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mod_nibblebill
by budi wibowo 20 Oct '10

20 Oct '10

dear all i install mod_nibblebil on centos 5.5 according to steps in http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core odbc.ini config: [fs_connect] Driver = mySQL SERVER = localhost PORT = 3306 DATABASE = fs OPTION = 67108864 Socket = /var/lib/mysql/mysql.sock switch.conf.xml i add this <param name="odbc-dsn" value="fs_connect:root:rootpass" /> if i execute isql fs_connect root rootpass it's the result +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> but i found this error, where's my mistake ? i 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:210 The sql server is not responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR: [unixODBC][Driver Manager]Connnection does not exist ][169] 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:218 The connection could not be re-established 2010-10-20 07:20:40.412241 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:210 The sql server is not responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR: [unixODBC][Driver Manager]Connnection does not exist ][169] 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:218 The connection could not be re-established 2010-10-20 07:20:41.412257 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2010-10-20 07:20:41.412257 [CRIT] switch_odbc.c:210 The sql server is not responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR: [unixODBC][Driver Manager]Connnection does not exist regards budi

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recording into perl script
by Nikolay Kondratyev 20 Oct '10

20 Oct '10

Hi all, Currently i can record a call via <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> But in this way i can only record to a file. I have some experience with perl and mysql. I'd like to record a call into database (mysql or postgre). I hope it's possible via mod_perl... is it? If yes, how can i pass voice into perl script? I guess that i can just do the following <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="perl" data="recorder.pl $${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav" /> So that my perl script recorder.pl will get recorded file name as it's first argument. But when my script starts, will the recording allready be finished? In this way (if it will work at all) i will first record a call in a file, and then put this file into database. Is there a way to write directly into database? Something like <action application="record_session" data=" |recorder.pl" /> ? Thanks in advance, Nikolay.

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Re: [Freeswitch-users] Sofia gateway incoming problem
by Maxim Balabaev 20 Oct '10

20 Oct '10

Problem solved by <param name="extension-in-contact" value="true"/> Thx! 2010/10/20 Maxim Balabaev <balabaev.m(a)gmail.com> > Here is valid registration log: > http://pastebin.com/mhkDuCWG > I really think that the problem can only come from gw+ in fs contact header > > 2010/10/19 Maxim Balabaev <balabaev.m(a)gmail.com> > > Here is asterisk invite dump: >> http://pastebin.com/Me53C1kV >> (I can't give any FS dump because there is just nothing) >> Fs reg dump: >> http://pastebin.com/63z5HPAS >> May be problem is in contact header? I think i have seen this problem >> somewhere... >> 2010/10/19 Brian West <brian(a)freeswitch.org> >> >> Without sip traces of any other valid information we can only guess... >>> >>> /b >>> >>> On Oct 19, 2010, at 9:42 AM, Maxim Balabaev wrote: >>> >>> > I`m having really strange problem. I connect new provider, changed all >>> gateway settings but i can`t take calls. While debuggins external and so as >>> global no invite packet is showing up. I even tryed tcpdump - absolutely >>> nothing. Just pings. Outcoming are ok, register is ok. If I switch to >>> asterisk - no problem occures. Support says that server refuses connection, >>> while another gateway works perfectly. I don`t know what to do... >>> > May be this can be usefull to. Prov have multiple realms on server and >>> accepts only @myprov.com format. While to asterisk invite is thrown with >>> sipid@myip. I tryed changing server hostname to ip, but still nothing. >>> > Here is my conf: >>> > http://pastebin.com/qKqAv38a >>> > Really appritiate any help. >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users(a)lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >

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energy level and conference recording
by covici＠ccs.covici.com 20 Oct '10

20 Oct '10

Hi. I have found that if the energy level is not 0 and I try to record the conference, the recording loses several milliseconds -- maybe up to 20-50 after silence. I don't think this should happen, but let me know and I can file a bug or is there another work around? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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mod_nibblebill hangs up A-Leg and continues dialing B-Leg
by Ash 20 Oct '10

