December 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How can I detect an execute failure using ESL?
by Ron McLeod 21 Dec '09

21 Dec '09

When I try and perform an operation on a channel which has gone, an error is returned. How can I detect this using the ESL? execute() and sendRecv() always return 0 (zero) regardless of whether the command returns +OK or -ERR. sendmsg 5d09753c-ede7-11de-85c6-27ab474dd533 call-command: execute execute-app-name: hangup execute-app-arg: UNALLOCATED_NUMBER Content-Type: command/reply Reply-Text: -ERR invalid session id [5d09753c-ede7-11de-85c6-27ab474dd533] Thanks, Ron

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Difference between ESL execute() and executeAsync()
by Ron McLeod 21 Dec '09

21 Dec '09

I don't notice any different in behavior between execute() and executeAsync(). I was expecting that executeAsync() would return right-away, and that execute() would only return after the specified application runs to completion (CHANNEL_EXECUTE_COMPLETE event). Running the sample app below, I see the "About to call execute(playback)" and "returned" displayed one right-after the other, even though the file being played takes about 4 minutes to play-out. Do I have this wrong, or is there something incorrect in my app? APP: #!/usr/bin/php <?php require_once "ESL.php"; $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon'); $eventSocket->events('plain', 'CHANNEL_STATE'); $eventSocket->filter('channel-state', 'CS_ROUTING'); // Wait for new call attempts while($eventSocket->connected()){ $event = $eventSocket->recvEvent(); $serializedBody = $event->serialize(); $listOfLines = toArrayOfLines($serializedBody); $nameValuePairs = toArrayOfNameValuePairs($listOfLines); $uuid = $nameValuePairs['Caller-Unique-ID']; printf("New call from uuid: $uuid\n"); // answer the caller and play announcement $eventSocket->execute('answer', Null ,$uuid); printf("About to call execute(playback)\n"); $eventSocket->execute('playback', '/tmp/ann.wav', $uuid); printf("returned\n"); } ?> DIALPLAN: <?xml version="1.0" encoding="utf-8"?> <include> <context name="mytest"> <extension name="mytest"> <condition field="destination_number" expression="^.*$"> <action application="park"/> </condition> </extension> </context> </include>

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Mod nibblebill deduct money but no hangup at zero and can call without money in database
by Senaka Amarakeerthi 21 Dec '09

21 Dec '09

Dear Sir, I have successfully installed freeSWITCH and it works fine in passthrough mode. I installed nibblebill and it deduct money from the accounts database and it works fine. but I have two problems. 1. Calls can be initiated even though there is a minus value in accounts database 2. Calls doesn't hangup when it goes to minus values. Any answers are greatly appreciated. This is my dialplan: <action application="nibblebill" data="flush"/> <extension name="hangup"> <condition field="destination_number" expression="^(hangup)$"> <action application="playback" data="no_more_funds.wav"/> <action application="hangup"/> </condition> </extension> <extension name="Omega_Out"> <condition field="caller_id_number" expression="^(\d{4})$"/> <condition field="destination_number" expression="^(\d{11})$"> <action application="set" data="nibble_rate=0.0448"/> <action application="set" data="nibble_account=${accountcode}"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> </condition> </extension> This is the configuration file; <configuration name="nibblebill.conf" description="Nibble Billing"> <settings>          </settings> </configuration>

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No Ringing tone when call is forwarded to PSTN
by Peter P GMX 21 Dec '09

21 Dec '09

Hello, we have the following scenario: A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For the called FS user, call forwarding has been enabled to another PSTN extension (B) . Result: The calling party does not hear any ringing tone. Here an Abstract of the SIP protocol we traced (PSTN(A) and PSTN(B) is in fact the same Patton Gateway): PSTN(A)====INVITE===>FS PSTN(A)<===TRYING===>FS FS===INVITE==>PSTN(B) FS<==TRYING===PSTN(B) FS<==RINGING==PSTN(B) PSTN(A)<==PROGRESS===FS FS<===OK======PSTN(B) FS====ACK====>PSTN(B) PSTN(A)<===OK========FS PSTN(A)====ACK======>FS I would expect that FS answers RINGING back to PSTN(A). Instead it only answers SESSION PROGRESS. When PSTN(B) answers, they can hear each other, but there was no ringing tone to PSTN(A) before. Are there any hints to overcome this, besides playing early media to PSTN(A)? Best regards Peter

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Setting Restrictions on Default Dialplan
by Edmar Cruz 21 Dec '09

