December 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

forcing codec on leg b
by Sameer Khan 21 Dec '10

21 Dec '10

Hello guyz, Need your help I want my fs to behave like this when call arrives for a number 1234656789 at fs i call a php file in which i check in the database that what should be the forwarding address of this number. for example the forwarding address is 123456789(a)1.1.1.1 then i return that forwarding address in bridge application in the end i want to define some setting in sip profiles, user directory or wherever so when ever ip is 1.1.1.1 it set the codec on leg b to pcma like it is defined in asterisk as [1.1.1.1] host=1.1.1.1 type=peer disallow=all allow=alaw i want my sip profile to check if network address is 1.1.1.1, it uses a particular codec lets say pcma but if network address is 2.2.2.2 it uses the defult and this i want to define for around 40 ip addresses

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Group calls & voicemail
by Marty Lee 21 Dec '10

21 Dec '10

Trying to get my head around the dialplan in FreeSwitch and can't find a good example of how to do call a group of extensions, but if there is no answer, leave it in a common voicemail. Think of an incoming line to a small business; it rings all the extensions and if nobody answers, the message is left in an 'office' voicemail, rather than an individual. I've got an entry in the public dialplan that maps the incoming number to the extension 'office'; what I currently have for the 'office' extension is below and is a blatant frankenstein job on the Local-extensions entry, but doesn't work. I'll keep looking, but if anyone has any ideas, then feel free to point me in the right direction. m <extension name="office"> <condition field="destination_number" expression="^office$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="ringback=${uk-ring}"/> <action application="set" data="call_timeout=10"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="bridge" data="${group_call(maui_group@${domain_name}+F)}"/> <action application="answer"/> <action application="sleep" data="10000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} 1000"/> </condition> </extension> Notes: 'maui_group' is defined as two test extensions; '1000' is the destination voicemail account I want it to go to if there is no answer or the handsets are offline. ----- Marty Lee e: marty(a)maui-systems.co.uk Technical Director v: +44 845 869 2661 Maui Systems Ltd f: +44 871 433 8922 Scotland, UK w: http://www.maui-systems.co.uk

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CallCenter Agent UUID-Standby
by Antonio Teixeira 21 Dec '10

21 Dec '10

Good Afternoon. I'm leaving Asterisk ( yayyyyyyyyyyyyy) and was curious in trying out the Call Center feature, uuid-standby , now i created a dialplan and I'm able to connect a caller to an agent. Now by my assumption as soon as the caller disconnects the agent should be trow back ( transfer) into the loop (4099 ext.) but instead i get an hangup and the sip client terminates the call. Any ideas ? Sample Dialplan <extension> <condition field="destination_number" expression="^(4099)$"> <action application="set" data="res=${callcenter_config(agent set uuid ${caller_id_number}@${domain_name} '${uuid}')}" /> <action application="set" data="res=${callcenter_config(agent set type ${caller_id_number}@${domain_name} 'uuid-standby')}" /> <action application="set" data="res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'Available (On Demand)')}" /> <action application="set" data="res=${callcenter_config(agent set state ${caller_id_number}@${domain_name} 'Waiting')}" /> <action application="set" data="cc_warning_tone=tone_stream://%(200,0,500,600,700)"/> <action application="answer" /> <action application="playback" data="$${hold_music}"/> <action application="transfer" data="4099"/>  </condition> </extension> Thanks For Your Time António Teixeira

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Lua not playing any wav files
by Marc de Corny 21 Dec '10

