- FreeSWITCH-users - FreeSWITCH Mailing Lists

Getting Started with Speech Recognition
by Bryan Smart 25 Jul '11

25 Jul '11

I've been attempting to experiment with this part of Freeswitch, but it seems that this is an area that hasn't yet received a lot of attention with regard to docs and samples. Perhaps a some of you that are experienced in this area wouldn't mind clearing up some of my confusion? First, I have FS working fine (outside of ASR). Pocketsphinx is built and loaded. As for me, I have some experience with VXML, and generally know how a grammar works, but the last phone ASR projects that I was involved with were several years back on Tellme Studio. First, as to grammars, I'm not sure, but it seems as though FS has used different formats at different times. The mod_pocketsphinx page says that JSGF is the current format. Can the JSGF grammar files be loaded without being compiled to another format? I ask that, as some of the examples, such as the LUA Directory example, talk about compiling grammars, and the format shown there doesn't appear exactly the same as what I see from the JSGF examples. Is the info about compiling grammars obsolete? Next, I'm not clear about the use of the detect_speech command. The wiki page lists some example forms of the command, but doesn't explain the purpose of the different forms, nor their arguments. Here are some examples: detect_speech <mod_name> <gram_name> <gram_path> [<addr>] >From the examples, I see that "pocket sphinx" can be used for the module. I think that gram_path is the path to the grammar file, without the ".gram" suffix, and uses the grammar folder as a base. However, what is addr? An address? To, or for, what? detect_speech grammaron <gram_name> I think that detect_speech grammaron <gram_name> Might be used when a large .gram file has been loaded, and the code needs to enable a grammar for a specific context. However, I don't see a command for just loading a .gram file without also specifying a grammar name. If I use this grammaron command, will it search all files in the grammar folder for grammars that match the name, or is this just for enabling a grammar that I've already loaded through other means? Maybe grammaron is used to activate a grammar that was disabled with detect_speech grammaroff <gram_name> But, if so, then what is the purpose of detect_speech nogrammar <gram_name> Does that also disable a grammar? Maybe it deletes the grammar from memory, also? For: detect_speech param <name> <value> What are the possible parameter names and acceptable values? detect_speech start_input_timers What are the purposes of the start_input_timers? Does this cause ASR to time out if no input is recognized? Does it cause ASR to pause for a time before beginning to recognize speech? I've tried to write up a simple grammar with only a few terms, and have a LUA IVR script attempt to detect speech using this grammar. My input callback is never called with any sort of speech event when I speak, though, and I don't receive any FS console errors. I must be making some incorrect assumptions about how the commands are used, or how the grammar is formatted. Perhaps my tests aren't working because there is a problem with Pocketsphinx. I can't try the Pizza demo to verify that Pocketsphinx is working correctly, as I'm unable to download its grammar while bkw.org is offline, and no other sources for the grammar come up on a Google search. Finally, I'd like to know if there are any compelling reasons to use one scripting language over another for speech-driven IVRs. I have experience with Lua, Javascript, and PHP, so could use any of them. The Freeswitch book uses Lua a lot, so that's what I've been trying so far. Perhaps I'd be better served with Javascript? Thanks for any pointers! Bryan

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Asterisk dial status
by Sam 25 Jul '11

25 Jul '11

Hi, I am trying to port over an Asterisk AGI script that does a call bridge to Freeswitch using perl, I wanted to know what is the equivalent of the asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION, CHANUNAVAIL ? I wanted to know if what methods can be used to gets these statues?

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Can't get variable set in dial plan
by Joshua Foshee 25 Jul '11

