April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

LUA directory script
by Boris Kovalenko 21 Apr '11

21 Apr '11

Hello! May somebody help? I want to integrate directory with sql db via LUA script. By I can't authenticate user. Same user defined with xml file authenticates without problems. The FS request is: 2011-04-20 23:15:34.543486 [ALERT] switch_cpp.cpp:1190 key_name name 2011-04-20 23:15:34.543486 [ALERT] switch_cpp.cpp:1190 key_value pbx.tagnet.hn 2011-04-20 23:15:34.543486 [ALERT] switch_cpp.cpp:1190 tag_name domain 2011-04-20 23:15:34.543486 [ALERT] switch_cpp.cpp:1190 section directory 2011-04-20 23:15:34.543486 [ALERT] switch_cpp.cpp:1190 'Event-Name: REQUEST_PARAMS Core-UUID: df5b6d29-49f1-41d3-8eb0-aa7ad7ee611d FreeSWITCH-Hostname: ipats.tagnet.ru FreeSWITCH-IPv4: 80.64.16.19 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2011-04-20%2023%3A15%3A34 Event-Date-GMT: Wed,%2020%20Apr%202011%2017%3A15%3A34%20GMT Event-Date-Timestamp: 1303319734543486 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2102 action: sip_auth sip_profile: ipbx sip_user_agent: SIPPER%20for%20PhonerLite sip_auth_username: 73435230045 sip_auth_realm: pbx.tagnet.hn sip_auth_nonce: 413ca6a6-84b8-41b0-89e4-43c7ed17abad sip_auth_uri: sip%3Apbx.tagnet.hn sip_contact_user: 73435230045 sip_contact_host: 192.168.3.253 sip_to_user: 73435230045 sip_to_host: pbx.tagnet.hn sip_from_user: 73435230045 sip_from_host: pbx.tagnet.hn sip_request_host: pbx.tagnet.hn sip_auth_qop: auth sip_auth_cnonce: 234abcc436e2667097e7fe6eia53e8dd sip_auth_nc: 00000001 sip_auth_response: d388a1da67bdaa9206d0f6430f998c54 sip_auth_method: REGISTER key: id user: 73435230045 domain: pbx.tagnet.hn ip: 192.168.3.253 The response is: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <domain name="pbx.tagnet.hn"> <params><param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables><variable name="accountcode" value="ats001"/> <variable name="domain" value="pbx.tagnet.hn"/> <variable name="domain_name" value="pbx.tagnet.hn"/> <variable name="user_context" value="pre_routing"/> <variable name="v_tagnet_ats_dstport" value="0"/> <variable name="v_tagnet_ats_srcport" value="0"/> <variable name="v_tagnet_context" value="home.tagnet.hn"/> </variables> <groups> <group name="default"> <users> <user id="73435230045"> <params> <param name="password" value="01234"/> </params> <variables> <variable name="toll_allow" value="internal,local,domestic,international"/> <variable name="user_context" value="pre_routing"/> <variable name="v_tagnet_context" value="home.tagnet.hn"/> </variables> </user> </users> </group> </groups> </domain> </document> -- With respect, Boris

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Freeswitch codec usage
by Stephen Wilde 21 Apr '11

21 Apr '11

In order to evaluate the number of necessary G729 codec, I have a question: if there is a sip-to-sip call in G729 and I do a tone_detect, surely it's necessary to do a decode operation. When the detect operation is completed (and I see in the log a message "removed media bug"), for this call the codec resource is released? The codec resource is allocated to the call for entire call duration or only during tone_detect phase? Stephen

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Polycom TLS
by Brian West 21 Apr '11

21 Apr '11

Anyone here using Polycom with TLS on 3.3.1.0933? I have a bug that I can't nail down nor can I get all the data required due to the phone locking up and rebooting before the complete tech dump is sent to syslog. Just want to see if anyone else is having the issue where the phone will lockup/reboot. Thanks, Brian

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Possible Google Voice -> FreeSWITCH directly?
by mazilo 21 Apr '11

21 Apr '11

I have noticed additional e-mail addresses entered into a GMail account will show up on GV Phone TAB as a Google Chat lines. In this case, I wonder if an FS system is hosted on an e-mail server can be used to interact with GV to place/receive GV calls. If not, then perhaps it is now a good project for FS developers to find out what protocol/handshake is used by GV to forward an incoming call to a Google Chat address and create/write a mod_gv to handle GV calls directly on an FS system. Anyone? ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Possible-Google-Voice-FreeSWI… Sent from the freeswitch-users mailing list archive at Nabble.com.

