- FreeSWITCH-users - FreeSWITCH Mailing Lists

Generate SIP INFO inside dialplan
by Grant Bagdasarian 22 Feb '13

22 Feb '13

Hello, Is there a way to generate a SIP INFO message inside the dial plan? If so, how? If not, are there any other ways to accomplish this during a call? Regards, Grant

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Jira does not work any more?
by Dmitry Lysenko 22 Feb '13

22 Feb '13

http://jira.freeswitch.org/ : "If you're experiencing a problem with one of our software products, please fill out, print, and submit the following bug report form *via FAX* to +1-918-420-9002. After an initial assessment by our support department, the resolution of the problem will be assigned to an appropriate engineer." Is this temporary? Thanks.

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Outbound mod event bug
by Stuart Mills 22 Feb '13

22 Feb '13

Hi All, I know some of you may not consider this as a bug, especially if things are done in the correct way this would never happen, but please read on and I'll explain the issue as I was a little surprised at how badly FreeSwitch handled this user error, seen as though it’s such a good piece of software. Basically, a call arrives into Freeswitch and a connection is established to a remote server, then the following relay occurs (in it’s simplest terms) - connect sent into Freeswitch FreeSwitch replies OK myevents sent into FreeSwitch FreeSwitch replies OK answer sent into FreeSwitch FreeSwitch replies OK bridge with arguments sent into FreeSwitch FreeSwitch replies OK Now, everything is perfect up until here and I'd say that as a call switching platform it’s doing exactly what is being asked of it – however here’s the bit most of you would agree is wrong and isn’t a “real” bug - if you reply again with another bridge, be it by mistake or a deliberate attempt to fool FreeSwitch, the system crashes and causes a core dump. I can get this to happen every time, no warning, no nothing, just a crash dump and FreeSwitch dies in front of my eyes. Answers on a postcard please – try not to be too harsh. I am particularly new to FreeSwitch and find it very very good, I was just really surprised to fin a bug that could crash it. Kind Regards, Stuart

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Is this an internal call?
by Steven Schoch 22 Feb '13

22 Feb '13

This must be a FAQ, but I can't find it. In the default dialplan, how can I tell if the call was from an internal phone, or from an external call that was transferred? One page said there was a variable ${internal_call}, but I can't find that anywhere in the source. Should I just look to see if a directory variable, like account_code or user_context is set? -- Steve

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NPDI tel URI parameters appearing in SIP URI
by Victor Chukalovskiy 22 Feb '13

22 Feb '13

Hello, SIP call comes on leg A as following: INVITE sip:6135555555*;npdi=yes*@1.2.3.4:5060 SIP/2.0 ${destination_number} var becomes "6135555555;npdi=yes" On leg B I send the call to gateway using unaltered ${destination_number}: <action application="bridge" data="sofia/gateway/test_gw/${destination_number}"/> I see the following on leg B: INVITE sip:6135555555@5.6.7.8:5060*;npdi=yes* SIP/2.0 So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as if it was a SIP URI parameter. Such behavior does not seem correct. Is this a bug? Found the following implementation discussion: http://comments.gmane.org/gmane.ietf.sip-implementors/21554 Please help to make FS send tel URI parameters unaltered using bridge application and a gateway. WiKi for bridge dial-plan tool has some examples but it does not make use of gateway and I need to use gateway. Thank you, Victor

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FreeSwitch Skype installation and clients startup issue
by Muhammad Shahzad 22 Feb '13

22 Feb '13

Hi Guys, I am trying to setup FreeSwitch for Skype calling, I have followed the steps mention in below URL and my Linux distribution is Centos 5.5 64bits.. http://wiki.freeswitch.org/wiki/Skypiax Face issue while starting the skype clients, pls help me out. [root@FreeSwitch install]# sh /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients .sh ERROR: Module snd_pcm_oss does not exist in /proc/modules ERROR: Module snd_mixer_oss does not exist in /proc/modules ERROR: Module snd_seq_oss does not exist in /proc/modules mknod: `/dev/dsp': File exists insmod: error inserting '/usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko': -1 File exists Fatal server error: Server is already active for display 101 If this server is no longer running, remove /tmp/.X101-lock and start again. bash: /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype101: No such file or directory Fatal server error: Server is already active for display 102 If this server is no longer running, remove /tmp/.X102-lock and start again. bash: /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype102: No such file or directory Fatal server error: Server is already active for display 103 If this server is no longer running, remove /tmp/.X103-lock and start again. bash: /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype103: No such file or directory Fatal server error: Server is already active for display 104 If this server is no longer running, remove /tmp/.X104-lock and start again. bash: /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype104: No such file or directory Fatal server error: Server is already active for display 105 If this server is no longer running, remove /tmp/.X105-lock and start again. bash: /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype105: No such file or directory Regards, Shazi

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mod-lua Vs mod-java
by JP 21 Feb '13

21 Feb '13

Hi, I want to access a webservice that exposes a SOAP interface. I was wondering if this is better to do in Lua or in Java. Any suggestions will be greatly appreciated. Also in general is it better to use mod-java instead of mod-lua for doing advanced IVR functions like database access or web service access? Are there any drawbacks to using java instead of Lua? Thanks, JP

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Rodney
by Rodney 21 Feb '13

21 Feb '13

http://www.fliesenmosaik.eu/xmmxodq/a6ncli091hqz3zx3lqu5irc3r16.x7ussihn6&n… Rodney

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Nat Issue with AT&T Uverse
by Ira Tessler 21 Feb '13

21 Feb '13

We are running Freeswitch in a hosted environment. We have a customer that uses AT&T U-Verse as an ISP and they have theirstandard router. Attached is an example of a call that after about 1600 seconds the session timer re-invite does not get a response and the audio dies. Does anyone have any experience with hosted Freeswitch and A&T U-Verse that can help? Thanks, Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira(a)connectmevoice.com

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nibblebill custom_sql_save
by David Villasmil 21 Feb '13

21 Feb '13

Hello guys, I have the following config for nibbebill: <param name="custom_sql_lookup" value="SELECT ${card_db_column_cash} AS nibble_balance FROM ${card_db_table} WHERE ${card_db_column_account}=${nibble_account};" /> <param name="custom_sql_save" value="UPDATE ${card_db_table} SET ${card_db_column_cash}=${card_db_column_cash}-${nibble_increment} WHERE ${card_db_column_account}=${nibble_account};" /> I see this on the log, which is obviously failing: [UPDATE cards_table SET balance=balance- WHERE pin=12345;] is ${nibble_increment} not the correct variable to use? It looks like it is empty! Thanks! David

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