July 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Getting DTMF during Fifo
by Eli Hayun 06 Jul '10

06 Jul '10

Hi I created a call senter using fifo. I used a LUA script to detect the caller position in the queue, as explained in the wiki. I want to play a file to tell the caller to enter his pin number (or SSN) and I cannot find any way to do that. I have his UUID but all the samples to get DTMF digits are from the session. How do I get DTMF digits if all I have is UUID? Thanks Eli

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fail to bridge 2 openzap channels
by Tony Tin 06 Jul '10

06 Jul '10

Hi, I'm trying to bridge 2 openzap channels, one is inbound call, another is outbound call. Below is the Lua script which originates the outbound call in the inbound call dialplan. If the "transfer" line is commented out, bridge is working fine. I suspect it is about call legs, but I can find the proper way to set then up. Could anyone please help, thanks. ------------------------------------------------------------------------------------- session_orig = freeswitch.Session("OpenZAP/2/a/98877666") while session:getState()~="CS_HANGUP" do os.execute("usleep 500000") if session_orig:answered() then break end end -- need to play message to called party before bridge -- comment out the below line, then bridge is working session_orig:transfer("orig_ext", "xml", "default") freeswitch.bridge(session, session_orig) ------------------------------------------------------------------------------------- Regards, Tony

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how to check whether openzap channel is idle
by Tony Tin 06 Jul '10

06 Jul '10

Hi, Could anyone please tell me that how to check whether a openzap channel is in idle, so I can use it to originate a call. Thanks. Regards, Tony

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Voicemail: can it advance to the next message when no user's input?
by Yehavi Bourvine 05 Jul '10

05 Jul '10

Hello, Our users are used to a voicemail system in which when you hear a voicemail message and do not touch any DTMF for 2 seconds it proceeds to the next message. Can this be done with Freeswitch as well? Thanks, __Yehavi:

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Call-ID format
by Syed Hussain 05 Jul '10

05 Jul '10

Hi, Is it possible to change the Call-ID: on the outbound leg while transferring the call to external service provider ? example froms snippet below -------------------------------------------- 02:23:10.513759 IP (tos 0x0, ttl 239, id 38921, offset 0, flags [DF], proto: UDP (17), length: 311) 64.122.22.22.sip > 66.89.245.66.5080: [udp sum ok] UDP, length 283 E..7. @....]@.A.Y.._.....#8.SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.89.245.66:5080;rport;branch=z9hG4bK9a91UZ2te93Fc From: "ecsvext00" <sip:6003@89.200.141.95>;tag=9rev06ra96D8K To: <sip:0044202929208@64.122.22.22> Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 CSeq: 132984959 INVITE Content-Length: 0 ------------------------------------------ >From above This Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 to s to Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001(a)66.89.245.66 I have tried <action application="export" data="sip_outgoing_call_id=$1(a)x.x.x.x"/> please advise , thank you regards S

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voice mail password
by luixsansan＠hotmail.com 05 Jul '10

05 Jul '10

Hello, I have the following user as 1001.xml <include> <user id="1001" "> <params> <param name="password" value="1234"/> <param name="vm-password" value="0288"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="yomismo"/> <variable name="effective_caller_id_number" value="1001"/> <variable name="mailbox" value="1001"/> </variables> </user> </include> I can call to this extension and if it does not answer I can leave a message in its mailbox, I use the default configuration and voicemail is accessed in extension 4000. If I want to listen to the messages the password I have to dial "1001" not "0288", why is this so?, where is 1001 set as the password for the mailbox?, how do I change the XML configuration files to use 0288 or any other number I may choose? Thank you for your help. Luis.

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Request for suggestions - FreeSwitch Appliance
by Sanjay Arora 05 Jul '10

05 Jul '10

Hello All I am looking for a freeswitch appliance, suitable for SMB. Something that can act as office pbx in addition to providing inbound & outbound VoIP calling services. 50 lines or less. Please advise if you are using any such appliance & your feedback or even if you know of one. Pointers/links/Comments welcome. With best regards. Sanjay.

