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British sounds
by Alex Crow 11 Mar '11

11 Mar '11

Does anyone know of either of the following: 1. A source (paid or free) of British English sounds (preferably female voice) for Freeswitch, preferably covering all the same sounds as those supplied. or 2. A script to convert and rename sounds from an Asterisk install to FS? We already paid for British female sounds for Asterisk so if conversion will cover at least voicemail prompts it would be a start. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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tones.conf parameters?
by Johan Helsingius 11 Mar '11

11 Mar '11

Hi! Trying to use freeTDM with FreeSWITCH, but the supplied tones.conf only has setups for US, SG and RU. I found the ones for India on the wiki, but haven't found anything for NL (that I need), nor any other European country. Any pointers to how to configure tones.conf for NL? Any way to use stuff from asterisk? Julf

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Error! cannot get read lock
by Laurentiu Ceausescu 11 Mar '11

11 Mar '11

Hi, I have the following dialplan: <extension name="line1"> <condition field="destination_number" expression="^333$"> <action application="answer"/> <action application="set" data="allow_voicemail=true"/> <action application="set" data="queue=Queue"/> <action application="set" data="brand=110011"/> <action application="playback" data="/usr/local/audio/welcome.wav"/> <action application="set" data="domain_name=$${domain}"/> <action application="erlang_sendmsg" data="freeswitch_media_manager toor(a)127.0.0.1 inivr ${uuid}"/> <action application="erlang" data="freeswitch_media_manager:! toor(a)127.0.0.1"/> </condition> </extension> When I call 333 I can hear welcome message but then the call is dropped. FS logs shows this error: [ERR] handle_msg.c:56 Error! cannot get read lock. [NOTICE] switch_channel.c:812 New Channel sofia/toor.mydomain.org/401@toor.mydomain.org [03ecb2ca-24df-4903-917a-c9720b43a511] [NOTICE] sofia_glue.c:3754 Pre-Answer sofia/toor.mydomain.org/401@toor.mydomain.org! [NOTICE] mod_dptools.c:929 Channel [sofia/toor.mydomain.org/401@toor.mydomain.org] has been answered [ERR] handle_msg.c:56 Error! cannot get read lock. [WARNING] mod_erlang_event.c:1365 Timed out when waiting for outbound pid 1.0.0@freeswitch@127.0.0.1 03ecb2ca-24df-4903-917a-c [NOTICE] switch_core_state_machine.c:189 sofia/toor.mydomain.org/401@toor.mydomain.org has executed the last dialplan instruct [NOTICE] switch_core_state_machine.c:191 Hangup sofia/toor.mydomain.org/401@toor.mydomain.org [CS_EXECUTE] [NORMAL_CLEARING] [NOTICE] switch_core_session.c:1306 Session 4 (sofia/toor.mydomain.org/401@toor.mydomain.org) Ended [NOTICE] switch_core_session.c:1308 Close Channel sofia/toor.mydomain.org/401@toor.mydomain.org [CS_DESTROY] [ERR] handle_msg.c:56 Error! cannot get read lock. This thing is happend only after I updated to last git head (419d7e2335042e2d86c678c0daf86c2d9c867036 /Mar 9). With a older revision (e88b9639624cef4f35901146241f515730b3b118 /Jan 31) everything works fine. Any idea what has been changed? Thanks, Laurentiu

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Limit the size of queue in mod_fifo
by Marc de Corny 11 Mar '11

11 Mar '11

Hi All, does anyone know of a way of limiting the numbers of calls that are held in the mod_fifo queue so that we can deny more calls after having reacded that limit ? thanks Marc

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Conference silent issue for more than 2 callers
by kapil.rastogi 10 Mar '11

10 Mar '11

Hi, Login with 5 subscribers. 1st subscriber logs in, 2nd Subscriber log in and listen each other, 3rd subscribe and it is not listened and can’t listen others. 4th as 3rd and 5th as 3rd and 4th. Currently i m using freeswitch with diastar, that create a bridge between dialogic and freeswitch. Please help me to resolve the issue. Thanks ----- Regards, Kapil Rastogi Telemune Software Solutions P Ltd. kapil.rastogi(a)telemune.net +919013204760 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Conference-silent-issue-for-m… Sent from the freeswitch-users mailing list archive at Nabble.com.

