Re: [asterisk-users] Open source speech recognition engine?
On 21/04/2012 at 16:41 +0300, Carl-Fredrik Enell wrote: Dear all, I am looking for an open source speech recognition engine for a hobby project. There used to be a Sphinx interface for the generic speech API (http://scribblej.com/svn/) but it does not compile on Asterisk versions later than 1.6.x Could anybody direct me on how to update this code, or should I simply change to the AGI script approach? To use asterisk with the open source CMUSphinx speech recognitoin engine please try AST-UniMRCP and unimrcp server with the pocketsphinx plugin: http://code.google.com/p/unimrcp/wiki/PocketSphinxPlugin It's a recommended way to use Pocketsphinx with asterisk. It's somewhat hard to compile but it will work fine if you will do everything properly. signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on Asterisk Conference
Be aware that there are several different conferencing solutions for asterisk. I've used app_meetme in asterisk 1.8 (the LTS release) pretty happily. It is reasonably full featured and well supported. It has 2 drawbacks : 1) it needs a kernel module (Dahdi) to do the mixing and timing 2) it only does narrowband codecs If you like living nearer to the edge, there is app_confbridge in asterisk 1.10 (aka 10) which removes those 2 issues, but is new and so lacks some of the support (e.g. web modules etc) In terms of numbers, I've always found that user management issues kick in before the software/hardware limitations. Your milage may vary. T. On 20 Apr 2012, at 18:20, Mitchell Johnson wrote: We're looking into using Asterisk to do our conferencing. Currently we do all our conferencing using Cisco, we have a router with PVDM modules so we can offload the hardware resources. I'm looking for some best practices on how to set it up. 1. DO I need a separate server for the conference server? 2. Do I need to offload the actual conference to a router with PVDM modules. 3. Does anyone have experience with transitioning from Cisco conferencing to Asterisk? 4. How many participants can participate in a conference? Thanks, Mitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding degradation G711-iLBC
I'm quite fond of GSM610 as a low(ish) bandwidth codec - although it isn't as good as (say) speex or Silk, it is widely supported, and European users have had years of cellphone use to get used to the specific sound of a GSM call. So you can often go from a GSM610 supporting handset all the way through to a GSM supporting ITSP without needing to transcode at all. If at all possible avoid creating a path which involves 2 different lossy codecs - e.g. 729 _and_ GSM the results are significantly worse than either. If you can control all of the call path and have devices that support it, Silk is _lovely_ . It takes a bit of tuning for your expected network (which is unfortunately manual in Asterisk 10) but it is worth it. Tim. On 15 Apr 2012, at 12:15, Gustavo Garcia Bernardo wrote: Is it a good idea to use asterisk transcoding from G711 to iLBC or should I find out any other solution not involving transcoding (f.e. using G.729 that is supported in both sides). I'm worried about voice quality and trying to avoid paying for G.729 licensing. Anybody with experience or quantitative measurements of the voice quality degradation in that scenario? Regards, G Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on Asterisk Conference
Hi Mitch, Firstly, I am not a conferencing guru; I just hope this helps. I'm looking for some best practices on how to set it up. 1. DO I need a separate server for the conference server? This depends on a few factors: (a) You won't be able to run MySQL alongside Asterisk with conferencing and get good results. If you plan to use a single Asterisk server to do conferencing and other voice functions (for example voicemail) then I wouldn't expect any major issues. It depends on the usage of each voice function of the system. (b) How many conference participants will you have and will they all be bi-directional audio or will it be more oriented towards having a single person addressing multiple channels which are only listening? (c) Are you planning on connecting SIP endpoints directly to Asterisk and perhaps passing some of the callers through other trunks (SIP or ISDN)? 2. Do I need to offload the actual conference to a router with PVDM modules. Got no idea, sorry. Haven't worked with Cisco voice equipment personally. 3. Does anyone have experience with transitioning from Cisco conferencing to Asterisk? A couple of my clients have wanted integration with other systems (Nortel / Avaya) and you will be best using the cheapest option of ISDN cross-over cabling (if you already have spare T1/E1 ports) or SIP trunks. The Cisco (or Asterisk server) can be programmed in such a way that the conference participants (or voicemail users etc) don't know which system they are interacting with. 4. How many participants can participate in a conference? Depending on 1b and your processor specification, you can host quite large conferences with only marked users speaking and many end points listening. You would likely have to get a testing budget and figure it out for your own system if no one else can provide you more detailed information. I hope this helps. Kind Regards Stuart -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6