21 Dec '10

Hi all, I have run into an issue on something so basic that I must be as simple as enabling a feature somewhere. I have been trying to get lua to play a message from a WAV file. I have tried session:execute("playback", main_msg) and session:streamFile(ivr_invalid_msg) but neither of them play any music to the caller. I tried both to answer and preAnswer the call first but it made no difference. However if I put the same file into the XML dialplan and play it with the commands below I hear the music fine. <action application="set" data="playback_terminators=#"/> <action application="playback" data="${401_sound_file_path}"/> The issue only seems to be from lua when playing any type of wav file and those files are definitelly there as can be read by the XML The error message is below for the execute(playback) command, but nothing can be seen for the 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application playback Requires media! pre_answering channel sofia/external/2031701665@194.0.147.16:5060 EXECUTE sofia/external/2031701665@194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav) But there is no mention of the streamFile command. I have had similar issue with the PlayAndGetDigits command. Is there something that I need to enable in lua so that is can playback messages to the caller. Many thanks to anyone who can help. Marc below is the XML dialplan and lua script as well as the log at the very end. XML DIALPLAN: <include> <extension name="IVR_FROM_MYSQL"> <condition field="caller_id_number" expression="Anonymous" break="never"> <action application="set" data="effective_caller_id_number=0000000000"/> </condition> <condition field="destination_number" expression="^(404)"> <action inline="true" application="lua" data="ivr_mysql.lua ${destination_number:3}"/> <action application="set" data="effective_caller_id_name=${404_tag}"/> <action application="bridge" data="${404_dial}"/> </condition> </extension> </include> The LUA script ivr_mysql.lua is callsed and this is it. -- IVR : PLAY IVR WAV FILES -- Global Variables: local dialstr_prefix = "sofia/gateway/CS2k/" local dialstr_main = "" local breakoutcode = "184" local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/" local ddi = argv[1] -- answer the call session:preAnswer(); freeswitch.consoleLog("info", "All Answered\n"); ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav" main_msg = sound_file_folder .. "default_autoattendant.wav" -- Play with Execute session:execute("playback", main_msg) -- Play with StreamFile session:streamFile(ivr_invalid_msg); dialstr_main = dialstr_main .. dialstr_prefix .. breakoutcode .. "02031701665" session:setVariable("404_dial",dialstr_main) session:setVariable("404_tag","IVR") RELEVANT LOGS : Dialplan: sofia/external/2031701665@194.0.147.16:5060 Regex (PASS) [IVR_FROM_MYS QL] destination_number(4042031956241) =~ /^(404)/ break=on-false Dialplan: sofia/external/2031701665@194.0.147.16:5060 Action lua(ivr_mysql.lua $ {destination_number:3}) INLINE EXECUTE sofia/external/2031701665@194.0.147.16:5060 lua(ivr_mysql.lua 2031956241 ) 2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media 2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP [sofia/external/2 031701665@194.0.147.16:5060] 10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c odec: 8 ms: 20 2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer [soft] 160 b ytes per 20ms 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send payload to 101 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf receive paylo ad to 101 2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP: v=0 o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105 s=FreeSWITCH c=IN IP4 10.5.2.105 t=0 0 m=audio 29900 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer sofia/external/2 031701665@194.0.147.16:5060! 2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544 (sofia/external/2031701 665@194.0.147.16:5060) Callstate Change RINGING -> EARLY 2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel sofia/external/203170166 5@194.0.147.16:5060skipping state [early][183] 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal sofia/e xternal/2031701665@194.0.147.16:5060[BREAK] 2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application playba ck Requires media! pre_answering channel sofia/external/2031701665(a)194.0.147.16: 5060 EXECUTE sofia/external/2031701665@194.0.147.16:5060playback(/usr/local/freeswit ch/sounds/svc_sound_files/default_autoattendant.wav) 2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972 sofia/external/2031701665@ 194.0.147.16:5060destroy/unlink session from object Dialplan: sofia/external/2031701665@194.0.147.16:5060 Action set(effective_calle r_id_name=${404_tag}) Dialplan: sofia/external/2031701665@194.0.147.16:5060 Action bridge(${404_dial}) 2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119 (sofia/extern al/2031701665@194.0.147.16:5060) State Change CS_ROUTING -> CS_EXECUTE 2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal sofia/ external/2031701665@194.0.147.16:5060[BREAK]

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VPS Host suggestions?
by Avi Marcus 21 Dec '10

21 Dec '10

I'm running less than 5 concurrent calls for the next month at least and slowly gaining volume, so I don't need anything fancy. The wiki seems to say that Xen virtualization works with FS. Any recommendations for VPS hosting? I know you get what you pay for, but I'm sure there's inexpensive that's not bad. Linode has a very slick website and feature list, 64bit hosting, and 512mb/200gb bandwidth for $20/month. Any VPS for voip suggestions? Thanks! -Avi Marcus

