February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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auth-all-packets
by Madovsky 22 Feb '11

22 Feb '11

Is auth-all-packets to true decrease performance ? Thanks

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OpenSER interface to FreeSWITCH
by Jason b.a 22 Feb '11

22 Feb '11

Hi, is it possible to implements is an interface to OpenSER over a TCP connection, so the design will be : Caller----IVR application----OpenSER-----Freeswitch | | Callee without the need to use the Sip sophia module, only openSER control the freeswitch for media processing using the socket interface. Thx

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javascript hanguphook
by Christian Hiller 22 Feb '11

22 Feb '11

Hello, i have a javscript that is called from xml-dialplan ... function on_hangup(hup_session, how) { console_log("info","got hungup"); exit(); } session.execute('bridge','sofia/internal....'); session.setHangupHook(on_hangup); ... Now i experience, that this function is only called if the session is not answered yet. Once its answered and then got hung up, the function on_hangup is not called anymore. Any ideas ? Kind regards Christian Hiller

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Re: [Freeswitch-users] OpenSER interface to FreeSWITCH
by Bernhard Suttner 22 Feb '11

22 Feb '11

currently not but might be a cool idea ... ----- Original Message ----- From: Jason b.a [mailto:k-b-24@live.com] To: freeswitch-users(a)lists.freeswitch.org Sent: Tue, 22 Feb 2011 20:35:55 +0100 Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH > > Hi, > > is it possible to implements is an interface to OpenSER over a TCP > connection, so the design will be : > > Caller----IVR application----OpenSER-----Freeswitch > | > | > Callee > > without the need to use the Sip sophia module, only openSER control the > freeswitch for media processing using the socket interface. > > Thx

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MOH not working in G729 passthru with native sound files
by Marcin Wojtowicz 22 Feb '11

22 Feb '11

I'm trying to enable MOH when both legs of the call are using G729 (FS is in passthru). I converted an edited sample wave file to G729 and put in the appropriate folder, and FS loads it correctly because this is the message that keeps popping up in the console: 2011-02-09 21:10:25.609375 [INFO] mod_native_file.c:94 Opening File [C:\freeswitch\sounds/music/8000\sample.G729] 8000hz I establish a call, and everything is fine, but when I press hold on my handset I see an error message that says that G729 is only useable in passthru (here is the debug message): 2011-02-09 21:14:01.828125 [DEBUG] switch_channel.c:1376 (sofia/internal/ sip:M9jdt73ig0oOJSbt6Uyy@192.168.1.50:5060) Callstate Change ACTIVE -> HELD 2011-02-09 21:14:01.828125 [DEBUG] switch_core_session.c:954 Send signal sofia/external/MYHOME#(a)74.63.41.218 [BREAK] 2011-02-09 21:14:01.859375 [DEBUG] switch_core_session.c:709 Send signal sofia/external/MYHOME#(a)74.63.41.218 [BREAK] 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr.c:563 sofia/external/MYHOME#@ 74.63.41.218 Command Execute playback(local_stream://moh/8000) EXECUTE sofia/external/MYHOME#@74.63.41.218playback(local_stream://moh/8000) 2011-02-09 21:14:02.000000 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr_play_say.c:1244 Codec Activated L16@8000hz 1 channels 20ms 2011-02-09 21:14:02.015625 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-02-09 21:14:02.015625 [ERR] switch_core_io.c:1042 Codec G.729 encoder error! 2011-02-09 21:14:02.015625 [DEBUG] switch_ivr_play_say.c:1581 done playing file I don't understand why that would be, since my music file is in G729 so I'm not asking freeswitch to convert, only stream. My custom ringback (before a call is established) works just fine using a similar method, so could anyone explain me why what I want to do is not permitted?

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