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Message Playback Stops
by Michael De Lorenzo 23 Feb '11

23 Feb '11

Hello, I have a Freeswitch installation that is intended to make many calls (thousands) and playback a single wav file. The calls are successfully processed (the recipient's phone rings), but the call almost immediately disconnects, after about 1s. Everything seems to work fine if I'm only pushing one or two calls through the Freeswitch instance, but as soon as I turn up the call rate (I'm still only doing about 50 concurrent sessions) the playback begins to fail. I've watched the calls go out from the console and nothing looks out of the ordinary, except that the calls are disconnected with NORMAL CLEARING prior to completion. Here's the Lua script I'm using... profile_id = argv[1]; account_code = argv[2]; client_id = argv[3]; caller_id_name = argv[4]; caller_id = argv[5]; dial_id = argv[6]; number_to_call = argv[7]; message_to_play = argv[8]; max_retries = argv[9]; retry_interval = argv[10]; local human_detected = false local voicemail_detected = false; local message_played = false; recordings_directory = "/usr/local/freeswitch/recordings/messages/"; function setDialVariables(set_as_session_variables) local s = "profile_id=" .. profile_id; s = s .. ",account_code=" .. account_code; s = s .. ",client_id=" .. client_id; s = s .. ",caller_id_name=" .. caller_id_name; s = s .. ",caller_id=" .. caller_id; s = s .. ",dial_id=" .. dial_id; s = s .. ",number_to_call=" .. number_to_call; s = s .. ",message_to_play=" .. message_to_play; freeswitch.consoleLog("notice", s .. "\n"); return s end function printSessionVariables() freeswitch.consoleLog("notice", "******* PRINTING SESSION VARIABLES **********\n"); -- ommitted freeswitch.consoleLog("notice", "**********************************************\n"); end function onInput(s, type, obj, arg) if(type == "event" and voicemail_detected == false) then freeswitch.consoleLog("debug","************ VOICE MAIL/ANSWERING MACHINE DETECTED *************\n"); voicemail_detected = true; return "break"; end return true; end function playbackMessage(sleepTime) message_played = false; session:sleep(sleepTime); -- play a file message_file = recordings_directory .. message_to_play; freeswitch.consoleLog("notice", "Playing file: " .. message_file .. "\n"); session:streamFile(message_file); freeswitch.consoleLog("notice", "!!!!! Finished playing the file !!!!!\n"); message_played = true; end session = freeswitch.Session("{" .. setDialVariables(false) .. ",ignore_early_media=true,origination_caller_id_name=" .. caller_id_name .. ",origination_caller_id_number=+1" .. caller_id .. "}sofia/gateway/gateway_" .. profile_id .. "/" .. number_to_call); while(session:ready()) do setDialVariables(true) session:answer(); -- session:execute("continue_on_fail","true"); session:setInputCallback("onInput","true"); session:execute("avmd","start"); playbackMessage(200); vm_status = voicemail_detected == true and "yes" or "no" freeswitch.consoleLog("info", "Was VM detected? " .. vm_status .. "\n"); if(voicemail_detected) then return "break"; end freeswitch.consoleLog("notice", "Played the message at least once and checked for VM, we should be exiting the loop.\n") end if (voicemail_detected) then freeswitch.consoleLog("info", "Playback for voicemail.\n"); session:execute("avmd","stop"); playbackMessage(5000); end freeswitch.consoleLog("info", "All finished, hanging up the session.\n"); session:hangup(); Any help would be greatly appreciated. Thank you, Michael

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VM to email error
by Spencer Thomason 23 Feb '11

23 Feb '11

Hello all, I keep seeing an error in the logs similar to the following when a VM is emailed. Everything seems to be working just fine so I'm trying to decipher the log entry. 2011-02-21 19:57:48.288269 [ERR] switch_utils.c:699 Unable to execute command: /bin/cat /tmp/mail.12983398676f47 | /usr/sbin/sendmail -f fromuser(a)fromdomain.com -t touser(a)todomain.com /bin/cat works as does /usr/sbin/sendmail. This is on Centos 5.5 w/ Postfix instead of sendmail and /var/log/ maillog clearly shows the message being delivered. Do I have something misconfigured? Thanks, Spencer

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Re: [Freeswitch-users] risky hangup
by Madovsky 23 Feb '11