25 Jul '11

I am trying to set a variable in the dial plan and use it on a condition. I have pushed the variable setting as close to the condition for testing and this is what I get. Why does it not see the variable set? <extension name="public_did"> <condition field="destination_number" expression="^(.*)$"> <action inline="true" application="set" data="hours=1"/> </condition> <condition field="hours" expression="1"> <action application="answer"/> <action application="ivr" data="GAC-BH"/> <action application="hangup"/> <anti-action application="answer"/> <anti-action application="ivr" data="GAC-BH"/> <anti-action application="hangup"/> </condition> </extension> Here is the console output Dialplan: FreeTDM/1:7/405790xxx8 Regex (PASS) [public_did] destination_number(405790xxx8) =~ /^(.*)$/ break=on-false Dialplan: FreeTDM/1:7/405790xxx8 Action set(hours=1) INLINE EXECUTE FreeTDM/1:7/405790xxx8 set(hours=1) 2011-07-25 12:05:25.191357 [DEBUG] mod_dptools.c:1063 FreeTDM/1:7/405790xxx8 SET [hours]=[1] Dialplan: FreeTDM/1:7/405790xxx8 Regex (FAIL) [public_did] hours() =~ /1/ break=on-false Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action answer() Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action ivr(GAC-BH) Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action hangup()

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Callcenter MOD question
by Joshua Foshee 25 Jul '11

25 Jul '11

I am setting up a queue and I have ring all set with everything else sent to default pretty much. What I am trying to accomplish is that agents have phone that show 4 lines for their extension. If they are on the phone with an outbound call I want to skip trying to send them a call. How is this accomplished where the callcenter mod knows to not send a call to them. All agents are static assigned in the callcenter config file. Copy of config file below. <configuration name="callcenter.conf" description="CallCenter"> <settings> </settings> <queues> <queue name="sales@default"> <param name="strategy" value="ring-all"/> <param name="moh-sound" value="$${hold_music}"/> <param name="time-base-score" value="queue"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="14400"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="120"/> </queue> <queue name="collection@default"> <param name="strategy" value="ring-all"/> <param name="moh-sound" value="$${hold_music}"/> <param name="time-base-score" value="system"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="60"/> <param name="abandoned-resume-allowed" value="false"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="120"/> </queue> </queues> <agents> <agent name="513@default" type="callback" contact="[call_timeout=10]user/513(a)192.168.1.3" status="Available" max-no-answer="3" wrap-up time="10" reject-delay-time="10" busy-delay-time="60" /> </agents> <tiers> <tier agent="513@default" queue="sales@default" level="1" position="1"/> <tier agent="513@default" queue="collection@default" level="1" position="1"/> </tiers> </configuration>

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Help
by Sanja Angelova 25 Jul '11

25 Jul '11

Dear all! I am new to FreeSwitch and I am trying to map a number for example 654321 to sip adress like this 1005(a)10.1.1.1 with configuring a local DNS server. When I write enum 654321 in fs_cli it gives me: Offered Routes: Order Pref Service Route ============================================================================== 10 100 E2U+sip sofia/${use_profile}/1005(a)10.1.1.1 and seems to be fine. But when I make a call from my soft phone, the FreeSwitch ends up my call and it gives me error 503 service unavailable. I have no idea what the problem is, can you please help me? Thank you in advance

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Re: [Freeswitch-users] One way calling issue over an internal trunk
by Mitch Johnson 25 Jul '11