20 Oct '10

Hi All, I am trying to setup mod nibble bill to bill based on the B-Leg. I have followed the wiki but have been unable to resolve my issue. I first had this issue with 1.0.6 and tried the latest git version and can reproduce the fault on both versions. When the customers credit is 0 or below the handset I am using to test gets transferred to the hangup destination and the A-Leg is hungup. After a couple of seconds my mobile will ring and when I answer it there is silence so a one way call leg. As anyone else seen this behaviour? This is my dialplan that I am outputting to XML Curl. <?xml version="1.0"?> <document type="freeswitch/xml"> <section name="dialplan" description="Outbound"> <context name="sip_customerdomain_net_au"> <extension> <condition field="destination_number" expression="^(04xxxxxxxx)$"/> <condition field="${toll_mobile}" expression="true"> <action application="export" data="service_id=113"/> <action application="set" data="effective_caller_id_number=613xxxxxxx"/> <action application="set" data="privacy_hide_name=1"/> <action application="bridge" data="{enable_heartbeat_events=5,nibble_rate=1,nibble_account=113}sofia/gateway/primary/614xxxxxxxxx"/> </condition> </extension> </context> </section> </document> <?xml version="1.0"?> <document type="freeswitch/xml"> <section name="dialplan" description="hangup"> <context name="default"> <extension> <condition field="destination_number" expression="^(hangup)$"> <action application="log" data="INFO Credit Exceeeded"/> <action application="hangup"/> </condition> </extension> </context> </section> </document> Here is my nibblebill.conf.xml <configuration name="nibblebill.conf" description="Nibble Billing"> <settings>   <param name="db_username" value="switch"/> <param name="db_password" value="XXXXXXXX"/> <param name="db_dsn" value="fs_odbc"/>  <param name="db_table" value="services"/>  <param name="db_column_cash" value="credit"/>  <param name="db_column_account" value="id"/>  <param name="global_heartbeat" value="60"/>  <param name="lowbal_amt" value="0"/> <param name="lowbal_action" value="play ding"/>  <param name="nobal_amt" value="0"/> <param name="nobal_action" value="nibble_hangup XML default"/>  <param name="percall_max_amt" value="100"/> <param name="percall_action" value="hangup"/> </settings> </configuration> Thanks in advance. Ash.

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Looking for configuration example for a Linksys PAP2t-NA
by Charles Bujold 20 Oct '10

20 Oct '10

Trying to connect a Linksys PAP2T-NA ATA to Freeswitch and trying to find a configuration example and settings to make it work. If anybody has a suggestion please forward it. Also do I need to upgrade the firmware of the Pap2T to the latest version for it to work with Freeswitch (v5.16)? Thanks cjb

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SPA921 Problem Bad Session Description
by Maxim Balabaev 20 Oct '10

20 Oct '10

I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs: ------------------------------------------------------------------------ recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397: ------------------------------------------------------------------------ INVITE sip:1001@xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx> Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:022@xxx:5060> Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 386 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 /8000 a=rtpmap:0 /8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060 From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx>;tag=ytUD271ypvy6r Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0

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Early media beginning clipped for certain callers
by Mario 19 Oct '10

19 Oct '10

I have replaced what the incoming caller hears (ringing) with a greeting, instructions, ringing. Works great but for some callers, about 1 second is clipped, it seems to do with who is calling: Caller A cell phone & B land line works every time and is never clipped. Caller C cell phone & D land line clips every time. I added up to a 5 second "sleep" before the media to test but it had no effect telling me the other end is clipping the audio when it starts. Any ideas on what I might try? I can try to lengthen the audio but I didn't want to make the working ones listen to extra silence. Thanks! Mario

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Re: [Freeswitch-users] Compile on Windows
by Tim B 19 Oct '10

19 Oct '10

Ok and update... I was able to try to compile on an WinXP x86 from GIT using VS2008. But the following failed.... - Win7 x64, GIT, VS2010 - Win Server 2008 x86, GIT, VS2008 and VC++ Express 2010

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Remove rport from Via header
by Nicolas Brenner 19 Oct '10

19 Oct '10

Hello, I need to either remove the rport parameter from the Via header on an initial invite or set its value to a specific port. I've looked through the mailing archives and the wiki but I haven't found anything. I appreciate any help or pointers I can get, thanks. Nico

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