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Getting WTF1..N messages in the fs log
by Vitalii Colosov 05 Jul '10

05 Jul '10

Hi, I am testing FreeSWITCH on Xen Linux VPS sever. FreeSWITCH Version 1.0.head (git-7566575 2010-05-26 10-45-52 +0200) Linux: Ubuntu 9.10. When I view the log using ./fs_cli I can see periodic messages in the log: WTF1 WTF2 WTF3 WTF4 WTF5 WTF1 WTF2 WTF3 WTF4 WTF2 ... Looks like FreeSWITCH works properly, however these messages make me warn. I never seen these messages on the other servers - could you please advise what can I check? Thank you, Vitalie

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Problems with git installation
by Mike 05 Jul '10

05 Jul '10

Hello, I'm relatively new to Freeswitch - having only really been playing around with it for the last few weeks. Apologies in advance if the following questions are as a result of something I'm doing wrong (more than likely) - I have the book pre-ordered from Amazon - hopefully that will help me a lot ;) A little background context - I already have Freeswitch up and running well on an Amazon EC2 (Fedora) instance. I had to get this up, running and in service very quickly for various reasons, being completely new to Freeswitch I installed it by downloading the file: wget http://files.freeswitch.org/freeswitch-1.0.6.tar.gz And followed the installation instructions. So far that installation has performed exactly as expected. Now I want to install a test and development Freeswitch on another EC2 instance. For this one I thought I'd follow the recommendations to install from the latest git tree. Bear in mind that my main objective here is to have a basic / default / vanilla installation to use for testing / exploring / playing. At this stage I don't want / need anything fancy. The exact commands I ran were: yum install git yum install gcc yum install gcc-c++ yum install autoconf yum install automake yum install libtool yum install ncurses-devel cd /usr/local/src git clone git://git.freeswitch.org/freeswitch.git cd /usr/local/src/freeswitch ./bootstrap.sh ./configure make all install sounds-install moh-install got a coffee... or two... I edited the vars.xml file to have the correct external (elastic IP address) for the RTP and SIP signalling and opened up the right ports in the Security Group (firewall). On starting Freeswitch I see the following errors, but am able to register my two test phones fine. 2010-07-04 02:32:38.211909 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_voicemail.so **/usr/local/freeswitch/mod/mod_voicemail.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.212462 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_valet_parking.so **/usr/local/freeswitch/mod/mod_valet_parking.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.233480 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_spandsp.so **/usr/local/freeswitch/mod/mod_spandsp.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.234111 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_speex.so **/usr/local/freeswitch/mod/mod_speex.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.236968 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_tone_stream.so **/usr/local/freeswitch/mod/mod_tone_stream.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.237129 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/mod/mod_spidermonkey.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.239198 [CONSOLE] mod_local_stream.c:161 Can't open directory: /usr/local/freeswitch/sounds/music/16000 2010-07-04 02:32:38.239337 [CONSOLE] mod_local_stream.c:161 Can't open directory: /usr/local/freeswitch/sounds/music/32000 When I try to dial the Music On Hold from one of the phones I get the following error: 2010-07-04 02:57:24.513520 [ERR] switch_core_file.c:122 Invalid file format [silence_stream] for [2000]! and unsurprisingly given the above errors, trying to dial voicemail gives: 2010-07-04 02:58:55.846720 [ERR] switch_core_session.c:1752 Invalid Application voicemail So guys - please advise - what am I doing wrong? Mike

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Re: [Freeswitch-users] mod_erlang_event problem: REPRODUCIBLE, fix needed
by Timur Irmatov 05 Jul '10

05 Jul '10

Hi, All and Andrew especially! ;-) I have managed to found a reason of my erlang and freeswitch mis-communication. I am receiving a call through 'external' sofia profile. Then I am sending it to my erlang application, *without* answering. The intent is for erlang application to test 'validity' of the call and then bridge it to external destination. Voice path needs to go directly between caller and callee, so I set inbound-bypass-media to true in external profile. When inbound-bypass-media is set to false, I am successfully managing a call through erlang application. When it is set to false, I am seeing 'got pid' and 'exit erlang_outbound_function' in log files, and I cannot manage calls in erlang. So, I guess there is a bug somewhere in erlang_event module or freeswitch itself. What should I do? Open a bug report in jira? -- Timur Irmatov, xmpp:irmatov@jabber.ru

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