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FS- specify installation path
by fieldpeak 21 Apr '11

21 Apr '11

Can anyone help advise how to specify a path to install the FS, as per the wiki, http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide, (./bootstrap.sh, ./configure, make, make install) it will be always installed at /usr/local/freeswitch. i want to install to another path, e.g. /usr/test... Thanks. Regards, Charles

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More on playing tones - can someone try this please?
by Clive Lansink 21 Apr '11

21 Apr '11

Hi listers. I did more experiments today with playing tones. Right now I don't exactly know what version of Freeswitch I have but I downloaded it as the Windows install about two weeks ago and I am running it on Windows XP. I am working just with the dialplan so it should be possible for others to test and see if you can reproduce the problem. Here is my extension definition: <extension name="test"> <condition field="destination_number" expression="^1998$"> <action application="answer"/> <action application="playback" data="tone_stream://v=-20;%(1000, 500, 1000);L=10"/> </condition> </extension> This should play a 1000Hz tone for 1 second, then pause for half a second, and repeat ten times. There are two problems I notice with this: 1. I get the 1000Hz beep ok but this is followed by two touchtones which in fact are 1 and 0. So it seems like it has interpreted the 10 from l=10 as touchtones, yet the pattern still repeates 10 times. 2. On roughly the second repeat, it gets noticeably faster so subsequent loops still repeat but at a faster speed. I believe I'm following the syntax correctly but in any case it does look like there are problems with the tone generator on Windows. It just isn't behaving like I would expect. Cheers. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150

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Khomp GSM not receiving messages
by Rodrigo Ferrari 20 Apr '11

20 Apr '11

Hello friends, Sorry for my bad english. I installed a khomp GSM board, set up 2 chips, b0c0 and b0c1. At the dialplan I configured this situation: <context name="khomp-00-00"> <extension name="exemplo_1"> <condition field="destination_number" expression="^s$"> <action application="log" data="DEBUG Recieving CALL ..." /> <action application="log" data="De ${caller_id_number}"/> <action application="curl" data=" http://app.cellmidia.com.br/receiver/receiver.php headers post teste=${caller_id_number}&teste2=Chamada"/> <action application="info"/> </condition> </extension> </context> <context name="khomp-sms-00-00"> <extension name="sms"> <condition field="destination_number" expression="^s$"> <action application="curl" data=" http://app.cellmidia.com.br/receiver/receiver.php headers post teste=SMS&teste2=Entrante"/> <action application="info"/> </condition> </extension> </context> If I call it will do what is set to do, but if I send a message, it will not operate, there is no log set or any event marked as happining. Is there some configuration that I miss? Some setting to my khomp board be able to receive messages. Thanks, best regard's. Rodrigo Ferrari.

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Where is mod_spidermonkey_etpan?
by Andrew Keil 20 Apr '11

20 Apr '11

To Freeswitch developers, FYI: I am running the latest Git HEAD build (downloaded today) on Windows XP SP3 with Visual C++ 2010 Express. I am attempting to send an e-mail from inside a javascript file. Looking on your website I found the example: http://wiki.freeswitch.org/wiki/Examples_answermachine This uses: use("etpan"); With the comment: Note: this script uses mod_spidermonkey_etpan which must be compiled and loaded. My question is where is mod_spidermonkey_etpan ?? I cannot find it anywhere inside the freeswitch directories. If there is no solution, is there another way you can recommend to send an e-mail from from within a javascript file? Or even from an XML Dialplan would be OK. Looking forward to your response. Andrew Keil

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FreeSWITCH Conference Call Today - VoIP Abuse/PBX Honeypot
by Michael Collins 20 Apr '11

20 Apr '11

Hello all! Our agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_04_20 At about 2PM EDT (11AM PDT) J. Oquendo will join us to talk about PBX Honeypot and VoIP Abuse projects. We will still start the conference at 1PM EDT/10AM PDT and do our agenda items until J. joins us. Talk to you soon! -Michael

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Disable IP change TRAP?
by Dmitry Sytchev 20 Apr '11

20 Apr '11

Hi All! How do I disable ip change trap and following profile restart? I have two profiles, bound to separate ip addresses on eth0, both SIP and RTP addresses. But when freeswitch detect IP change (about once a hour), it tries to apply changes, but actually nothing changed as profiles are bound to fixed ip addresses. What can be done to make FS ignore this traps or at least not to touch working profiles? -- Best regards, Dmitry Sytchev, IT Engineer

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