually haven't made a call because of another issue but at least I can confirm that mod_portaudio sees my audio devices in this situation. Then I cleaned the solution and built the release version. Again I got three errors but again there was an executable so I ran that. Now the pa devlist command produces no devices. This suggests to me that something is amiss with the process for generating the release version for Windows which also makes the Windows installer, that impacts on PortAudio. In that case, the result In my experience is that PortAudio just doesn't work because it doesn't see any audio devices. It's beyond me I'm afraid to figure out the difference here. I'm just not experienced with Visual Studio to understand what is going on with these two scenarios. Hope this helps. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Clive Lansink <clive(a)lansink.co.nz> To: "Freeswitch users list" <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] PortAudio and Windows devices Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Fri, 8 Sep 2017 10:03:32 +1200 Hi everyone. Can someone please help with the following problem? I have spent several hours on this today, just googling around, and although I think I am not the only one with this problem, I haven't actually found the solution. Problem is PortAudio reports there are no input and output devices. I am running Freeswitch on Windows 10 64-bit. The Freeswitch installer is FreeSWITCH-1.6.18-x64-Release.msi which I downloaded recently from files.freeswitch.org. I should explain that I have been using Freeswitch for a number of years and PortAudio used to work, even on this Windows 10 PC. The problem seems to be with the more recent versions of Freeswitch. We know that Freeswitch for Windows includes PortAudio. That's good because I don't know how to build Freeswitch for Windows. I just know how to install and use it. I think I read somewhere that the Freeswitch PortAudio is built for DirectEx. I came across the Dxdiag tool which tells me about DirectEx devices. I have several input and output devices on this machine, but Dxdiag only lists one output device and it lists no input devices. It tells me I have DirectEx version 12. I do a lot of programming in Python and that includes a Python module that provides bindings for PortAudio. That code still works. I read that PortAudio by default is built for MME, or maybe it is PyAudio that is built for MME. Whatever the case, that code works. But if Freeswitch's PortAudio module is built for DirectEx, could there be a problem with DirectEx on Windows 10? I just get the impression that Microsoft is favouring newer Windows sound APIs. Issuing the pa devlist command just lists nothing so it is as if it is just not seeing any devices, but they are certainly there. I think this has been a problem for several years because I have Freeswitch version 1.5.15 on another Windows 10 64-bit PC and PortAudio on that PC has the same problem. I just haven't had to use it on that PC. By the way, I did try running Freeswitch in console mode with administrator privileges and it made no difference. I hope someone can suggest something to help me fix this problem. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org 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

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Re: [Freeswitch-users] Stop packet radius
by Nicolas Paleari 09 Sep '17