23 Feb '11

apparently I found the problem, there were 2 instances of FS running... btw, is there any trick (other than killall -9 freeswitch before start the fs startup script) to avoid to run 2 instances of FS ? ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Tuesday, February 22, 2011 7:06 PM Subject: Re: [Freeswitch-users] risky hangup Example: I make an external call without credits so the call is transferred (by nibblebill) to a "nofunds" extension with plaback wav file ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Tuesday, February 22, 2011 6:42 PM Subject: Re: [Freeswitch-users] risky hangup please give some context... how are you testing this? What is calling what, etc.? -MC On Tue, Feb 22, 2011 at 3:26 PM, Madovsky <infos(a)madovsky.org> wrote: on last git I noticed that every hangup after TTS is made with a dfferent delay. sometimes it cuts the voice, sometimes it cuts suddenly before the voice starts and sometimes it cuts the voice 2/3 seconds before. after 10 tests all hangups are done with a different delay... please advice thanks ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Easy IVR?
by EdPimentl 23 Feb '11

23 Feb '11

Is this not a good example of how easy IVR design/creation should be? http://asymmetrical-view.com/2011/02/20/twilio-in-ten-minutes.html

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newbie question: how do you set the folder that freeswitch uses for sounds?
by Edward de Jong 23 Feb '11

23 Feb '11

I see that in the default config, freeswitch uses sounds from the en/us/callie folder. so when you ask for playback of hello.wav, it goes to that folder by default. how does one change the folder used by freeswitch? i want to let the user pick between different voices, which of the many config files sets up which voice to use? and what if I want to have better voice on my user's phone? can I go to 16000 sample rate? if the user has HD capability on their phone, shouldn't I be using better voice samples? and how can I choose the voice folder depending on the phone connected to the PBX at that moment? In other words, if the user has an HD phone can I connect with them at the better codec? obviously if we connect to some other lesser device the dreaded transcoding will occur, but i would think for internal use, for example when the user is navigating menus, they would appreciate a better voice. Or should i just use all waveforms at 16000 and let it drop them down... thanks in advance.

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Newbie question / what is the best text to speech module out there?
by Edward de Jong 23 Feb '11

23 Feb '11

I see Cepstra and some other text to speech modules. Which is the best of the free ones? for my experiments I can't afford a fancy one like Cepstra. edj

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Question regardin freeswitch startup info
by Hareem Haque 23 Feb '11

