May 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

I have problem in compiling freeswitch with mod_opal
by Sadjad Seyed-Ahmadian 27 May '09

27 May '09

I faced a problem when I want to compile freeswitch with mod_opal. It gives me a compilation error like bellow I used ptlib-2.6.2 and opal-3.6.2. Would someone please help me? Sincerely, Sadjad ________________________________ making all mod_opal make[5]: Entering directory `/root/fs/freeswitch/freeswitch.trunk/src/mod/endpoints/mod_opal' make[6]: Entering directory `/root/fs/freeswitch/freeswitch.trunk/src/mod/endpoints/mod_opal' Compiling mod_opal.cpp... Compiling mod_opal.cpp ... mod_opal.h:151: error: conflicting return type specified for 'virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)' /usr/include/opal/opal/localep.h:267: error: overriding 'virtual ptlib_virtual_function_changed_or_deprecated****** OpalLocalEndPoint::CreateConnection(OpalCall&, void*)' make[6]: *** [mod_opal.lo] Error 1 make[6]: Leaving directory `/root/fs/freeswitch/freeswitch.trunk/src/mod/endpoints/mod_opal' make[5]: *** [all] Error 1 make[5]: Leaving directory `/root/fs/freeswitch/freeswitch.trunk/src/mod/endpoints/mod_opal' make[4]: *** [mod_opal-all] Error 1 make[4]: Leaving directory `/root/fs/freeswitch/freeswitch.trunk/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/root/fs/freeswitch/freeswitch.trunk/src' Making all in build make[3]: Entering directory `/root/fs/freeswitch/freeswitch.trunk/build' +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ ________________________________

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mod_nibblebill question
by Diego Viola 27 May '09

27 May '09

Hi everyone, I'm currently developing a calling card application that uses event socket and mod_nibblebill to bill calls. Well, the question is: can mod_nibblebill disconnect a call when the balance is depleted, or when it reaches 0 cash? The wiki says: "Allow for disconnecting or re-routing calls when balance is depleted." But then at the bottom in "future goals" it says: "We don't yet warn or terminate calls when they've reached their limits." So I'm confused, I also don't see any options in how to set that. In case that mod_nibblebill doesn't support this yet, how hard would it be to add this? I'm willing to do it, I guess it's a variable on the XML config and then read that from the mod_nibblebill.c, and do some logic there. Just wondering if anyone had some experience with this, and if the feature is already there. Thanks, Diego

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Re: [Freeswitch-users] Unicast isn't working
by Артем Васильев 27 May '09

27 May '09

Of course, it will be useful to try it. Artem > Hi Artem, > > Please to see that some of the stuff I wrote is useful to someone..! > > I've written an FS module which will send the audio over - it's more > efficient than using unicast. Let me know if you'd like a copy. > > Cheers -- > > Dave >

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PBX on a (Sheeva|Pogo)Plug?
by Fred-145 26 May '09

26 May '09

Hello I was wondering if someone had successfully ran a PBX on those tiny, $100 devices that run Linux? www.pogoplug.com www.plugcomputer.org I'm thinking of hooking up a Sangoma U100 (http://wiki.sangoma.com/sangoma-wanpipe-usbfxo) to the USB port so it can handle a couple of analog lines. And installing a compact LAMP server so the user can manage the PBX server from a web browser. What do you think? Thank you. -- View this message in context: http://www.nabble.com/PBX-on-a-%28Sheeva%7CPogo%29Plug--tp23731619p23731619… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Sounds order
by Brian West 26 May '09

26 May '09

I'm getting ready for the next sound file order for FreeSWITCH. I have a rather large set of files to be recorded for the zRTP integration if anyone wants to help out. ;) Please contact me off list. I would like everyone to update and try out voicemail and nitpick anything that you feel is wrong there too and let me know so I can have them corrected in this order also. Thanks, Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

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IVR xml dialplan for outbound calls
by Paul Li 26 May '09

26 May '09

There are a few examples in the wiki, showing how to configure IVR for inbound calls. My question lies in whether it is possible to write a dialplan in xml or scripts to configure IVR for outbound calls. Here is a typical scenario (of usage): (1) Call any extension or external endpoints (maybe PSTN numbers); (2) Activate IVR menu once the recipient answers call; (3) Take feedback (DTMF keystrokes or survey recordings). Your assistance is highly appreciated!

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How to work with the debug logfile?
by Dennis 26 May '09

26 May '09

hi, we encounter some small problems withing the past 2 days and we are trying to find out more about the problems. for this we downloaded the debug logfiles written by fs, but we do not manage to filter all log-entries for one single special call. we are using socket outbound and would like to see all entries (inbound/outbound) of one call. is this possible? kind regards dennis

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NOTIFY Messages
by Andrey Nepomnyaschih 26 May '09

26 May '09

Hello everyone, Does anyone know is it possible to disable NOTIFY messages coming out of FreeSwitch for particular endpoint? The reason I'm asking it is because I have a gateway (D-Link DVG-7111S) that reboots when it receives such a packet. Kind Regards, Andrey Nepomnyaschih

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STUN and rport on Polycom phones
by Brian West 26 May '09

26 May '09

It has come to my attention that Polycom hasn't had a business case to support rport and STUN. If you can please kindly email Marek.Dutkiewicz(a)polycom.com and let him know you would like to see STUN and rport support in the polycom products. Its really one of the last things missing in the phone to make it easy to deploy in a NAT env. Thanks, Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

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Help understanding call_timeout in two dialplans
by Larry Marshall 26 May '09

26 May '09

On inbound calls made to 5551212, which call_timeout will be active, 15 or 20? Is it the last hit? Is there a URL which describes all the applications, for example, export in the default.xml? Thanks for this amazing software, Lars In conf/dialplan/public/00_inbound.xml: <include> <extension name="Inbound-5551212"> <condition field="destination_number" expression="^1323(5551212)$"> <action application="ring_ready" /> <action application="set" data="call_timeout=15"/> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="1000 XML default"/> </condition> </extension> </include> In conf/dialplan/default.xml: <?xml version="1.0" encoding="utf-8"?>  <include> <context name="default"> <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <anti-action application="bind_meta_app" data="1 a a execute_extension::dx XML features"/> <anti-action application="bind_meta_app" data="2 a a record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m -%d-%H-%M-%S)}.wav"/> <anti-action application="bind_meta_app" data="3 a a execute_extension::cf XML features"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=20"/> <anti-action application="set" data="sip_exclude_contact=${network_addr}"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/> <anti-action application="answer"/>  <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> </context> </include>

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