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loops
by Sam 21 Dec '10

21 Dec '10

Hello, I am trying to get a loop done using if ... then .. return function, using below syntax but getting error on it. if (session:ready()) then return ack() end function ack() session:answer(); session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/zrtp/8000/zrtp-somethings_wrong.wav") end error:--- EXECUTE sofia/internal/7001(a)192.168.2.190 lua(hello.lua) 2010-12-21 14:55:34.724790 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/hello.lua:1: attempt to call global 'ack' (a nil value) stack traceback: /usr/local/freeswitch/scripts/hello.lua:1: in main chunk Any ideas to resolve this. Regards Sam

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No media when bind_meta_app
by Nagalenoj H. 21 Dec '10

21 Dec '10

Dear friends, I've tried using bind_meta_app for recording the call in my ESL script. It works fine as intended, But I face a serious media breakage(a and b leg doesn't hear the opposite leg's voice) for about 5 seconds once the person enters the DTMF to record. ESL script: http://pastebin.freeswitch.org/14734 Freeswitch log: http://pastebin.freeswitch.org/14735 I tried applications other than record_session and I face the same with all applications. Have I done something?! Help me to solve this. -- Thanks.

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members audio conference
by Madovsky 21 Dec '10

21 Dec '10

when the first member creates and enters in a new conference everything is ok. but if a new memeber enters there is no audio in the conference, unless the ivr. I have a very simple conference dialplan like this <extension name="create_conference"> <condition field="${sip_to_uri}" expression="^000(\d{10,15})@$${domain}$"> <action application="set" data="instant_ringback=true"/> <action application="ring_ready"/> <action application="sleep" data="3000"/> <action application="answer"/> <action application="conference" data="$1-${domain_name}@wideband"/> </condition> </extension> I tried to add <action application="export" data="absolute_codec_string=speex@16000k"/> but no success Thanks

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On what condition P-Asserted-Identity is replaced by P-Preferred-Identity?
by Lucian Marginean 21 Dec '10

21 Dec '10

I try to setup P-Asserted-Identity on setup but I get P-Preferred-Identity and I don't understand why Did I do something wrong or some other flags affect the PID feature? <action application="set" data="effective_caller_id_name=647476xxxx"/> <action application="set" data="effective_caller_id_number=647788xxxx"/> <action application="set" data="origination_privacy=screen"/> <action application="bridge" data="[sip_cid_type=pid,origination_caller_id_name=647476xxxx,origination_caller_id_number=647788xxxx]sofia/private/ 905780xxxx@10.10.67.82:5060"/> and invite generated look like this: U 10.10.67.83:5060 -> 10.10.67.82:5060 INVITE sip:905780xxxx@10.10.67.82:5060 SIP/2.0. Via: SIP/2.0/UDP 10.10.67.83;rport;branch=z9hG4bKH8KNm98v4Qv1N. Max-Forwards: 68. From: "647476xxxx" <sip:647788xxxx@10.10.67.83<sip%3A647788xxxx(a)10.10.67.83> >;tag=F1QF1Z26cBQZm. To: <sip:905780xxxx@10.10.67.82:5060>. Call-ID: ff01d92d-86ec-122e-cba2-001143e7e50c. CSeq: 6092608 INVITE. Contact: <sip:mod_sofia@10.10.67.83:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-79ced28 2010-08-22 20-58-52 -0400. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Privacy: none. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 242. X-FS-Support: update_display. *P-Preferred-Identity*: "647476xxxx" <sip:647788xxxx@10.10.67.83<sip%3A647788xxxx(a)10.10.67.83> > Lucian

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questions about VAD and echo cancellation
by joy this 21 Dec '10

21 Dec '10

Dear all: I have questions about VAD and echo cancellation. My FS is Version 1.0.head (git-) under Windows XP. My soft-phone is X-Lite. I use earphones and microphones for sip 1 which means the talking and hearing are separated; on the other hand, I use NB for sip 2 which makes the talking and hearing in the same place. Sip 1 (1000) calls sip 2 (1001) via FS. When I say something via sip 1, the echo will occur, and the echo will only occur on sip 1. How do I cancel the echo? Buy a hardware card or something else? I have two gateways which are Wellgate2644 and ata171m. Please give me some suggestions. When I enabled VAD, I found something strange. In the start of the session, I can hear the noise and echo on sip 1. If I say something via sip 2, the echo and the noise will disappear. Then the echo will not occur on sip 1 only in a short time, about several seconds. The echo and the noise will occur gradually if I say something via sip1. Is it a normal situation? Sincerely yours, Thisjoy.

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