09 Sep '17

Dear Joseph, thanks for the reply, the password is not real, but you're right, I should have deleted the field, it is not necessary. As for the solution I will try and answer the results, thanks El 9 sep. 2017 9:38 AM, "Joseph Waite" <joelists(a)tm.net.uk> escribió: Ok Firstly, you should always redact and sensitive parts of config files before posting to a public user group. And Hacker now has the access credential for your radius server, I would advise changing the password immediately. This line should have had the IP and password blanked out <param name="authserver" value="170.84.252.34:1812:J79RSaEh5dswcGdffyP5rg4u”/> In regards to the issue your having, I would be surprised if it is actually an issue with the radius module as Dmitriy suggests and would suspect that it is the way FS passes the 2 different values used. If you compare the radius packet in your original email, with the radius config file, dst-number-in is using FS variable sip_to_user with a secondary of dialed_extension which is passing the information correctly in the radius packet, however both Called-Station-Id & dst-number-out are using dialed_extension as the variable. A check of the values of the variables at the relevant time would confirm. Assuming I’m correct above the following change should resolve the issue for you. In the xml_radius config file change the following line <param vendor="Cisco" name="Cisco-AVPair" variable="destination_number" format="dst-number-out=%s”/> to look like this <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_user” secondary_ variable="destination_number" format="dst-number-out=%s”/> Also if you need the same for the Called-Station-Id the change the following line <param name="Called-Station-Id" variable="destination_number" format="%s"/> to look like <param name="Called-Station-Id" variable="sip_to_user” secondary_va riable="destination_number" format="%s"/> Hope this helps. Regards On 6 Sep 2017, at 18:39, Phonecall SRL - Nicolas Paleari < npaleari(a)phonecallsrl.com.ar> wrote: Hi, copy the contents of the file: <configuration name="xml_radius.conf" description="Radius XML Gateway"> <auth_invite> <connection name="jerasoft_vcs"> <param name="authserver" value="170.84.252.34:1812: J79RSaEh5dswcGdffyP5rg4u"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/local/freeswitch/ conf/dictionaries/dictionary"/> <param name="seqfile" value="/var/run/freeswitch/radius.seq"/> </connection> <fields> <param vendor="Cisco" name="Cisco-AVPair" variable="ip" format="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" format="src-gw-name=%s"/> <param vendor="Cisco" name="h323-conf-id" variable="Core-UUID" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="ip" format="request-type=number"/> <param name="Called-Station-Id" variable="sip_to_user" format="%s"/> <param name="Calling-Station-Id" variable="sip_from_user" format="%s"/> <param name="User-Name" variable="sip_from_user" format="%s"/> <param name="Digest-Response" variable="sip_auth_response" format="%s"/> <param name="Digest-Realm" variable="sip_auth_realm" format="%s"/> <param name="Digest-Nonce" variable="sip_auth_nonce" format="%s"/> <param name="Digest-Username" variable="sip_auth_username" format="%s"/> <param name="Digest-URI" variable="sip_auth_uri" format="%s"/> <param name="Digest-Method" variable="sip_auth_method" format="%s"/> <param name="Digest-Algorithm" variable="sip_auth_method" format="MD5"/> <param name="Digest-Qop" variable="sip_auth_qop" format="%s"/> <param name="Digest-CNonce" variable="sip_auth_cnonce" format="%s"/> <param name="Digest-Nonce-Count" variable="sip_auth_nc" format="%s"/> </fields> </auth_invite> <auth_reg> <connection name="jerasoft_vcs"> <param name="authserver" value="170.84.252.34:1812: J79RSaEh5dswcGdffyP5rg4u"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/local/freeswitch/ conf/dictionaries/dictionary"/> <param name="seqfile" value="/var/run/freeswitch/radius.seq"/> </connection> <fields> <param vendor="Cisco" name="Cisco-AVPair" variable="ip" format="request-type=user"/> <param vendor="Cisco" name="Cisco-AVPair" variable="ip" format="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" format="src-gw-name=%s"/> <param name="User-Name" variable="sip_from_user" format="%s"/> <param name="Digest-Response" variable="sip_auth_response" format="%s"/> <param name="Digest-Realm" variable="sip_auth_realm" format="%s"/> <param name="Digest-Nonce" variable="sip_auth_nonce" format="%s"/> <param name="Digest-Username" variable="sip_auth_username" format="%s"/> <param name="Digest-URI" variable="sip_auth_uri" format="%s"/> <param name="Digest-Method" variable="sip_auth_method" format="%s"/> <param name="Digest-Algorithm" variable="sip_auth_method" format="MD5"/> <param name="Digest-Qop" variable="sip_auth_qop" format="%s"/> <param name="Digest-CNonce" variable="sip_auth_cnonce" format="%s"/> <param name="Digest-Nonce-Count" variable="sip_auth_nc" format="%s"/> </fields> </auth_reg> <auth_app> <connection name="jerasoft_vcs"> <param name="authserver" value="170.84.252.34:1812: J79RSaEh5dswcGdffyP5rg4u"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/local/freeswitch/ conf/dictionaries/dictionary"/> <param name="seqfile" value="/var/run/freeswitch/radius.seq"/> </connection> <fields> <param vendor="Cisco" name="h323-conf-id" variable_secondary="uuid" variable="originating_leg_uuid" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="uuid" format="h323-call-id=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_network_ip" format="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" format="src-gw-name=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" format="src-number-in=%s" /> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_user" format="dst-number-in=%s" /> <param name="Called-Station-Id" variable="sip_to_user" format="%s"/> <param name="Calling-Station-Id" variable="sip_from_user" format="%s"/> </fields> </auth_app> <acct_start> <connection name="jerasoft_vcs"> <param name="acctserver" value="170.84.252.34:1813: J79RSaEh5dswcGdffyP5rg4u"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="0"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/local/freeswitch/ conf/dictionaries/dictionary"/> <param name="seqfile" value="/var/run/freeswitch/radius.seq"/> </connection> <fields> <param vendor="Cisco" name="h323-call-origin" variable="h323-call-origin" default="answer" format="%s"/> <param vendor="Cisco" name="h323-conf-id" variable_secondary="uuid" variable="originating_leg_uuid" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="uuid" format="h323-call-id=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_contact_host" format="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-gw-name=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-number-in=%s" /> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-number-out=%s" /> <param