- FreeSWITCH-dev - FreeSWITCH Mailing Lists

ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem
by Michel Habib 07 Mar '11

07 Mar '11

Hello All, I have MS OCS Speech Server 2007 [working correctly, as i can make SIP calls and use its ASR and TTS Services successfully] I am also using MRCP Connector from AumTech - which allows me to use ASR and TTS Services through an MRCP Client . Now, i am using Freeswitch mod unimrcp to use ASR and TTS. for TTS, I can successfully make the call, the Audio RTP of the TTS voice is transferred succesfully from Speech Server [through MRCP Connector] back to the Freeswitch Server. However, Freeswitch is not sending back the Audio RTP to the SIP client. for ASR, I can successfully define the grammar and start recognition, but the audio RTP sent to speech server [through MRCP Connector] is silent [empty]. I am suspecting something is wrong with the RTP Configuration - can you help me? Let me now if you need any specific logs/scripts/configuration? Thank you, Michel.

1 0

FreeSWITCH Conference Call Today
by Michael Collins 02 Mar '11

02 Mar '11

Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_02 We are going to have an update from Mark Crane on the FusionPBX project. Bring your questions! -MC

1 0

[ANNOUNCEMENT] EVERYONE PLEASE READ
by Anthony Minessale 01 Mar '11

01 Mar '11

Hi, I just want to give a reminder that we have a growing community and we want to put our resources to their best use. Here are some important things to consider. Many of you are already doing these things so if you already are, thank you, please help encourage others to follow your example. 1) We have a bug/issue tracker site using JIRA: http://jira.freeswitch.org This site is for reporting and discussing issues. Please use this tool to keep track of issues you have. Even if it's not a bug we can easily close it tagged "not a bug". Also please help out when you can by reviewing this site to see if you can provide answers to other people. 2) If you see someone complain about the WIKI please remind them that anyone can edit the WIKI and if it's wrong we can clean it up. It takes longer to complain about the wiki than it does to edit it to be correct or add the missing info. That is the idea behind wikis. 3) Please take the time to provide answers to any questions you may know when perusing the list. The more people who help the less work it is to keep everyone informed. 4) Take the time to test the latest development build. Its better to find bugs before we release than after. People tend to anchor themselves to the "official" releases and never try the BETA versions. The sad truth is the tagged release will begin to build a large list of known bugs starting 30 seconds after its tagged. You don't have to run HEAD in production every morning but you should stay up to speed with what's going on with the software and help to produce a stable release that truly is stable. 5) We will be branching 1.0 so it will slow down and taper off from upstream patches from HEAD. Anyone who wants to help maintain that via JIRA and GIT should let us know. Once we branch, most of our attention will still be focused on HEAD and only critical patches or bug fixes will be shared between 1.0 and 1.2 and we will rely on those interested in slower paced development to help keep that code base in order. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

1 0

FreeSWITCH Conference Call Starting Shortly
by Michael Collins 23 Feb '11

23 Feb '11

Hey folks, feel free to hop on! http://wiki.freeswitch.org/wiki/FS_weekly_2011_02_23 Talk to you shortly, MC

1 0

ASR from Freeswitch to Loquendo - No Audio passed to MRCP Server
by Michel Habib 18 Feb '11

18 Feb '11

Dears, I am seeking expert advice on setting up Freeswitch to Loquendo MRCP Connection. I have freeswitch on one computer, configured for MRCP V1 - Loquendo Speech Suite 7. I have Loquendo set-up on the other computer - with MRCP V1 Profile. Using a simple lua script driving my extension, I can successfully:- - establish the connection to Loquendo Server. - define grammar - compile grammar successfully. - Begin Recognition Then it waits forever until i hangup the call. Looking at the Wireshark Traces, the RTP packets are transferred succesfully from softphone to freeswitch [i could hear my voice on the playback] But, when playing back the RTP packets between freeswitch and Loquendo, there is only deep silence [same thing with audio dump created by Loquendo]. Can you direct me where to troubleshoot this on the Freeswitch side? I will be more than happy to provide any logs you need to resolve this. Thank you, Michel.

1 0

Skype 2.0.72 for Linux
by Michael Collins 18 Feb '11

18 Feb '11

Does anyone know where to find this? It's the only one that the wiki says works with mod_skypopen but not even Google seems to be able to find one. (I find "2.0.72" but it links to the latest beta on the skype.com site which is not what we want.) Any suggestions are appreciated. -MC

1 0

FreeSWITCH Conference Call Today
by Michael Collins 16 Feb '11

16 Feb '11

Hey folks, We are having a "get caught up with the latest in FreeSWITCH" session today. Here's the agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_02_16 I will be reviewing a lot of the stuff that went into FreeSWITCH in the month of January. If you are familiar with some of the new things please feel free to add your input to the call today and/or update the wiki page. For example, if you are familiar with the mod_snmp or mod_fsk we would definitely welcome your observations. Talk to you soon, Michael

1 0

Bridge to PSTN Number Not Using Call Timeout Value
by Jerry Richards 15 Feb '11

15 Feb '11

Hello All, When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this? Here is the line that sets the call_timeout variable: switch_channel_set_variable(channel, "call_timeout", "6"); Here is the PSTN bridge log: EXECUTE sofia/internal/2003@192.168.72.79:5060 bridge({presence_id=2001(a)192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248142341@g1) Thanks, Jerry

3 9

INFORMATION: FreeSWITCH Training, West Coast
by Michael Collins 15 Feb '11

15 Feb '11

This is a reminder for those who are interested in the FreeSWITCH Bootcamp put on by the 2600hz project. There are two upcoming training sessions, both on the west coast of the USA: Mar 9-11, San Francisco May 11-13, Seattle The complete list is available at: http://www.voipkb.com/ There are a few international training sessions that will be happening later in the year, so check the VoIPKB site for updates. Thanks to all those who are promoting the use of FreeSWITCH around the world! -Michael

1 0

DB update of 11th February -- is it just me?
by Lawrence Conroy 14 Feb '11

14 Feb '11

Hi there, Just wondered -- has anyone had problems since the registration db commit of Friday 11th? I'm running fS on a Mac TiBook (PPC, running Mac OS X 10.5.8), so this is waaaay off the standard target. Having updated to head on a cleaned system, the new fS dumps a core on startup. Did a git revert -n a2c0da53f368f0b11340c3a72814c93b182753b7, followed by make sure. Problem went away. ---- As a minor issue, the new code seems to generate an error report well before fS segs. This error says that the registration db doesn't have a field "real". Looking at the patch for the a2c0da53f368f0b11340c3a72814c93b182753b7 commit, this error seems to tie in with the change to switch_core_sqldb.c, and the static char create_registrations_sql[]= bit, where there's an added line (1638): create index regindex1 on registrations (user, real, hostname)\n"; Was that intended? Putting this as: create index regindex1 on registrations (user, realm, hostname)\n"; removes the error report. Sorry for the delay in reporting, but it takes ages to do anything on this old machine and it is locked away without a reliable Internet connection. all the best, Lawrence

2 1