23 Feb '11

i start freeswitch it says max sessions 1000 and session rate 30.. what exact do these numbers mean and how can i increase them Your help is greatly appreciated. Best Regards Hareem. Haque On Sun, Feb 20, 2011 at 12:11 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Codec negotiation, again (Steven Ayre) > 2. Re: Any experience with DTMF from FreeSwitch to Sonus with > Vega ATAs? (Yehavi Bourvine) > 3. Need help with CDR setup and call routing patterns (Hareem Haque) > 4. Re: Need help with CDR setup and call routing patterns > (Steven Ayre) > 5. Re: Need help with CDR setup and call routing patterns > (Steven Ayre) > > > ---------- Forwarded message ---------- > From: Steven Ayre <steveayre(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Sun, 20 Feb 2011 12:36:45 +0000 > Subject: Re: [Freeswitch-users] Codec negotiation, again > In default behaviour, FS will negotiate the codec for the aleg, then do the > dialplan, then do the bleg. As a result it can have already picked the 729 > codec by the time it hits the dialplan. With transcoding that's fine but > with 729 not. > > The workaround is the late-negotiation profile parameter. It delays picking > the aleg codec until the bleg picks one, then uses that if possible for the > aleg. > > Steve on iPhone > > > On 20 Feb 2011, at 01:33, Serge Yuriev <me(a)nevian.org> wrote: > > > Hello, > > > > A offers g729, g711a, g711u > > FS allows all of them > > B offers g711a, g711u > > > > Call fails with INCOMPATIBLE DESTINATION. > > > > Cant understand why FS offers only ONE codec (g729) to B - wiki says it > should offer ordered LIST of.. > > GIT few days old, default config > > > > -- > > wbr, > > Serge > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: Yehavi Bourvine <yehavi.bourvine(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Sun, 20 Feb 2011 14:39:18 +0200 > Subject: Re: [Freeswitch-users] Any experience with DTMF from FreeSwitch to > Sonus with Vega ATAs? > Hello Steve, > > Thanks for clarifying it. > > __Yehavi: > > 2011/2/20 Steve Underwood <steveu(a)coppice.org> > >> Hi Yehavi, >> >> Section 3.2 of RFC2833 is poorly worded, and causes a lot of issues in >> the real world. The first paragraph says you can choose to send both >> audio and events, but doesn't clarify how. Presumably RFC2198 should be >> used. Is your system using RFC2198 encoding? I doubt it. The second >> paragraph talks about tone onset getting through as audio. Valid DTMF >> can be as short as 45ms, so 60ms is far more than anything which might >> be termed onset. Because they didn't specify anything about what is >> tolerable as tone onset passing through the channel as audio we have >> chaos today. >> >> RFC2833 is an obsolete spec, and we should be talking about RFC4733 >> today. Section 2.5.1.3.1 of RFC4733 does clearly call for combined >> payloads to be sent as RFC2198 packets. RFC4733 is even vaguer about the >> onset issue, though. >> >> Steve >> >> >> On 02/20/2011 06:33 PM, Yehavi Bourvine wrote: >> > Hello Steve, >> > I uderstand from paragraph 3.2 in RFC-2833 that the sender may send >> > some audio of the DTMF while sending the events, until it revognises >> > that this is a DTMF tone. The 60mSec is what I've been told by Vega >> > engineers. >> > Thanks! __Yehavi: >> > >> > 2011/2/20 Steve Underwood <steveu(a)coppice.org <mailto: >> steveu(a)coppice.org>> >> > >> > On 02/20/2011 03:31 PM, Yehavi Bourvine wrote: >> > > Hello, >> > > Anyone has an experience with the above configuration? All >> > sides are >> > > marked as RFC-2833 and the Sonus recognises most of the DTMF's >> > twice. >> > > The Vega sends both RFC-2833 events and about 60msec of the DTMF >> > tone >> > > (according to Vega's engineers it takes it about 60 msec to >> > detect the >> > > DTMF tones). >> > > According to the RFC this is acceptable. The service provider I am >> > > working with says that it shouldn't be like that. Before I waste >> > time >> > > in wars, has anybody had this issue and knows whether it is a >> matter >> > > of configuration at the Sonus side? >> > > Thanks! __Yehavi: >> > Are you saying the Vega sends 60ms of DTMF as audio, then sends >> > RFC2833 >> > DTMF packets and mutes the audio? In what reading of the RFC is that >> > considered appropriate action? In addition, a DTMF detector should >> > detect is more like 40ms. 60ms is rather long. >> > >> > Steve >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org >> > <mailto:FreeSWITCH-users@lists.freeswitch.org> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org <http://www.freeswitch.org/> >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > ---------- Forwarded message ---------- > From: Hareem Haque <hareem.haque(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Sun, 20 Feb 2011 11:08:08 -0500 > Subject: [Freeswitch-users] Need help with CDR setup and call routing > patterns > Thank you very much for helping me out with the initial switch setup. I > really appreciate it. However, I am stuck with the cdr and call routing > patterns setup. > > > A. I have attached my cdr_csv file and its cdr log file. As you may see in > the conf xml that i have 22 items that i need posted onto my cdr logs. > However, the csv file only shows 15 items. How can i make this work > > B. I have setup a few sip trunks. Now how do i set them up so that if i > dial 01192 that call goes to gateway A and if i dial 1XXX then that call > goes to gateway B > > Many thanks for all the help. I really appreciate it > > Best Regards > Hareem. Haque > > > > > ---------- Forwarded message ---------- > From: Steven Ayre <steveayre(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Sun, 20 Feb 2011 17:01:47 +0000 > Subject: Re: [Freeswitch-users] Need help with CDR setup and call routing > patterns > After changing the config, did you reload the config? > > reloadxml > reload mod_cdr_csv > > Steve on iPhone > > > > On 20 Feb 2011, at 16:08, Hareem Haque <hareem.haque(a)gmail.com> wrote: > > > Thank you very much for helping me out with the initial switch setup. I > really appreciate it. However, I am stuck with the cdr and call routing > patterns setup. > > > > > > A. I have attached my cdr_csv file and its cdr log file. As you may see > in the conf xml that i have 22 items that i need posted onto my cdr logs. > However, the csv file only shows 15 items. How can i make this work > > > > B. I have setup a few sip trunks. Now how do i set them up so that if i > dial 01192 that call goes to gateway A and if i dial 1XXX then that call > goes to gateway B > > > > Many thanks for all the help. I really appreciate it > > > > Best Regards > > Hareem. Haque > > > > > > <cdr_csv.conf.xml> > > <hareem.csv> > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: Steven Ayre <steveayre(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Sun, 20 Feb 2011 17:10:34 +0000 > Subject: Re: [Freeswitch-users] Need help with CDR setup and call routing > patterns > >> B. I have setup a few sip trunks. Now how do i set them up so that if i >> dial 01192 that call goes to gateway A and if i dial 1XXX then that call >> goes to gateway B >> > > Handle each in a different extension in the dialplan: > > <extension name="route1"> > <condition field="destination_number" expression="^(01192)$"> > <action application="bridge" data="sofia/gateway/gatewaya/$1"/> > </condition> > </extension> > <extension name="route2"> > <condition field="destination_number" expression="^(1\d\d\d)$"> > <action application="bridge" data="sofia/gateway/gatewayb/$1"/> > </condition> > </extension> > > Adjust the expressions to suit the numbers you want to handle better. The > contents of the () brackets are placed in variable $1, which is being used > in the bridge to dial that number via the gateway. You can add prefixes etc > there if you need them. ^ and $ match the start and end of the string, not > actual characters. They're called regular expressions if you want to read up > on them, and are more flexible than just matching absolute strings because > you match patterns instead. > > An example of changing a prefix, should your gateway need that (for example > converting a local number to an international prefix): > <condition field="destination_number" expression="^0(1192)$"> > <action application="bridge" data="sofia/gateway/gatewaya/44$1"/> > </condition> > 01192 would be dialed as 441192 through gateway a. > > -Steve > > > > > On 20 February 2011 16:08, Hareem Haque <hareem.haque(a)gmail.com> wrote: > >> Thank you very much for helping me out with the initial switch setup. I >> really appreciate it. However, I am stuck with the cdr and call routing >> patterns setup. >> >> >> A. I have attached my cdr_csv file and its cdr log file. As you may see in >> the conf xml that i have 22 items that i need posted onto my cdr logs. >> However, the csv file only shows 15 items. How can i make this work >> >> B. I have setup a few sip trunks. Now how do i set them up so that if i >> dial 01192 that call goes to gateway A and if i dial 1XXX then that call >> goes to gateway B >> >> Many thanks for all the help. I really appreciate it >> >> Best Regards >> Hareem. Haque >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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FS gateway which registers into FS
by Shigeru Kanemoto 23 Feb '11

