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[INFO] mod_dialplan_xml.c:331 Processing Unavailable <Restricted>->2901 in context default
by Mario 24 Sep '10

24 Sep '10

I ran some traces and can't seem to figure out if this is an error I can fix. Otherwise it works. No luck on web searches so far. Thanks. [INFO] mod_dialplan_xml.c:331 Processing Unavailable <Restricted>->2901 in context default My 2901 dial plan is: <extension name="uid1vm"> <condition field="destination_number" expression="^2901$"> <action application="answer"/>  <action application="voicemail" data="default ${domain_name} 200"/> </condition> </extension>

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Re: [Freeswitch-users] limits on text to speech output length
by Daniel Neubert 24 Sep '10

24 Sep '10

It's not only good for having a better scalable infrastructure - it also gives you the ability to freely change your tts engine or add functionality like speech recogniction or recording. Daniel Am 24.09.2010 05:59, schrieb A. Lester Buck III: > Dear Daniel, > > Thanks so much for the pointer. I'll certainly check that out. I don't > really have much of a scaling issue at the moment, but I sure hope I do > someday! Always good to keep the architecture as small parts loosely > joined. > > Best regards, > > Lester > > >> Hi, >> >> instead of calling swift directly you can consider calling it via an >> MRCP server like UniMRCP. >> >> It's all free software and works like a charm in my setup. I'm using >> mod_unimrcp on our FreeSWITCH servers and am running UniMRCP as a >> server on a single node that is our dedicated speech server. >> >> That scenario also offloads the task of speech generation to another >> machine to save resources on the FreeSWITCH nodes. If you use other >> cepstral languages than english you need to manually patch the sources. >> >> Daniel >> >> Am 23.09.2010 22:57, schrieb A. Lester Buck III: >>> Hi, >>> >>> I'm running Freeswitch built from trunk a week or so ago, 1.0.head. >>> >>> I am using a Cepstral voice for text to speech. I generate a long >>> string of text, including SSML breaks, that is rendered to speech by the >>> swift utility packaged with Cepstral voices. The test string I built >>> renders as 1m:46s of speech, and a .wav file about 1.6MB, from the swift >>> command line. >>> >>> I changed single quotes to double quotes and escaped them, as mentioned >>> on the wiki page for mod_cepstral, and I now have a working text to >>> speech path using ESL built for Ruby 1.9.2, using the sample Ruby code >>> on the wiki. >>> >>> Everything basically works fine, even on my test machine which is a >>> $5/mo Xen VPS with 10ms kernel timer resolution. (lowendbox.com) >>> >>> My issue is that the speech cuts off midway through the call, when I >>> guess some limit on generated output is being reached. Before I dig >>> into the guts of mod_cepstral and Freeswitch, could someone point me to >>> what might be limiting the length of output? Swift seems to generate a >>> wav file of indefinite length, so at least the issue is somewhere in the >>> open source parts of the server. >>> >>> Having yet looked at the guts of mod_cepstral, I presume that a >>> high-performance output module does not have the swift utility write a >>> .wav file to disk and then play it back, but instead lets swift shove >>> the bytes directly into a socket. What is a good Freeswitch example of >>> this architecture? >>> >>> >>> Thanks, >>> >>> Lester >>> >>> _______________________________________________ >>> 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 >> ATT00001.txt >> >> >> _______________________________________________ >> 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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CODEC2 Support in FreeSWITCH
by Brian West 23 Sep '10

23 Sep '10

I have been working on adding CODEC2 support to FreeSWITCH which is a low bitrate LGPL codec by David Rowe. I have made changes to create the lib required to make this work and waiting on them to be pushed back up stream so I have checked the lib into the libs folder for now. You can find the lib and module in the latest git pull. ./bootstrap.sh ./configure make mod_codec2-install edit modules.conf.xml to load the mod add the codec to your allowed codec list on sofia profiles. Please check it out and let me know what you think. Thanks, Brian

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Registration of SIP client not lasting
by bsd 23 Sep '10

23 Sep '10

Hello, I am using FreeSwitch embedded in a pfSense device. Things are running ok… My SIP client is Acrobits SoftPhone running on IPhone 4g. The Sip client is registering ok, but when I look at the registration on FreeSwitch, it disappears after a couple of minutes… Is there a way for me to tune that somewhere at the level of FreeSwitch ? Thanks. ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯ Gregober ---> PGP ID --> 0x1BA3C2FD bsd @at@ todoo.biz ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯

