December 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Can FreeSWITCH be setup to look like a SIP carrier?
by Jon Young 25 Dec '10

25 Dec '10

I am trying to determine if using FS would work for what I need. I have a training class where I have 10-15 students with SIP PBXs. I would like each student to have access to a SIP trunk. What I would like to do is have each student Register with FS using a 10 digit number. I have a SIP Trunk group with 6 call paths that I would like to have FS Register to the SIP provider. This will allow them to make outbound calls and I may have inbound calls hit the IVR to select the student. Thanks for any feedback.

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ESL outbound /inbound
by Rafqat . 24 Dec '10

24 Dec '10

Hi, I am writing a calling card app using lua and java. As advised, Java will predominatley do all the billing side of things (via ESL) leaving lua do simple things like asking for the calling card pin number etc. When a call comes in, my lua script answers the call and asks the user for a pin. Instead of querying the DB inline from within lua, I would like my app server to do this (please let me know if this should be done inline instead). I understand my app server (Java ESL inbound socket) can register for a pin checking custom event and I can generate such an event from within my lua script. My lua script would then wait for an appropriate repsonse event: -- Check if pin is valid local event = freeswitch.Event("CUSTOM", "check_pin_request"); event:addHeader("pin_number", digits); event:fire(); -- wait for response con = freeswitch.EventConsumer("CUSTOM", "check_pin_reponse"); con:pop(1); print("event\n" .. e:serialize("xml")); I was wondering if the above is the right way of doing things, or should I be using ESLOutboundSocket and have lua script do something like this instead: session:execute("set", "pin_to_check=12345"); session:execute("socket", "192.168.0.2:8084"); Not sure how lua will be told whether the pin is valid or not in this scenario. I appreciate FreeSWITCH is very flexible and would like to make sure I develop a scalable and performant application. Any help will be much appreciated. Cheers Raf

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FreeSWITCH as Debian Package?
by Michelle Konzack 24 Dec '10

24 Dec '10

Hi *, I loss my last nerv, compiling all the time FreeSWITCH from source... Is there someone providing a Debian Package from a repository? Also it would be nice if FreeSWITCH go into the Debian distribution. Thanks, Greetings and nice Day/Evening Michelle Konzack Systemadministrator Electronic Engineer Tamay Dogan Network Debian GNU/Linux Consultant -- Linux-User #280138 with the Linux Counter, http://counter.li.org/ ##################### Debian GNU/Linux Consultant ##################### <http://www.tamay-dogan.net/> Michelle Konzack <http://www.can4linux.org/> Apt. 917 <http://www.flexray4linux.org/> 50, rue de Soultz Jabber linux4michelle(a)jabber.ccc.de 67100 Strabourg/France IRC #Debian (irc.icq.com) Tel. DE: +49 177 9351947 ICQ #328449886 Tel. FR: +33 6 61925193

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nibblebilll low balance
by Madovsky 24 Dec '10

24 Dec '10

I have this config in nibblebill.conf.xml -----  <param name="lowbal_amt" value="1"/> <param name="lowbal_action" value="lowbalance XML default"/> ---- But no effect if the cash field starts to be under 1 I'm wondering if the cash field has to be exactly 1 if not it fails ? THanks

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X-FS-SUPPORT header
by Alexandr Gevlichenko 24 Dec '10

24 Dec '10

Hi list Freeswitch adds to all INVITE messages header: X-FS-Support: update_display. How can I remove it from the INVITE message? Thanks in advance.

