[asterisk-users] Making Asterisk a Voice Router
Hi, I'm interested in what software (Free or course) that people use when they want to add a dial by voice service to their asterisk system. Meaning I pick up the phone.. dial some extension. it prompts me for name.. I say John Smith.. and it dials his extension and connects the call.. TIA, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Making Asterisk a Voice Router
Coool... thanks man.. do you have any installation procedures or notes? Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Monday, October 22, 2007 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Making Asterisk a Voice Router On Mon, 2007-10-22 at 09:39 -0400, end1r wrote: I’m interested in what software (Free or course) that people use when they want to add a “dial by voice” service to their asterisk system. Meaning I pick up the phone.. dial some extension… it prompts me for name.. I say “John Smith”.. and it dials his extension and connects the call.. I've done this using Asterisk and the LumenVox speech engine... in fact, I spoke about it at AstriCon Europe in 2006. My slides are available at http://www.astricon.net/files/Jared_Smith_EUR06.pdf. (They may be slightly out of date, but it should at least get you started.) -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Making Asterisk a Voice Router
Is this free? I see the tuner is free.. but the speech rec isn’t? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lenz Sent: Monday, October 22, 2007 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Making Asterisk a Voice Router Nice job! I took the liberty to post it on AstPligg as well: http://tinyurl.com/268bac Thanks l. In data Mon, 22 Oct 2007 16:22:13 +0200, Jared Smith [EMAIL PROTECTED] ha scritto: On Mon, 2007-10-22 at 09:39 -0400, end1r wrote: I’m interested in what software (Free or course) that people use when they want to add a “dial by voice” service to their asterisk system. Meaning I pick up the phone.. dial some extension… it prompts me for name.. I say “John Smith”.. and it dials his extension and connects the call.. I've done this using Asterisk and the LumenVox speech engine... in fact, I spoke about it at AstriCon Europe in 2006. My slides are available at http://www.astricon.net/files/Jared_Smith_EUR06.pdf. (They may be slightly out of date, but it should at least get you started.) -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
Do you have any console messages? SCCP uses a station start tone message with a value of Inside Dial Tone and a direction of Tone Output User. and the line instance and Station Tone Output Direction should be set to something other than 0. SCCP runs over TCP so you should get this message, but it would be interesting to see if you get this message the phone still doesnt play dial tone. My experiences with chan_sccp have been disappointing at best. If you can get a trace of both SCCP legs send it to me and i can take a look at it. -- Original message -- From: Jason Parker [EMAIL PROTECTED] - [EMAIL PROTECTED] wrote: [snip] I have a feeling I'm forgetting something fairly easy and stupid, but I can't seem to see what it is. Anyone have any suggestions? Dial(SCCP/[EMAIL PROTECTED]) -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outdial to phone for new VM notification
Hi all, Does anyone have an application/script or extensions.conf file which will do the following? When a new VoiceMail is left for a user, the asterisk system will place a call to a cellphone/pstn number(via some provider). When the user answers his cell/home phone, comedian mail will ask for his password and he can check his Asterisk VM? Anyone have any examples of it working? If not, how hard would this be to implement. TIA! Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom autoprovision behind a NAT
You really need 2 packet traces from the "outside and "inside" side of the router showing the TCP process between the phone and the server. Is the connection even getting opened, is there an FTP error code? Post traces of TCP and we can look what happening -- Original message -- From: "Curt Shaffer" [EMAIL PROTECTED] If you want that is fine. But as I mentioned when I put the phone on the same subnet as the ftp server with no NAT it works like a charm. Is there something in the config that deals with NAT traversal with regards to how it is provisioned? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ BeaupreSent: Monday, November 06, 2006 8:26 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Polycom autoprovision behind a NAT I if you like, I can take a config file(s) and put up over here as a test. Our ftp is working. It might be informative. -Original Message-From: "Curt Shaffer" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Mon, 6 Nov 2006 20:17:07 -0600Subject: RE: [asterisk-users] Polycom autoprovision behind a NAT To be honest I donât know for sure. I am using VSFTPD. I have never needed to set this with setups I have used it before and there is nothing in the config that says passive. So Iâm guessing that its not. After you asked this I have googled passive FTP and it seems to be on the money as to what is going on so I will try passive and see if that helps. