- FreeSWITCH-users - FreeSWITCH Mailing Lists

sending custom events
by Niall Crosby 22 Feb '10

22 Feb '10

Dear List, Can someone give an example of sending a custom event use the Event Socket Layer directly, and not one of the wrapper APIs? I'm writing in Java and have my own ESL wrapper implementation, but can't work out how to send custom events. Thanks in advance, Niall. -- -- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the sender. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer.

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Freeswitch and mod_opal
by Gilbert Amar 22 Feb '10

22 Feb '10

Thank you Tihomir We tried to build the svn trunk of mod_h323 and succeded but got a core dump a the first call hanging up. >mod_h323 >you need to build h323plus & ptlib before you build the module itself. >check http://wiki.freeswitch.org/wiki/Mod_h323 it works out of the box. >regarding you call setup, faststart = true, h245tuneling = true, >h245InSetup = false >It has to work.... if not post the logs here. >T.

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mod_nibblebill
by phunk000 22 Feb '10

22 Feb '10

Hello there! I am using the fusionPBX freeSWITCH installation. I am attempting to setup the mod_nibblebill module to keep track of billing using a database. I have installed ODBC, mod_spidermonkey, and nibblebill appears to be working, except for the blank line after "last successful billing time was____" based on the freeSWITCH log as follows: 2010-02-22 08:46:55.535673 [DEBUG] switch_ivr_bridge.c:131 sofia/internal/sip:3008@192.168.15.176:42866 receive message [DISPLAY] 2010-02-22 08:46:55.555674 [DEBUG] switch_ivr_bridge.c:131 sofia/internal/3007(a)192.168.15.177 receive message [DISPLAY] 2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:529 Received request via SESSION_HEARTBEAT! 2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:397 Attempting to bill at $1.0 per minute to account 3007 2010-02-22 08:47:23.735442 [INFO] mod_nibblebill.c:447 Beginning new billing on 68cde83c-26f4-44e3-8060-4188a106ff51 2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:455 28 seconds passed since last bill time of 2010-02-22 08:46:55 2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:461 Billing $0.474338 to 3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 0.000000 so far) 2010-02-22 08:47:42.701637 [DEBUG] switch_core_sqldb.c:111 Dropping idle DB connection db="core";user="";pass="";thread="3070618512" 2010-02-22 08:47:42.701637 [DEBUG] switch_core_sqldb.c:111 Dropping idle DB connection db="sofia_reg_internal-ipv6";user="";pass="";thread="3070618512" 2010-02-22 08:47:42.701637 [DEBUG] switch_core_sqldb.c:111 Dropping idle DB connection db="sofia_reg_external";user="";pass="";thread="3070372752" 2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:529 Received request via SESSION_HEARTBEAT! 2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:397 Attempting to bill at $1.0 per minute to account 3007 2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:449 Last successful billing time was 2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:455 30 seconds passed since last bill time of 2010-02-22 08:47:23 2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:461 Billing $0.502331 to 3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 0.474338 so far) 2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:529 Received request via SESSION_HEARTBEAT! 2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:397 Attempting to bill at $1.0 per minute to account 3007 2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:449 Last successful billing time was 2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:455 30 seconds passed since last bill time of 2010-02-22 08:47:53 2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to 3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 0.976670 so far) 2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:529 Received request via SESSION_HEARTBEAT! 2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:397 Attempting to bill at $1.0 per minute to account 3007 2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:449 Last successful billing time was 2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:455 30 seconds passed since last bill time of 2010-02-22 08:48:23 2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to 3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 1.478668 so far) 2010-02-22 08:49:12.271231 [NOTICE] sofia.c:329 Hangup sofia/internal/3007(a)192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING] Any help in regards of how to get nibblebill to access the database properly would be great. Again I have installed ODBC and created the accounts table with the required fields. Thanks a ton ----- Todd who is to learn -- View this message in context: http://n2.nabble.com/mod-nibblebill-tp4612298p4612298.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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Dialplan question
by Matthew Law 22 Feb '10