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count
by Sam 10 Mar '11

10 Mar '11

Hello, Whats the command to get the participants count in the conference ? Rwgards Sam

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Not able to install gsmopen module on FC 13
by ovvenkat 10 Mar '11

10 Mar '11

Hi to all, I have followed this wiki page to intall gsmopen module http://wiki.freeswitch.org/wiki/GSMopen#Linux Installed following prerequisites packages # yum -y install alsa-lib-devel alsa-utils then, go to gsmopen directory #/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1 I wanted to compile only SMS support, no voice call. When, I compiled the source code , I am getting following error , #make ...... ...... gsm_unix_serial.cc:89: error: 'strerror' was not declared in this scope make[1]: *** [gsm_unix_serial.lo] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch/ src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1/gsmlib' make: *** [install-recursive] Error 1 When, I enter into following directory and trying the compile the source code, #/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen # make I am getting following error gsmopen_protocol.cpp:5:28: error: gsmlib/gsm_sms.h: No such file or directory gsmopen_protocol.cpp:9:36: error: gsmlib/gsm_unix_serial.h: No such file or directory gsmopen_protocol.cpp:11:30: error: gsmlib/gsm_me_ta.h: No such file or directory gsmopen_protocol.cpp:16: error: ‘gsmlib’ is not a namespace-name ... ... gsmopen_protocol.cpp:3223: error: ‘DCS_DEFAULT_ALPHABET’ was not declared in this scope gsmopen_protocol.cpp:3225: error: ‘DCS_SIXTEEN_BIT_ALPHABET’ was not declared in this scope gsmopen_protocol.cpp:3226: error: ‘bufToHex’ was not declared in this scope make[1]: *** [gsmopen_protocol.o] Error 1 make: *** [all] Error 1 I do not know, what I am doing wrong. Anyone please help me regarding this. -- Regards Venkatesan OV.

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sip auth challenge HACKING ???
by Erkan Ünlü 10 Mar '11

10 Mar '11

Hi FS Users, in last time i see in my console of FS this kind of error messages. [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile 'internal' for [5828@?????????] from ip ???7?.1??.7??.??? i check my config files ever again and again, but today the console is only given this kind of messages. Maybe 20 messages per second. i see the ip that given in the console "from ip xx.xx.xx.xx" i block this ip in my firewall and everything is fine. Now i understand that this a trying to hacking my server. The blocking of the ip is a solution but can not handle this in Freeswitch, because i see this problem sometimes on different FS servers also and in normally the FS server maybe must can handle this problem. For example with automatic black lists if an ip trys more than 20 times with wrong login. So the ip will be banned for 1 hour or so. i'm interesting in if other users have the same problems and ideas in how we can handle this. Kind regards Erkan

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SIP Trunk Registration
by Mark Holloway 10 Mar '11

10 Mar '11

Does FreeSwitch support single phone number registration for a Trunk Group? For example, in Broadsoft Broadworks I can build a Trunk Group with 100 numbers but I identify one number as the Pilot Number. I have a SIP endpoint (such as a Cisco ISR or Adtran TA900) that registers with Broadworks using the main Pilot Number only (Business Contact) so I don't have to register all 100 numbers. It's like a parent-child relationship where the other 99 numbers are a child of the main parent (pilot) number. Thanks, Mark

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mod callcenter
by Sam 10 Mar '11

10 Mar '11

Hello, Was testing callcenter module and found out that at times it gives error " invalid application callcenter " and after reloading the module it works fine. Some time also happens that if I reload the module it do not reloads the parameters of the callcenter like the agents & tires. It just unload & reloads even if there are changes to the specifications of agents. Regds Sam

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