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ BeaupreSent: Monday, November 06, 2006 7:45 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom autoprovision behind a NAT I can confirm that the linksys routers cause ftp problems. Is your FTP server set to use pasive mode? -rb -Original Message-From: "Curt Shaffer" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Mon, 6 Nov 2006 19:19:48 -0600Subject: [asterisk-users] Polycom autoprovision behind a NAT I am having an issue with doing FTP auto provisioning of Polycom 501âs when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router. Any ideas? Curt ---BeginMessage--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco CCM - Asterisk
Looks like the CallManager is unable to find the endpoint in its database. Make sure asterisk trunk on the Call manager is in the same calling Search Space as the phones are in, or make sure there is access between the calling search spaces -Eric -- Original message -- From: Alyed Tzompa [EMAIL PROTECTED] Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integrat ion but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages everytime I send a call. Had play a lot with the way SIP messages are sent to the Cisco, but always been unseccessful. I'm begining to think this is more of a Cisco config problem than Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so dun't know if I need to enable SIP messageing/reception in the Cisco. Regards, Alyed ---BeginMessage--- Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages everytime I send a call. Had play a lot with the way SIP messages are sent to the Cisco, but always been unseccessful. I'm begining to think this is more of a Cisco config problem than Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so dun't know if I need to "enable" SIP messageing/reception in the Cisco. Regards, Alyed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 911 versus 9.911
I know every second counts in a real 911 situation, but what about adding a pause in the call flow. Maybe a 1 second pause before actually passing the digits to the provider. This gives the user 1 second to realize the mistake and hang up, longer than 1 seconds is a real emergency. Just a thought. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Aarons (US) Sent: Wednesday, August 30, 2006 10:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 911 versus 9.911 Is there a FCC or other North America requirement that I provide 911 versus 9.911. I want to require users to dial 9.911 in our office, and remove 911. Are there any statutory requirements or laws about this? User accidentially dial 9 then 1 then another 1 and hangup. Weve educated them to stay on the line and ever hang up, but they hang up anyway, resulting in fines for excess hangups to 911. Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hitting # to Transfer out of a Queue
I thought # to transfer didnt work if you have a t,t orr in your dial
string since asterisk remains in the media path?
but its just a guess.
-- Original message --
From: Douglas Garstang [EMAIL PROTECTED]
-Original Message-
From: Douglas Garstang
Sent: Tuesday, July 18, 2006 12:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Hitting # to Transfer out of a Queue
-Original Message-
From: Patrick [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 18, 2006 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Hitting # to Transfer out of a Queue
On Tue, 2006-07-18 at 10:29 -0600, Douglas Garstang wrote:
[snip]
exten = oe_ccare,1,NoOp(*** Incoming call
from ${CALLERID} to queue oe_ccare)
exten = oe_ccare,n,Set(TIMEOUT(response)=5)
exten = oe_ccare,n,
GotoIfTime(8:00-17:00|mon-fri|*|*?one_queue_acd,oe_ccare-open,1)
exten = oe_ccare,n,Goto(oe_ccare-shut,1)
exten = oe_ccare-open,1, Answer
exten = oe_ccare-open,n,
Set(__TRANSFER_CONTEXT=one_start)
exten = oe_ccare-open,n, NoOp(${__TRANSFER_CONTEXT})
exten = oe_ccare-open,n(queue1), Queue(oe_custcare30)
Is this a literal copy of your dialplan? If so I was not
aware you could
put spaces between priorities and actions. Have you tried
removing them:
exten = foo,1,NoOP(spaces are evil, mostly)
Patrick, yes, this is a literal portion. I have no reason to
believe that spsaces between the priority, and the command
cause problems, so I haven't tried that yet. Just trying to
make the horrible assembler-like Asterisk dialplan language
more readable.
I just tried this with a very simple dialplan example that
didn't involve queues.
exten = 4001,1,Set(__TRANSFER_CONTEXT=footest)
exten = 4001,2,Dial(SIP/2944093,20,tr)
[footest]
exten = 1234,1,Answer
exten = 1234,2,Wait,1
exten = 1234,3,Playback(blue-eyed-polar-bear)
I dial 4001, and answer the call at 2944093. I then hit #1,
and asterisk plays 'pbx-transfer' followed by dial tone. I
put in 1234, and extension 1234 in context footest is called.