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Can't get demo_ivr to work - just goes to Music-on-hold
by Tom C 24 Dec '10

24 Dec '10

When I dial extension 6000 to use the demo_ivr, I hear the "please dial it now" message. But no matter what number I dial, it just sends me to Music On Hold. I've tried this on my Dockstar (debian squeeze) and my P4 (debian lenny) with the same results. I want my screaming monkeys!!! :-) What am I doing wrong? Is there another module I need to load? I had this working a week ago. module_exists mod_flite true Here's the debug log after it plays the initial greeting and I start to dial: 2010-12-22 22:33:13.592915 [DEBUG] switch_rtp.c:3018 RTP RECV DTMF 5:480 2010-12-22 22:33:14.272565 [DEBUG] switch_rtp.c:3018 RTP RECV DTMF #:480 2010-12-22 22:33:14.272565 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz 2010-12-22 22:33:14.272565 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16@8000hz 1 channels 20ms I've tried with internally registered clients as well as an external SIP call. I've re-copied the demo_ivr.xml from the recently git-pulled source directory, no help.

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bind_meta_app and bind_digit_action with inband DTMF?
by Dmitry Sytchev 23 Dec '10

23 Dec '10

Hi! We are trying to migrate our Asterisk system to Freeswitch, and faced with some trouble: We use inband at user side, and want Freeswitch to regenerate it to RFC2833 before sending to media gateway. When user endpoints use rfc2833 bind_meta_app and bind_digit_action work as expected, but not with start_dtmf (as needed to parse inband DTMF). We can't use rfc2833 DTMF on user endpoints as there seems to be a problem in freeswitch or our media gateway leading to audible DTMF duplicates. When user endpoints are set to inband, I see messages on console about DTMF detection: 2010-12-22 18:50:30.802253 [DEBUG] switch_ivr_async.c:2089 DTMF DETECTED: [*] 2010-12-22 18:50:30.803253 [DEBUG] switch_rtp.c:1985 Send start packet for [*] ts=503795131 dur=160/160/2000 seq=43701 2010-12-22 18:50:30.823282 [DEBUG] switch_rtp.c:1921 Send middle packet for [*] ts=503795131 dur=320/320/2000 seq=43702 ...skip... 2010-12-22 18:50:30.923393 [DEBUG] switch_rtp.c:1921 Send middle packet for [*] ts=503795131 dur=1920/1920/2000 seq=43712 2010-12-22 18:50:30.923393 [DEBUG] switch_rtp.c:1921 Send end packet for [*] ts=503795131 dur=2080/2080/2000 seq=43713 ...skip... 2010-12-22 18:50:31.242695 [DEBUG] switch_ivr_async.c:2089 DTMF DETECTED: [1] 2010-12-22 18:50:31.242695 [DEBUG] switch_rtp.c:1985 Send start packet for [1] ts=503798651 dur=160/160/2000 seq=43732 So digits are received but no in-call action launched by bind_meta_app. I tried start_dtmf before and after bind_meta_app but with no success. Is there a way to make inband dtmf launch in-call actions while transcoding it to RFC2833 on trunk side? I can't make inband dtmf trigger my actions :( -- Best regards, Dmitry Sytchev, IT Engineer

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Dropping the Subject: header
by Kane, Michael (mkane02) 23 Dec '10

23 Dec '10

Hello all, we are developing a custom app that will require sending a Subject: header (with a specific value) with the initial invite to trigger additional calling features and behavior. What we're finding when we send the Invite, is that FS is dropping the Subject: header and continues normal call processing. Is there a configuration parameter that allows the header to pass through FS? I submitted a SIP trace in pastbin and the reference number is 14853. Mike

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Using auto-record
by Wagner 22 Dec '10

22 Dec '10

Hello, I'm quite new to FreeSwitch and so far I'm liking it I have 2 questions about FS first: If I start as a daemon using the -nc command, how do I access the console again? Second: I have a conference that I want to record, I'm trying to use auto-record but so far I'm not getting it to work my public.xml looks like that: <extension name="conferences"> <condition field="destination_number" expression="^(7\d{4})$"> <action application="answer"/> <action application="conference" data="$1@wideband"/> <profiles> <profile name="default"> <action applicatioon="conference" data="auto-record /tmp/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> </profile> </profiles>  </condition> </extension> I got no error on freeswitch.log but the file it's not created =/ I'm trying with more than one user, as I saw that it only start to record with 2 or more users Thanks for the help

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(no subject)
by Rex Alex 22 Dec '10

22 Dec '10

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