name="Calling-Station-Id" variable="sip_from_user" variable_secondary="ani" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_host" format="dst-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="destination_number" format="dst-gw-name=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_user" format="dst-number-in=%s" /> <param vendor="Cisco" name="Cisco-AVPair" variable="destination_number" format="dst-number-out=%s" /> <param name="Called-Station-Id" variable="destination_number" format="%s"/> <param vendor="Cisco" name="h323-setup-time"/> </fields> <conditions> <condition> <param var="sip_to_host" regex="^127\.0\.0\.1" anti="true"/> </condition> </conditions> </acct_start> <acct_end> <connection name="jerasoft_vcs"> <param name="acctserver" value="170.84.252.34:1813: J79RSaEh5dswcGdffyP5rg4u"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="0"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/local/freeswitch/ conf/dictionaries/dictionary"/> <param name="seqfile" value="/var/run/freeswitch/radius.seq"/> </connection> <fields> <param vendor="Cisco" name="h323-call-origin" variable="h323-call-origin" default="answer" format="%s"/> <param vendor="Cisco" name="h323-conf-id" variable_secondary="uuid" variable="originating_leg_uuid" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="uuid" format="h323-call-id=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_contact_host" format="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-gw-name=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-number-in=%s" /> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-number-out=%s" /> <param name="Calling-Station-Id" variable="sip_from_user" variable_secondary="ani" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_host" format="dst-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_user" variable_secondary="dialed_extension" format="dst-gw-name=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_user" variable_secondary="dialed_extension" format="dst-number-in=%s" /> <param name="Called-Station-Id" variable="destination_number" format="%s"/> <param vendor="Cisco" name="h323-setup-time"/> <param vendor="Cisco" name="h323-connect-time"/> <param vendor="Cisco" name="h323-disconnect-time"/> <param vendor="Cisco" name="h323-disconnect-cause"/> <param name="Acct-Session-Time" variable="billsec" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable_secondary="progresssec" variable="progress_mediasec" format="pdd-time=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="destination_number" format="dst-number-out=%s"/> </fields> <conditions> <condition> <param var="sip_to_host" regex="^170\.84\.252\.34" anti="true"/> </condition> </conditions> </acct_end> </configuration> Best regards <Firma Nico Phonecall_new.jpg> El 6/9/2017 a las 11:59 a. m., Jospeh Waite escribió: Could you supply a copy of your xml_radius.conf.xml? Might have a clue in there. On 6 Sep 2017, at 12:03, Nicolas Paleari <npaleari(a)phonecallsrl.com.ar> wrote: Hello, change does Freeswitch, I understand why it does a coding, I need to receive #, this is how the carrier calls waiting El 6 sep. 2017 7:08 AM, "Joseph Waite" <joelists(a)tm.net.uk> escribió: > I'm your radius config are you using the same variable to set both? > > If not and they will always be the same, may be worth trying changing to > use the same. That way can figure out if it's the radius module or > freeswitch making the change. > > Joe Waite > > On 5 Sep 2017, at 14:22, Nicolas Paleari <npaleari(a)phonecallsrl.com.ar> > wrote: > > Friends, I need help, I get incorrect information in the stop packet of > raduis that sends freeswitch, in the field dst-number-out the number has > %23 instead of #, I do not understand why it replaces it,dst-number-out should > be equal to field dst-number-in = 492482 # 541151995330 > Send stop package where you see the problem: > > > [2017-09-01 19:03:46,119] DEBUG WORKER 38 - [pkt#175515/ACCT-STOP] > --- START: [pkt#175515/ACCT-STOP] -------------------------------------- > pdd-time => 0 > called-station-id => 492482%23541151995330 > nas-port => 0 > dst-number-out => 492482%23541151995330 > h323-conf-id => cd0f2a0d-4b7f-414c-b5be-d1b937a414db > h323-setup-time => 2017-09-01T15:03:36.328927-0400 > acct-status-type => Stop > h323-disconnect-cause => 10 > dst-gw-ip => 190.210.240.37 > h323-disconnect-time => 2017-09-01T15:03:43.929039-0400 > h323-call-id => 7a34a7be-d643-4f82-a66a-cd1eb0658b6c > src-gw-ip => 190.210.240.37 > dst-number-in => 492482#541151995330 > h323-call-origin => originate > nas-ip-address => 107.170.35.75 > src-number-in => 7680858053 > dst-gw-name => 492482#541151995330 > h323-connect-time => 2017-09-01T15:03:38.268917-0400 > src-gw-name => 7680858053 > acct-session-time => 5 > acct-delay-time => 0 > src-number-out => 7680858053 > calling-station-id => 7680858053 > --- END: [pkt#175515/ACCT-STOP] -------------------------------------- > > I hope you can help me > Thank you > > > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Libre > de virus. www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > 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 > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cl… FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org 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

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mod_avmd doesn't work on parked channels?
by Federico Omoto 08 Sep '17

08 Sep '17

Hi! I'm trying to use mod_amvd to detect voicemail on a parked channel, but the beep is never detected. On the other hand, if I use mod_amvd on a bridged call it's always detected. Is mod_amvd usable on a parked channel? Thanks in advance and best regards. Fede

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Possible enhancement - load mod_logfile way earlier
by Mario G 08 Sep '17

08 Sep '17

While working on an issue, I realized I was missing a lot of the startup log. The reason is that FreeSWITCH is started in the background with -NC and only the log file it writes is all that’s available. Problem is mod_logfile is loaded long after other things and all messages prior to mod_logfile are lost. Loading mod_logfile as soon as possible at a startup would prevent all the lost startup messages. Let me know if you want a jira. Mario G

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Greenswitch ESL for Python
by Michael Avers 08 Sep '17

08 Sep '17

Hello, I'm looking at greenswitch as an alternative Python inbound ESL connector. Other than Italo's company (thanks for releasing this BTW), anyone else has seen success using it in a moderate to heavy call environment? Project is at https://github.com/EvoluxBR/greenswitch Thanks Mike

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