Works fine.
I'm starting to wonder if this is a bug of some sort, and
TRANSFER_CONTEXT cannot be used with queues. Has anyone
actually tried it?
exten = oe_ccare,1,NoOp(*** Incoming call
from ${CALLERID} to queue oe_ccare)
exten = oe_ccare,n,Set(TIMEOUT(response)=5)
exten = oe_ccare,n,
GotoIfTime(8:00-17:00|mon-fri|*|*?one_queue_acd,oe_ccare-open,1)
exten = oe_ccare,n,Goto(oe_ccare-shut,1)
exten = oe_ccare-open,1, Answer
exten = oe_ccare-open,n, Set(__TRANSFER_CONTEXT=one_start)
exten = oe_ccare-open,n, NoOp(${__TRANSFER_CONTEXT})
exten = oe_ccare-open,n(queue1), Queue(oe_custcare30)
... more stuff here
and we also have the context where agent callbacks are. I
even tried putting the TRANSFER_CONTEXT where the agent is called.
[one_callback]
;
; Agent callbacks. Used by the AgentCallBackLogin app to dial agents.
;
exten = 80014054,1,NoOp(Dialling Customer Care Spare)
exten = 80014054,n,Set(__TRANSFER_CONTEXT=one_start)
exten = 80014054,n,Dial(SIP/80014054)
The one_start context should match any number dialled, as it
has _X. as a pattern match. However, as I said, as soon as I
enter a digit, asterisk plays pbx-invalid.
Further to this, I've added some debugging statements...
exten = oe_ccare,1,NoOp(*** Incoming call from ${CALLERID}
to
queue oe_ccare)
exten = oe_ccare,n,Set(TIMEOUT(response)=5)
exten = oe_ccare,n,
GotoIfTime(8:00-17:00|mon-fri|*|*?one_queue_acd,oe_ccare-open,1)
exten = oe_ccare,n,Goto(oe_ccare-shut,1)
exten = oe_ccare-open,1, Answer
exten = oe_ccare-open,n, Set(__TRANSFER_CONTEXT=one_start)
exten = oe_ccare-open,n, NoOp(BEFORE Q ${TRANSFER_CONTEXT})
exten = oe_ccare-open,n(queue1), Queue(oe_custcare30)
exten = oe_ccare-open,n, NoOp(AFTER Q ${TRANSFER_CONTEXT})
The variable TRANSFER_CONTEXT is not modified by the Queue command. It
remains
unchanged. I also put debugging where we dial the agent...
exten = 80014054,1,NoOp(BEFORE DIAL AGENT
${TRANSFER_CONTEXT})
exten = 80014054,n,Dial(SIP/80014054)
The variable is still unchanged before dialling the agent. HOWEVER, the
asterisk
console still logs this when I try and do a transfer. It looks like the DIAL
command is IGNORING the TRANSFER_CONTEXT variable when called from a queue.
Jul 18 11:51:48 VERBOSE[30143]
RE: [asterisk-users] Hitting # to Transfer out of a Queue
Yes sorry.. i was thinking of something else. I had a problem where I put the T
in the dial string and the media wouldnt go end-end, but thats because Asterisk
has to remain in the RTP stream to hear the #.
Much different that what your reporting... sorry to mis-lead ya...
-- Original message --
From: Douglas Garstang [EMAIL PROTECTED]
And I thought that t and T allowed the caller and callee to transfer a call?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 18, 2006 3:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Hitting # to Transfer out of a Queue
I thought # to transfer didnt work if you have a t,t orr
in your dial string since asterisk remains in the media path?
but its just a guess.
