August 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

start_dtmf on BLeg Channel from the dialplan
by Phillip Jones 21 Aug '10

21 Aug '10

Hi there, Is there a way to execute start_dtmf on the B Leg, from the dialplan, before you bridge call? I am using group_confirm, so that a called user is bridged when they answer and press 1. With one of my carriers, DTMF is not working - pressing 1 does nothing. If I use group_confirm=exec and send the B leg to a script on answer, and execute "start_dtmf" on the b leg session in the script, all works as expected and pressing 1 bridges the call. I would perfer not to use a script if there is a way of achieving this via the dialplan. Any help appreciated. Thanks Pj

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why dingaling is using G722 codec?
by jesse zhao 20 Aug '10

20 Aug '10

I am dialing from SIP to Jingle. in mod_jingle.conf.xml I only configured PCMU and GSM codec. How come FS sends G722 codec and gtalk side immedially rejects the call. 2010-08-20 14:23:56.777993 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- <iq xmlns="jabber:client" from="XYZ(a)gmail.com/talk88DE1430" to="DEF(a)gmail.com/AGoogleVoice6C59F787" type="set" id="306"> <ses:session xmlns:ses="http://www.google.com/session" type="initiate" id="3486862015" initiator="XYZ(a)gmail.com/talk88DE1430"> <pho:description xmlns:pho="http://www.google.com/session/phone" xml:lang="en"> <pho:payload-type xmlns:pho="http://www.google.com/session/phone" id="9" name="G722" clockrate="8000" bitrate="64000"></pho:payload-type> </pho:description> <transport xmlns="http://www.google.com/transport/p2p"></transport> </ses:session> </iq> 2010-08-20 14:23:57.000832 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- <iq to="XYZ(a)gmail.com/talk88DE1430" type="set" id="7D0A279E57C8C64B" from="DEF(a)gmail.com/AGoogleVoice6C59F787"> <ses:session type="terminate" initiator="XYZ(a)gmail.com/talk88DE1430" id="3486862015" xmlns:ses="http://www.google.com/session"> <ses:reason> <ses:general-error></ses:general-error> <ses:initiate-failed></ses:initiate-failed> </ses:reason> <pho:internal-server-error xmlns:pho="http://www.google.com/session/phone"></pho:internal-server-error> </ses:session> </iq>

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radius
by Jean-Yves F. Barbier 20 Aug '10

20 Aug '10

Hi list, what can bring a RADIUS server in the equation? (I never used one, so my knowledge is basic: centralized authentication using PAP or CHAP protocols) JY -- The human race never solves any of its problems. It merely outlives them. -- David Gerrold

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IVR menu failure/timeout action
by alan lo 20 Aug '10

20 Aug '10

Dear all, Referring to the threat below, where should the dp extension stored? In the same XML of ivr or should store it in other locations? Thx! Alan I have confirmed this behavior in the latest git as of last night. I don't know if this is by design or not. I would open JIRA as a feature request to have this behavior modified. The other alternative would be to have each IVR submenu called via transfer instead of menu-sub: <snip>  <entry action="menu-exec-app" digits="1" param="transfer sub_menu_1 XML"/> </snip> Then have a dp extension to handle that: <extension name="sub menu 1"> <condition field="destination_number" expression="^(sub_menu_1)$"> <action application="ivr" data="m800_ivr_submenu-m_ch"/> </condition> </extension> -MC On Mon, Jul 19, 2010 at 2:37 AM, Raymond Chan <raymondchan at commverge.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >wrote: > Hi all, > > > > I am experiencing an IVR problem. Except the top menu, all other sub menu > will not hang up after reached max invalid input or timeout limit. It will > go back up upper menu. I want it to load the exit-sound and then hang the > call when it reached max invalid input or timeout limit. Do you tell how to > configure? > > > > <menu name="m800_ivr_submenu-m_ch" > > > greet-long="C:/FreeSWITCH/recordings/800-ivr-2nd-m.wav" > > > greet-short="C:/FreeSWITCH/recordings/greet-short-m.wav" > > > invalid-sound="ivr/ivr-that_was_an_invalid_entry-m.wav" > > exit-sound="voicemail/vm-goodbye-m.wav" > > timeout="5000" > > max-failures="2"> > > <entry action="menu-exec-app" digits="1" param="playback > C:/FreeSWITCH/recordings/800-ivr-announcement-m_ch.wav"/> > > <entry action="menu-exec-app" digits="2" param="lua > endcall-m.lua"/> > > </menu> > > > > Thanks > > > > Raymond