25 Jul '11

Concerning the NAT. I will always be on the internal network, all external phones will come in on a VPN and then connect onto FreeSWITCH using the VPN. I did the ./freeswitch -nonat. I also changed the sip-ip and rtp-ip to the internal ip of the Freeswitch server on both external and internal xml files: Internal: <param name="rtp-ip" value="172.16.200.60"/> <param name="sip-ip" value="172.16.200.60"/> external: <param name="rtp-ip" value="172.16.200.60"/> <param name="sip-ip" value="172.16.200.60"/> <param name="ext-rtp-ip" value="local-network-acl"/> <param name="ext-sip-ip" value="local-network-acl"/> I'm still plodding around FreeSWITCH so if I did this wrong please let me know. Thanks, Mitch On Jul 25, 2011, at 6:43 AM, freeswitch-users-request(a)lists.freeswitch.org wrote: > From: Steven Ayre <steveayre(a)gmail.com> > Subject: Re: [Freeswitch-users] One way calling issue over an internal trunk > Date: July 25, 2011 3:05:39 AM EDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > I'll always be on the internal network so should never need the nat. > > Try starting FS with the -nonat option. It'll start faster and skip all the stuff where it tries to find your external IP and open routes through the NAT firewall. > > On your sip profiles set sip-ip and rtp-ip to your internal IP. > > Once you've done that you should have 2-way audio to anywhere on your LAN. > > -Steve > > > > > On 25 July 2011 05:08, Mitch Johnson <mitch.johnson7(a)gmail.com> wrote: > Sorry for the delayed response, got tied up in other things. A quick recap. I can dial from my call manager to Freeswitch but cannot dial from freeswitch to call manager. > > I wasn't sure what you meant by setting the ext-sip-ip and ext-rtp-ip. So I replaced the auto-nat with local-network-acl. I'll always be on the internal network so should never need the nat. > > I saw in the acl.conf.xml that it automatically populates all my local subnets. The networks are correct under the domains of the acl.conf. > > I'm not sure where else to look. Any help would be greatly appreciated. > > Thanks so much, > > Mitch > > On Jul 19, 2011, at 11:29 PM, freeswitch-users-request(a)lists.freeswitch.org wrote: > >> From: Brian West <brian(a)freeswitch.org> >> Subject: Re: [Freeswitch-users] One way calling issue over an internal trunk >> Date: July 19, 2011 8:55:02 PM EDT >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> >> >> you need to set the ext-sip-ip and ext-rtp-ip and deal with setting local-network-acl to the proper network/mask because you're answering with RFC1918 media IP's to the cisco… its retarded and will just start sending to those IP's and they'll usually never make it back to you. >> >> /b >> >> On Jul 19, 2011, at 7:16 PM, Mitch Johnson wrote: >> >>> I have a connection between my FreeSWITCH and a Cisco CallManager: >>> >>> This is from the default.xml >>> >>> <extension name="Dial to CM8"> >>> <condition field="destination_number" expression="^(800\d)$"> >>> <action application="bridge" data="sofia/internal/$1(a)172.16.200.100"/> >>> </condition> >>> </extension> >> >

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Altering arbitrary channels
by Isaac Jurado 25 Jul '11

25 Jul '11

Hi all, >From the diaplan, is it possible to set a variable in any other channel? That is, by specifying the channel UUID. Thanks. -- Isaac Jurado Internet Busines Solutions eConcept

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memory leak
by Tihomir Culjaga 25 Jul '11