23 Feb '11

Hello, I am new to FreeSWITCH and having a headache configuring it. My experiences to Asterisk do not work completely. I appreciate your suggestion. I am trying to setup a central SIP server which accepts registers from subsidiary SIP servers which is behind NAT and has no fixed IP address. Each subsidiary servers have extension number ranges. I have to route such extension ranges to each SIP servers. FS central server <--- (NAT) --- Server A (ext. 1000 - 1099) <--- (NAT) --- Server B (ext. 1100 - 1199) <--- (NAT) --- Phone ext. 1201 <--- (NAT) --- Phone ext. 1202 According to the wiki page http://wiki.freeswitch.org/wiki/Clarification:gateways , I created an XML file under "conf/directory/default" for a gateway setting inside a <user> block. In my configuration, the gateway setting is parsed in the profile "internal". My current problem is the dialplan "bridge" application does not work to bridge to the destination like "sofia/gateway/server-a/1001". The call terminates after printing a log message like "[NOTICE] sofia.c:5082 Hangup sofia/internal/1001 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]". What is my mistake? Is it wrong to use the "gateway" feature for such purpose? sgk

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risky hangup
by Madovsky 23 Feb '11

23 Feb '11

on last git I noticed that every hangup after TTS is made with a dfferent delay. sometimes it cuts the voice, sometimes it cuts suddenly before the voice starts and sometimes it cuts the voice 2/3 seconds before. after 10 tests all hangups are done with a different delay... please advice thanks

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OpenSER interface to FreeSWITCH
by Jason b.a 23 Feb '11

23 Feb '11

1- i need an IVR application separated from the media server 2- the application server should control the media server using MSCML , VXML or any control protocol i know it is easier to use the embedded IVR in freeswitch, but this is not my case. Beside by using openSER between the application and the media server ==> openSER can interface with Freeswitch through socket interface,to control it ! i dont know if the sip sophia interface can handle the request from the application , and pass it to the core for processing, and generate a responce and pass it through the same interface, so here no need for OpenSER if we took this example: 1-Caller call IVR application 2-Application as media server to play welcome message and to collect digit 3-caller enter digits 4-media server pass the digits to the application 5-application bridge caller and callee together

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