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dtmf events outbound in 'freeswitcher' ruby library
by Nigel Kent 20 Aug '10

20 Aug '10

Does anyone know how to get dtmf events in outbound with the freeswitcher ruby library. I see that in inbound 'FSL::Inbound.add_event_hook(:DTMF) {|event| dtmf_handler(event) ' can be used but how can it be done in outbound

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bootstrap.sh hanging on
by Neil Patel 20 Aug '10

20 Aug '10

Hi All, bootstrap.sh is hanging for me while running on vanilla Ubuntu 9.04 at this point: automake: compiling `check_dlopen_sofia.c' with per-target flags requires `AM_PROG_CC_C_O' in `configure.ac' tests/Makefile.am:37: while processing program `check_dlopen_sofia' Just sits here until I hit enter and the script exits. I also tried running make from libs/sofia-sip/tests, but that doesn't work. I am running the latest from git. Thanks, Neil

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nosmp timing issue
by stephen at stephenjc 19 Aug '10

19 Aug '10

Just as an FYI, on a Single Processor without hyper threading or multiple cores adding nosmp to the kernel options fixed my inaccurate time keeping. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print

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Paging calls
by Sean Holt 19 Aug '10

19 Aug '10

Hello list, I wondering if there¹s anyone on the list that has had success configuring paging/intercom using an all polycom phone environment and freeswitch. I¹ve been able to configure the polycom config files to auto answer on the second line but I¹m lost when configuring my dialplan. Currently I¹m pointing an extension to a ring group, but the only phone that auto-answers is the first phone in the ring group list. Phones: 321 polycom Thanks for any help Sean

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FS and Gtalk integration
by jesse zhao 19 Aug '10

19 Aug '10

hi, help: I searched online pages and FS wiki , basically the doc about Gtalk and FS is pretty lousy. I still couldn't get my gtalk and FS work together. two major issues: 1) when I call gtalk(gtalk=XYZ) from SIP phone, call is established, but no audio. 2) how could I call SIP from gtalk ? when I request chat to sip, gchat in gmail requests an invitation. suppose the voice request is : conf+1000(a)172.18.115.73 here are my config files: jingle_profiles/client.xml <profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="XYZ(a)gmail.com/talk"/> <param name="password" value="XXXXX"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/>  <param name="sasl" value="md5"/>  <param name="server" value="talk.google.com"/>  <param name="tls" value="true"/>  <param name="use-rtp-timer" value="true"/>  <param name="exten" value="jingle2sip"/>    <param name="vad" value="both"/>   <param name="local-network-acl" value="localnet.auto"/> </profile> </include> dialplan/default.xml <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^conf\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/default/$1%172.18.115.73"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^gtalk=([a-zA-z0-9.-]+)$"> <action application="set" data="no_media=false"/> <action application="set" data="effective_caller_id_number=myuser at gmail.com"/> <action application="bridge" data="dingaling/gmail.com/$1@gmail.com"/> </condition> I tried suggestion from http://lists.freeswitch.org/pipermail/freeswitch-users/2007-February/028456…, above two issues still not work. I heard id: intralanman is good at this, if you saw the post, please share your advice. -jesse

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Which db/*.db files are not suited for tmpfs?
by Patrick Lists 19 Aug '10

19 Aug '10

Hi, Earlier I read the interesting postings about the use of tmpfs. Looking at the files in db/ I wonder if anyone knows which ones needs to be persistent across FreeSWITCH restarts or which are unsuitable for use with tmpfs e.g. in case of a power failure. From Steven Ayre's comment I understand that voicemail_default.db should not be put on tmpfs. Are there any other db files that should *not* be put on tmpfs? Overview of files I have in db/ on my test box: call_limit.db core.db directory.db fifo.db sofia_reg_external.db sofia_reg_internal.db sofia_reg_internal-ipv6.db voicemail_default.db zrtp.dat Regards, Patrick

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