Remove the Asterisk source dir, unpack the tarball again and run configure. 1.4 is weird about configure being built before DAHDI is installed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Saturday, April 21, 2012 5:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6 Dear; The output of the ./configure that is related to dahdi is: checking for DAHDI_RESET_COUNTERS in dahdi/user.h... yes checking dahdi/tonezone.h usability... yes checking dahdi/tonezone.h presence... yes checking for dahdi/tonezone.h... yes And the dependecies of the chan_dahdi as I saw in the make menuselect is (but really I do not know what the M and E means, and how I can be sure if they are existed or not)? Depends on: res_smdi(M), dahdi(E), tonezone(E), res_features(M), pri(E) Any help? Regards Bilal --- Well, I did make menuselect and I really found the XXX and did not get the ability to select the channel. So what could be the reason? When you are in menuselect, looking at the 'channels' page, scroll the cursor down to chan_dahdi (marked with 'XXX'), and look at the bottom of the window/screen. In that area there will be information about the chan_dahdi dependencies that were or were not found by the the configure script. If you can copy and paste that information here, we can try to help you figure out what is going on. It's quite strange that codec_dahdi successfully built but chan_dahdi did not; the problem is likely not related to DAHDI, but due to some other dependency that chan_dahdi has. As I said before, what we really should be looking at is the configure script output that indicates what it was able to find and what it was not able to find, but the menuselect information is a reasonable next step. As Kevin said, you need to check the out put when you run ./configure. You could pipe it through less or copy-and-paste it into a text editor to search it for anything about dahdi. It should tell you what's wrong. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6
In article C262B52114110B4586FAF49F074F05801090DF2FCA@mailserver2007.nyigc.globe, Eric Wieling ewiel...@nyigc.com wrote: Remove the Asterisk source dir, unpack the tarball again and run configure. 1.4 is weird about configure being built before DAHDI is installed. No need to remove and unpack again. Just run: make distclean Cheers Tony From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Saturday, April 21, 2012 5:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6 Dear; The output of the ./configure that is related to dahdi is: checking for DAHDI_RESET_COUNTERS in dahdi/user.h... yes checking dahdi/tonezone.h usability... yes checking dahdi/tonezone.h presence... yes checking for dahdi/tonezone.h... yes And the dependecies of the chan_dahdi as I saw in the make menuselect is (but really I do not know what the M and E means, and how I can be sure if they are existed or not)? Depends on: res_smdi(M), dahdi(E), tonezone(E), res_features(M), pri(E) Any help? Regards Bilal -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on Asterisk Conference
On Sun, 22 Apr 2012, Stuart Elvish - IP Exchange Systems wrote: 1. DO I need a separate server for the conference server? This depends on a few factors: (a) You won't be able to run MySQL alongside Asterisk with conferencing and get good results. If you plan to use a single Asterisk server to do conferencing and other voice functions (for example voicemail) then I wouldn't expect any major issues. It depends on the usage of each voice function of the system. I don't think there was enough information in the OP's post to support the statement that running MySQL and Asterisk on the same box will not yield good results. I prefer to run them on separate boxes. Database servers and 'telco' servers have different resource requirements and seem to need different administration styles but they are not fundamentally incompatible. Aside from the OP asking 'how many participants' (which leads me to assume he wants 'a lot') we have no clue to his requirements. If the OP wants 'lots of participants' and is doing 'a lot' of database activity then separate servers are justified. Otherwise, a dozen or so participants and the database on the same box would not concern me from a 'can do' perspective. I'd still vote for separate boxes if the budget can support it. Hey OP, better details yield better suggestions :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme identify user number