22 Feb '10

I have FS installed and I can make outgoing calls through my SIP provider. I can also call other extensions (FS is running on a small Xen domU on the internet), but I am having problems getting the dialplan for incoming calls to work. What I want to do is have incoming calls on my number ring all extensions, e.g. 1000 - 1005 for 10 seconds and then go to voicemail for extension 1000. If there are no logged-on users, then it should go straight to voicemail. Rather than bite off too much, I thought I would try and get a very basic setup working and take it from there... At the moment it goes straight to voicemail for extension 1000 even if 1000 is logged in. Here are the dialplan files I have (everything else is default from the trunk install): dialplan/public/00_inbound_did.xml: <include> <extension name="08433xxxxxxx"> <condition field="destination_number" expression="^(0843xxxxxxx)$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> </include> dialplan/default/12_voiptalk.xml: <include> <extension name="public_did"> <condition field="destination_number" expression="^(0843xxxxxxx)$"> <action application="transfer" data="1000 XML default"/> </condition> </extension> <extension name="outgoing"> <condition field="destination_number" expression="^9([0-9]+)$"> <action application="bridge" data="sofia/gateway/voiptalk/$1" /> </condition> </extension> </include> I would be very grateful if someone could tell me where I am going wrong. I've been looking at various FS wiki pages for hours as well as the example configs and can't seem to make any headway. My other question is what command should I be run after changing the dialplan? is it just 'reloadxml'? Many thanks, Matt.

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Freeswitch and mod_opal
by Gilbert Amar 22 Feb '10

22 Feb '10

Hello, I try Freeswitch and mod_opal on CenTos and on Windows XP Calling FS IVR from Openphone or a regular H323 phone works But on those two platforms I could not bridge two calls using h323. There is always a mute or deaf leg. I also try to build mod_h323 with no success. Did anyone have tried this on the svn trunk and succeeded. If yes I will be glad to know how you did and what parameters you choose regarding faststart, h245 tunneling, codecs, etc. Gilbert

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Using FS with Asterisk as a PBX
by Ryan Perry 22 Feb '10

22 Feb '10

I'm new to FS. I am trying to get started with implementing a phone system to manage 12+ small companies. I'd planned to use Asterisk, but I've come to understand the problems with it on a larger scale. My question is will I avoid potential problems by using FS to manage 12 Asterisk PBXs? OR is it advantageous to use FS's PBX abilities? Thanks for your opinions and expertise.

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altering callerid
by Brian May 22 Feb '10

22 Feb '10

Ok, now for something maybe a little bit dodgy. How do I alter the callerid for an incoming call? Example, the SIP provider I use provides callerid in the format 613XXXXXXXX, however some of the analogue phones have fixed width displays and cannot display this long number correctly, as such, I would like to change that to 03XXXXXXXX - compatible with what the local telephone company uses. Is this possible? In asterisk I used: exten => number/_61NXXXXXXXX,1,Set(CALLERID(num)=0${CALLERID(num):2}) exten => number/_X.,1,Set(CALLERID(num)=+${CALLERID(num)}) (disclaimer - 2nd line not tested with analogue phones) -- Brian May <brian(a)microcomaustralia.com.au>

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groups
by Brian May 22 Feb '10

22 Feb '10

Hello, Can I add a analogue FXS port to a group that is then accessed using group_call(...)? If so, how? I can only see support for SIP members, as defined in directory/default.xml Thanks -- Brian May <brian(a)microcomaustralia.com.au>

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variable substitutions
by Brian May 22 Feb '10

22 Feb '10

In the sample dialplan, I see the syntax ${...} and the syntax $${...}. Are both these correct? Using eval suggests that the later simply prefix the result with a dollar sign, I am not sure if this intended... examples - copied from random non-sequential lines in default.xml: <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> and for the $${...} syntax: <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="ringback=$${uk-ring}"/> <action application="bridge" data="sofia/$${domain}/2000(a)bkw.org"/> Maybe both mean the same thing? -- Brian May <brian(a)microcomaustralia.com.au>

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outgoing dialplans for australia
by Brian May 22 Feb '10

22 Feb '10

Hello again! Ok, after spending several hours trying to debug why incoming calls didn't work, for two separate issues (2nd time I accidentally set "Do not disturb" on my VOIP client. Duh!), I am feeling slightly lazy... Anyone got a set of rules for outgoing calls (Australia) that they are willing to share? Even if I don't use it directly, it may help me understand this new XML based syntax (I am only really familiar with the Asterisk dialplan syntax). As a specific example of something I am unsure of, is it possible to have it try and dial using a sip provider, and if that fails try the zap port? I don't want it to fall back to the zap port though if the person doesn't answer or is engaged, only if there is a problem with the SIP connection (e.g. Internet connection down). Thanks. -- Brian May <brian(a)microcomaustralia.com.au>

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