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Voicemail not to announce extension number
by mazilo 23 Sep '10

23 Sep '10

I have used the examples/conf files to configure my FS (v1.0.6) hosted on a Seagate DockStar running on an OpenWRT OS. So far, so good. I am able to place/receive outbound/receive calls, respectively sans any problems. The Voicemail (VM) nicely kicks in when no one answer the phone and annouces the person at extension XXXX is not available ... It certainly will be nice if anyone here can help me to modify the VM not to announce the extension number. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Voicemail-not-to-announce-ext… Sent from the freeswitch-users mailing list archive at Nabble.com.

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mod_xml_curl - sample php example file location
by Aloysius Lloyd 23 Sep '10

23 Sep '10

Hi All, Help me find the Example directory.php and dialplan.php for the mod_xml_curl. Wiki says "There is a PHP example that does everything from a webserver, with per switch configuration capability in FreeSWITCH™ SVN ( scripts/contrib/trixter/xml-curl<http://fisheye.freeswitch.org/browse/Unsupported/trixter/xml-curl> )." But the link is broken. Any help is appreciated. Thanks Lloyd

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Re: [Freeswitch-users] sip stack problem in fsw
by JRichey 23 Sep '10

23 Sep '10

The gateway sending calls to the FreeSwitch box ends up dropping the calls. There are also some pops and clicks that aren't there when playing a continuous audio file. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]On Behalf Of Brian West Sent: Monday, September 20, 2010 12:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] sip stack problem in fsw You shouldn't see RTP between files when sleeping... its 100% valid to NOT send any media when their is NOTHING to be sent. Does your phone freak out or something? /b On Sep 20, 2010, at 2:08 PM, JRichey wrote: > I'm in a similar situation where I need to disable VAD because the other end > will disconnect the call if the RTP stops. I've tried the setting Anthony > suggested as well as several others I've found on the wiki, but I've not > been able to get it to work. The variables I've been trying are listed > below. > > suppress-cng > suppress_cng > rtp_disable_vad_out > rtp_disable_vad_in > bridge_generate_comfort_noise > send_silence_when_idle > > > For my test calls I'm just playing back sound files with sleep in-between > them and I see no RTP when calling sleep. I'm using tcpdump and wireshark > to analyze the calls. > > > -Justin Richey > jrichey(a)ITLtd.net _______________________________________________ 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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Sending Alpha Numeric characters using uuid_send_dtmf
by rex.alex 23 Sep '10

23 Sep '10

Hello, I am trying to send alpha numeric characters when responding to an IVR like, "api uuid_send_dtmf 6a677f06-fa1a-4257-b566-32fe2715b3dc WMW081". But it is not recognizing whereas the numeric characters are getting recognized. Please assist, how do I send? Thanks, Rex -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Sending-Alpha-Numeric-charact… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch Register Failure
by Raja Rokkam 23 Sep '10

23 Sep '10

Hi All, I have setup FreeSWITCH Version 1.0.6 (hacked-20100922T074940Z) on a Ubuntu 10.0.4 Virtual Machine on my Windows XP desktop. I am using another softphone X-Lite to connect to it. The VM uses a bridged network so essentially has its own IP assigned by a DHCP server. For testing I am using 1000.xml and changed the password fields to 1234. However I am not able to register. The error I get is: 2010-09-23 11:55:48.240027 [WARNING] sofia_reg.c:1030 SIP auth failure (REGISTER) on sofia profile 'internal' for [1000(a)192.168.6.210] from ip 192.168.6.144 Can anyone please let me know what other configurations I have to make ? Thanks, Raja.

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Codec Error! v=0
by Frank Ochere 23 Sep '10

23 Sep '10

Hello, I've been seeing the error below for calls sent to a particular SIP gateway, anyone knows what it means and should I be concerned? (I use g729 for my inbound and outbound codec prefs in FS) 2010-09-22 10:34:23.398832 [ERR] sofia.c:4919 Codec Error! v=0 o=00345254725065043 1285213687 1285213687 IN IP4 1.1.1.1 s=AddPac Gateway SDP c=IN IP4 1.1.1.1 t=1285213687 0 m=audio 24686 RTP/AVP 18 a=rtpmap:18 G729/8000/1 a=recvonly a=ptime:20 Kind Regards Frank Ochere

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