Hi Group, is in MeetMe any option to identify the own number (from the view of a caller)? I would like to write an option to set on the telephone an request for voice, if the room is muted. That request should display on our Conference Control Website and an Admin should unmute this person. Thanx for help. Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
On 18/04/2012 6:39 AM, Kevin P. Fleming wrote: On 04/17/2012 06:17 AM, Larry Moore wrote: The send log you have posted does not show any outgoing T.38 packets from your system. I set up a test build of 1.8.11.0 using the patch recently released, I have difficulties sending T.38 with this patch, in fact I cannot send successfully however I can receive. I did however observe some outgoing T.38 packets. The analogue fax modem I was dialling into is under my control hence the log files showed there was no signalling coming from my ITSP, The T.38 session on my Asterisk server show the CRP's which were sent from the analogue fax device during the negotiation. The patch I have used for a while seems to give me outgoing functionality as well as incoming. I can't reproduce your scenario whereby your T.38 session is communicating with a different gateway to the SIP server you use hence can only speculate that Asterisk has difficulty wanting to send T.38 SDP traffic when it is a different device than the SIP server it negotiates with. We know for a fact that Asterisk has no trouble with the signaling and media going to different addresses/ports. Honestly, I just don't understand why all of this effort is being put into trying to use an old (and clearly broken) patch for adding T.38 gateway support to Asterisk 1.8. You guys know that it works in Asterisk 10, but you say you can't use Asterisk 10 for some reason that I don't understand. I have downloaded asterisk 10.3.0 and compiled on a Centos 6 system I setup to compare behaviour on OpenBSD with a Linux version of asterisk based upon the OpenBSD port. Unfortunately the T.38 Gateway functionality in my build of 10.3.0 doesn't appear to work. Looking at the upgrade documentation from 1.8 there doesn't appear to be any considerations applicable to my setup. As an excercise in futility I downloaded the Asterisk 1.8.11.0 source and compiled using the version of T.38 patch I have maintained and tested by sending a fax via an IAX channel out through my SIP provider, the fax was sent successfully. I then removed and recreated the asterisk 1.8.11.0 directory and applied the back-port patch and observed the same problem when attempting to send through the T.38 gateway as was observed in Asterisk 10. Console output of Asterisk 1.8.11.0 with Asterisk 10 backport patch: asterisk-dev*CLI -- Accepting AUTHENTICATED call from 192.168.54.12: requested format = slin, requested prefs = (), actual format = slin, host prefs = (slin|alaw|ulaw), priority = mine -- Executing [@FAX-T30:1] Set(IAX2/iaxmodem1-4445, FAXOPT(t38gateway)=yes) in new stack -- Executing [@FAX-T30:2] Dial(IAX2/iaxmodem1-4445, SIP/@itsp-fax,55) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/@itsp-fax -- SIP/itsp-fax-0004 is making progress passing it to IAX2/iaxmodem1-4445 -- SIP/itsp-fax-0004 answered IAX2/iaxmodem1-4445 [Apr 23 21:14:11] NOTICE[14165]: channel.c:4152 __ast_read: Dropping incompatible voice frame on SIP/itsp-fax-0004 of format slin since our native format has changed to 0x8 (alaw) == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 [Apr 23 21:14:12] ERROR[14165]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL [Apr 23 21:14:23] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: RTP Read too short [Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: RTP Read too short [Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: RTP Read too short [Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: RTP Read too short [Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: RTP Read too short [Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: RTP Read too short [Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: RTP Read too short [Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: RTP Read too short Console output of Asterisk 10.3.0: Connected to Asterisk 10.3.0 currently running on asterisk-dev (pid = 14798) Verbosity is at least 3 Core debug is at least 3 -- Accepting AUTHENTICATED call from 192.168.54.12: requested format = slin, requested prefs = (), actual format = slin, host prefs = (slin|alaw|ulaw), priority = mine -- Executing [@FAX-T30:1] Set(IAX2/iaxmodem1-863, FAXOPT(t38gateway)=yes) in new stack -- Executing [@FAX-T30:2] Dial(IAX2/iaxmodem1-863, SIP/@itsp-fax,55) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/@itsp-fax -- SIP/itsp-fax- is making progress passing it to IAX2/iaxmodem1-863 -- SIP/itsp-fax- answered IAX2/iaxmodem1-863 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 [Apr 23 21:25:15] ERROR[14983]: astobj2.c:110 INTERNAL_OBJ: user_data is
