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inbound concurrent calls per provider
by shayne.alone＠gmail.com 25 Feb '13

25 Feb '13

Hi all; where should i define the maximum number of concurrent calls which is allowed from each of my providers? * I had about 30 providers which are defined under directory each of witch have more that one origination source address. example: <user id="998" cidr="XXX.XXX.XXX.201/32,XXX.XXX.XXX.202/32"> <variables> <variable name=accountcode value=998/> <variable name=user_context value=CallPlan/> </variables> </user> -- Regards, Ali R. Taleghani

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FreeSWITCH suddenly core dump
by Vittorio Guglielmo 25 Feb '13

25 Feb '13

Hello everybody, I'm new to freeswitch (I recently switched form Asterisk). This evening, after one week of work (I have only 3 accounts enabled), freeswitch suddenly core-dumped. Now, every time I try to start it, it pushes a core dump. I read the wiki about procedures to gain back traces, and used the fscore_pb from the support-d of the source-dir, and here is the URL it gave to me for you: http://pastebin.freeswitch.org/20623 Thanks for your help Vic

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How to enable dtmf prompt while in callcenter queue?
by Wilsonbrett85 25 Feb '13

25 Feb '13

So I've got my callcenter setup working, using the following queue position announcement script: http://pastebin.com/sP3Gdq2g My dialplan looks like this: <action application="set" data="tts_engine=flite"/> <action application="set" data="tts_voice=kal"/> <action application="set" data="result=${luarun(callcenter-announce-position.lua ${uuid} support(a)***.net 15000)}"/> <action application="callcenter" data="support(a)***.net"/> I'm wondering if it's possible to allow the caller to press a key during the announcements to exit the callcenter queue and go straight to voicemail? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-enable-dtmf-prompt-whi… Sent from the freeswitch-users mailing list archive at Nabble.com.

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mod_cdr and MySQL
by Sayyed Mohammad Emami Razavi 25 Feb '13

25 Feb '13

Can i use some modules like mod_cdr_mongodb for MySQL?

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conference-flags
by Emrah 25 Feb '13

25 Feb '13

Hi all, Where can I find some explanations on how to use conference-flags? How do you use video-bridge and video-floor and what do they do? Thanks for any pointers. E

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CCS_RINGING too early
by Hynek Cihlar 24 Feb '13

24 Feb '13

When originating a call the respective call channel's call state is set to RINGING right after progress 100 is received. Here's the captured flow: |Time | <src ip> | | | | <dst ip> | |2.848 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.848 | 407 Proxy Authentication Required |SIP Status | |(5080) <------------------ (5060) | |2.849 | ACK | |SIP Request | |(5080) ------------------> (5060) | |2.849 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.850 | 100 Trying| |SIP Status | |(5080) <------------------ (5060) | |13.444 | 180 Ringing |SIP Status | |(5080) <------------------ (5060) | |13.445 | 183 Session Progress SDP (g711A g711U GSM tele...ne-eventRTPType-101) |SIP Status | |(5080) <------------------ (5060) | |13.445 | RTP (g711A) |RTP Num packets:230 Duration:4.574s SSRC:0x1E777E26 | |(26056) <------------------ (19312) | |13.601 | RTP (g711A) |RTP Num packets:220 Duration:4.419s SSRC:0xA530C519 | |(26056) ------------------> (19312) | After the 100 Trying is received the switch executes switch_channel_perform_set_running_state (switch_channel.c) and the channel call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is generated. I would expect the channel call state to be set to RINGING only after 180 Ringing is received from the far endpoint. Could anybody give me a hint what could be wrong or what steps to take to figure out? I am already out of ideas. Thanks! Hynek

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outbound calls control
by shayne.alone＠gmail.com 24 Feb '13

24 Feb '13

I need to control over the return value of bridge application: How should on find if the call was rejected bye Carrier_One ( before I fill 10 calls which I am permitted to ) and route then via Carrier_Two? <include> <extension name="noPerfixed"> <condition field="destination_number" expression="^([0-9]+)$"> <action application="set" data="ignore_early_media=true"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION"/> <action application="limit_execute" data="hash outbound carrierOne 10 bridge sofia/gateway/xxx.xxx.xxx.101/$1" />  <action application="hangup"/> <action application="bridge" data="sofia/gateway/xxx.xxx.xxx.102/12$1"/> <action application="hangup"/> </condition> </extension> <extension name="Perfixed"> <condition field="destination_number" expression="^[0-9]+#([0-9]+)$"> <action application="transfer" data="$1 XML CallPlan"/> </condition> </extension> </include> -- Regards, Ali R. Taleghani

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Short word drop-outs
by Alex Lake 23 Feb '13

23 Feb '13

Got a customer who's complaining that short words are being lost in conversations over VOIP/Freeswitch. I presume this is over-enthusiastic silence suppression which could be tweaked in the freeswitch config. I wondered if this might be a handset config issue, but they claim that they can use their handsets with other VOIP providers that don't show the same problem. Any tips?

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Possible to avoid sending callcenter call to agent if they are on the line?
by Wilsonbrett85 23 Feb '13

23 Feb '13

When a call comes in through the callcenter queue and an agent answers, naturally another call coming in will NOT ring their phone. But what if I want to have this same behavior if an agent calls a local extension or calls out the gateway? So if an agent is on the phone at all, they are skipped over for incoming queue calls. Is it possible? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Possible-to-avoid-sending-cal… Sent from the freeswitch-users mailing list archive at Nabble.com.

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lua and playAndGetDigits with tts
by Wilsonbrett85 23 Feb '13

23 Feb '13

FreeSWITCH Version 1.3.13b+git~20130208T015804Z~7989798ec4 (git 7989798 2013-02-08 01:58:04Z) Not sure if this is a bug: in lua: session:set_tts_parms("flite", "kal") acct_num = session:playAndGetDigits(1, 5, 3, 3000, "#", "say:Please enter your account number now, followed by the pound key.", "say:Your entry was incorrect.", "\\d+") I get the following in the log and the tts does not work: 2013-02-21 21:15:20.736283 [ERR] switch_ivr_play_say.c:1207 Invalid Args The only way to get it to work is to call the script from an ivr with tts-engine and tts-voice set, or call after the dialplan like so: <action application="set" data="tts_engine=flite"/> <action application="set" data="tts_voice=kal"/> <action application="lua" data="shell_demo.lua"/> Why does session:set_tts_parms("flite", "kal") not function as expected? Interestingly I have no issues with session:speak in the same use case. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/lua-and-playAndGetDigits-with… Sent from the freeswitch-users mailing list archive at Nabble.com.

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