21 Dec '09

Hi Sir, How can I allow international calling in the dialing plan but for select accounts only? For example i want to restrict 8555555 to call this ip address 182.138.252.12 using the default configuration.. Does this command should be put in the default.xml or in the default folder and the filename is 00_restict.xml? <extension name="dialsample"> <condition field="destination_number" expression="^(8555555)$"\> <action application="set" data="effective_caller_name=${effective_caller_id_name}"/> <action application="set" data="effective_caller_number=${effective_caller_id_number}"/> <action application="set" data="hang_up_after_bridge=true"/> <action application="bridge" data="sofia/default/$1(a)182.138.252.12"/> </condition> </extension> When i tried this command both of them nothing happen 8555555 can call 182.138.252.12 i want it to restrict this account for not calling 182.138.252.12.. Please help.. Thanks, Edmar -- View this message in context: http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26869283p2… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Interfacing to RabbitMQ
by afshin afzali 20 Dec '09

20 Dec '09

Hi, I'll appreciate if someone who has a practice in interfacing FreeSWITCH to RabbitMQ or suggestions could share it to me. Regards, -- afshin

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fs_cli connection error
by Joseph L. Casale 20 Dec '09

20 Dec '09

Trying to setup a new config in the pfSense 1.2.3 final package and when I try to connect to the console I get an auth error? # ./fs_cli -H 10.0.0.1 [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Authentication Error] I tried to search for docs to indicate where one might set the password for this (it never used to have one) but I could only see docs suggesting to provide one, not set one. There is no .fs_cli_conf anywhere. Socketstat shows it listening on 8021... Thanks! jlc

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Dec 20 Global Freeswitch & All Free SW HW Culture meeting - BerkeleyTIP
by john_re 20 Dec '09

20 Dec '09

Hi FreeSwitchers, & Anthony Anthony - Thanks for letting me post the monthly announcement here. :) FSers: We are working toward moving to FS from Asterisk. We welcome you to join the BTIP Global VOIP bimonthly meetings, & if you like, help us get the FS sw running on our server. :) ===== A great December Solstice to you & yours. :) JOIN the Global All Free SW, HW, Culture meeting via VOIP Dec 20 Sunday, 12N-3PM (Pacific = UTC-8) = 3P-6P Eastern = 8P-11P UTC [Jan 2009 meetings: 2nd, 17th - mark your calendar] http://sites.google.com/site/berkeleytip/schedule == WATCH some VIDEOS: Mark Shuttleworth Interview - 10.04 Lucid Larynx Learning from Code History , Andreas Zeller Why does my program fail? Your version history might have the answer. Audio Hardware Enablement Session, UbuntuDevelopersSummit in Dallas Distributed Development, UDS in Dallas Splunk, Jeremy Thurgood CLUG Upstart, Stefano Rivera CLUG Interfacing with the real world, Mark Ter Morshuizen, Marc Welz CLUG Accelerating Graphics; Camp KDE 2009 http://sites.google.com/site/berkeleytip/talk-videos == Join the MAILING LIST & tell us which videos you will watch & why: http://groups.google.com/group/BerkTIPGlobal == JOIN the meeting via IRC & VOIP: Come discuss any & everything, & work on your individual or group projects. HOT TOPICS: Ub or KUb 9.10?, Ubuntu 10.04 plans, Android, Python3000 in 2010? Start on the #berkeleytip irc.freenode.net channel, & we'll help you get your VOIP system up & working. For VOIP SW, & connection info, see: http://sites.google.com/site/berkeleytip/remote-attendance Berkeley meeting LOCATION: Watch the website & mail list for latest details, perhaps at the Berkeley Public Library, or a cafe, due to Free Speech Cafe closed for winter break. http://sites.google.com/site/berkeleytip/ == OPPORTUNITIES to VOLUNTEER or learn new JOB SKILLS for 2010: Help set up our: Mailing list, FreeSwitch VOIP server, website http://sites.google.com/site/berkeleytip/opportunities Inquire & discuss at the meeting. == For Forwarding - You are invited to forward this announcement wherever it would be appreciated.

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Third Party device connectivity from Freeswitch
by Scott Fernandez 19 Dec '09

19 Dec '09

Hi, Is there a way to have the Freeswitch to route the calls to physical device/phone rather than just routing the calls to soft phones like Xlite?. If available, What sort of settings that are required in Freeswitch to communicate with third party applications/hardwares (like PBX) so that the calls be switched to physical devices from Freeswitch? Can anyone help me in this regard?. Thanks, Scott.

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Deleting event name in Filter
by Magesh R 19 Dec '09

19 Dec '09

Dear All, I have filter a Event like the following in Perl. $IVR->{_esl}->filter("Event-Name","CHANNEL_EXECUTE_COMPLETE"); But I don't how to delete that filter by using filter method. Because 'filter' method accepts only two arguments. Could you any one tell me a way to do it? Thanks, Magesh.

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