25 Jul '11

hello, i upgraded to the latest GIT compiled configured FS without any issues... everything looks great... but when i put this into some traffic i noticed FS uses more and more memory... until it finished into swap. I didn't have issues like that previously but unfortunately i cannot recall what was the version with no issues. The server is 64bit CentOS 5.4 and the FS version is the latest (well 4 days old tops). Im using just the standard modules (the list provided below) and FS keeps leaking memory <load module="mod_console"/> <load module="mod_logfile"/> <load module="mod_enum"/> <load module="mod_cdr_csv"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_loopback"/> <load module="mod_commands"/> <load module="mod_conference"/> <load module="mod_db"/> <load module="mod_dptools"/> <load module="mod_expr"/> <load module="mod_fifo"/> <load module="mod_hash"/> <load module="mod_voicemail"/> <load module="mod_esf"/> <load module="mod_fsv"/> <load module="mod_cluechoo"/> <load module="mod_valet_parking"/> <load module="mod_dialplan_xml"/> <load module="mod_dialplan_asterisk"/> <load module="mod_spandsp"/> <load module="mod_g723_1"/> <load module="mod_g729"/> <load module="mod_amr"/> <load module="mod_ilbc"/> <load module="mod_speex"/> <load module="mod_h26x"/> <load module="mod_siren"/> <load module="mod_sndfile"/> <load module="mod_native_file"/> <load module="mod_local_stream"/> <load module="mod_tone_stream"/> <load module="mod_spidermonkey"/> <load module="mod_lua"/> <load module="mod_say_en"/> This is the dialplan im triggering... <extension name="doRedirect"> <condition field="destination_number" expression="^300030.*$"> <action application="info"/> <action application="redirect" data="<sip:0999901234567889@10.1.1.11:5060;user=phone>;q=0.99"/> <action application="hangup"/>  </condition> </extension> I have i have tried a pure "hello world" dialplan as: <extension name="testDP"> <condition field="destination_number" expression="^300030.*$"> <action application="info"/> <action application="hangup"/> </condition> </extension> But the situation is exactly the same. This is the behavior i'm having .... memory just keep growing and growing until it hits the swap where the server is doomed. tail -f smaps_log.log (this is from cat /proc/$PID/smaps | grep heap) TIM,SIZE,RSS,SHARED_CLEAN,SHARED_DIRTY,PRIVATE_CLEAN,PRIVATE_DIRTY,SWAP 2011-07-13 13:50:34,19268,18916,0,0,0,18916,0 2011-07-13 13:51:34,19268,18972,0,0,0,18972,0 2011-07-13 13:52:34,20324,20088,0,0,0,20088,0 2011-07-13 13:53:34,21264,20920,0,0,0,20920,0 2011-07-13 13:54:34,21048,20856,0,0,0,20856,0 2011-07-13 13:55:35,22396,22076,0,0,0,22076,0 2011-07-13 13:56:35,22396,22164,0,0,0,22256,0 2011-07-13 13:57:35,23220,22936,0,0,0,22936,0 2011-07-13 13:58:35,23480,23288,0,0,0,23288,0 $ tail -f mem.log (this is from ps) TIMESTAMP PID RSS %MEM 2011-07-13 13:48:55 2405 65144 12.8 2011-07-13 13:49:55 2405 66932 13.2 2011-07-13 13:50:55 2405 69576 13.7 2011-07-13 13:51:55 2405 71552 14.1 2011-07-13 13:52:55 2405 73576 14.5 2011-07-13 13:53:55 2405 75804 14.9 2011-07-13 13:54:55 2405 78388 15.4 2011-07-13 13:55:55 2405 80544 15.9 2011-07-13 13:56:55 2405 81784 16.1 2011-07-13 13:57:55 2405 83720 16.5 2011-07-13 13:58:55 2405 86524 17.0 FS runs under root account and it is started as: /usr/local/freeswitch/bin/freeswitch -nc -nonat -db /dev/shm -conf /usr/local/freeswitch/conf/ -log /usr/local/freeswitch/log/ -run /usr/local/freeswitch/run/ Also ulimit is like this; ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 240 ulimit -l unlimited so, does anyone have a clue on why the memory goes up ? ... what I'm doing wrong ? Regards, Tihomir.

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Re: [Freeswitch-users] One way calling issue over an internal trunk
by Mitch Johnson 25 Jul '11

25 Jul '11

Sorry for the delayed response, got tied up in other things. A quick recap. I can dial from my call manager to Freeswitch but cannot dial from freeswitch to call manager. I wasn't sure what you meant by setting the ext-sip-ip and ext-rtp-ip. So I replaced the auto-nat with local-network-acl. I'll always be on the internal network so should never need the nat. I saw in the acl.conf.xml that it automatically populates all my local subnets. The networks are correct under the domains of the acl.conf. I'm not sure where else to look. Any help would be greatly appreciated. Thanks so much, Mitch On Jul 19, 2011, at 11:29 PM, freeswitch-users-request(a)lists.freeswitch.org wrote: > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] One way calling issue over an internal trunk > Date: July 19, 2011 8:55:02 PM EDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > you need to set the ext-sip-ip and ext-rtp-ip and deal with setting local-network-acl to the proper network/mask because you're answering with RFC1918 media IP's to the cisco… its retarded and will just start sending to those IP's and they'll usually never make it back to you. > > /b > > On Jul 19, 2011, at 7:16 PM, Mitch Johnson wrote: > >> I have a connection between my FreeSWITCH and a Cisco CallManager: >> >> This is from the default.xml >> >> <extension name="Dial to CM8"> >> <condition field="destination_number" expression="^(800\d)$"> >> <action application="bridge" data="sofia/internal/$1(a)172.16.200.100"/> >> </condition> >> </extension> >

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Sofia stack sip rfc conformance.
by Narendra Sirugudi 24 Jul '11

24 Jul '11

Hi All, I wanted to know what all SIP standards/rfc(s) does Sofia stack conform to ? I could not find this informantion anywhere so far. thanks, --naren

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