-- Original message --
From: Douglas Garstang [EMAIL PROTECTED]
-Original Message-
From: Douglas Garstang
Sent: Tuesday, July 18, 2006 12:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Hitting # to Transfer out of a Queue
-Original Message-
From: Patrick [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 18, 2006 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Hitting # to Transfer out
of a Queue
On Tue, 2006-07-18 at 10:29 -0600, Douglas Garstang wrote:
[snip]
exten = oe_ccare,1,NoOp(*** Incoming call
from ${CALLERID} to queue oe_ccare)
exten = oe_ccare,n,Set(TIMEOUT(response)=5)
exten = oe_ccare,n,
GotoIfTime(8:00-17:00|mon-fri|*|*?one_queue_acd,oe_ccare-open,1)
exten = oe_ccare,n,Goto(oe_ccare-shut,1)
exten = oe_ccare-open,1, Answer
exten = oe_ccare-open,n,
Set(__TRANSFER_CONTEXT=one_start)
exten = oe_ccare-open,n,
NoOp(${__TRANSFER_CONTEXT})
exten = oe_ccare-open,n(queue1), Queue(oe_custcare30)
Is this a literal copy of your dialplan? If so I was not
aware you could
put spaces between priorities and actions. Have you tried
removing them:
exten = foo,1,NoOP(spaces are evil, mostly)
Patrick, yes, this is a literal portion. I have no reason to
believe that spsaces between the priority, and the command
cause problems, so I haven't tried that yet. Just trying to
make the horrible assembler-like Asterisk dialplan language
more readable.
I just tried this with a very simple dialplan example that
didn't involve queues.
exten = 4001,1,Set(__TRANSFER_CONTEXT=footest)
exten = 4001,2,Dial(SIP/2944093,20,tr)
[footest]
exten = 1234,1,Answer
exten = 1234,2,Wait,1
exten = 1234,3,Playback(blue-eyed-polar-bear)
I dial 4001, and answer the call at 2944093. I then hit #1,
and asterisk plays 'pbx-transfer' followed by dial tone. I
put in 1234, and extension 1234 in context footest is called.
Works fine.
I'm starting to wonder if this is a bug of some sort, and
TRANSFER_CONTEXT cannot be used with queues. Has anyone
actually tried it?
exten = oe_ccare,1,NoOp(*** Incoming call
from ${CALLERID} to queue oe_ccare)
exten = oe_ccare,n,Set(TIMEOUT(response)=5)
exten = oe_ccare,n,
GotoIfTime(8:00-17:00|mon-fri|*|*?one_queue_acd,oe_ccare-open,1)
exten = oe_ccare,n,Goto(oe_ccare-shut,1)
exten = oe_ccare-open,1, Answer
exten = oe_ccare-open,n,
Set(__TRANSFER_CONTEXT=one_start)
exten = oe_ccare-open,n, NoOp(${__TRANSFER_CONTEXT})
exten = oe_ccare-open,n(queue1), Queue(oe_custcare30)
... more stuff here
and we also have the context where agent callbacks are. I
even tried putting the TRANSFER_CONTEXT where the agent is called.
[one_callback]
;
; Agent callbacks. Used by the AgentCallBackLogin app to
dial agents.
;
exten = 80014054,1,NoOp(Dialling
Customer Care Spare)
exten = 80014054,n,
Set(__TRANSFER_CONTEXT=one_start)
exten = 80014054,n,Dial(SIP/80014054)
The one_start context should match any number dialled, as it
has _X. as a pattern match. However, as I said, as soon as I
enter a digit, asterisk plays pbx-invalid.
Further to this, I've added some debugging statements...
exten = oe_ccare,1,NoOp(*** Incoming call
from ${CALLERID} to
queue oe_ccare)
exten = oe_ccare,n,Set(TIMEOUT(response)=5)
exten = oe_ccare,n,
Re: [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ?
To convert the phone to SIP you have to unlock and set the TFTP server to your TFTP server address as explained below. But... you also need to make sure you have a SIP image in the root directory of your TFTP server and also edit the OS79XX.TXT file to contain only the SIP image name (no .bin or .sbin extension). This should tell the phone what image it should be running and will then start a tftp transaction to download the SIP image. Once the SIP image is loaded on your phone, you will have to reset the TFTP Server addresses again so it can then download the SIP configuration files (SIPdefault.cnf and SIPMAC.cnf) good luck! -- Original message -- From: Aaron Daniel [EMAIL PROTECTED] You have to press settings, then **#, and wait a moment to make sure it unlocks. Then you can configure a tftp server to use. The alternative is to configure your dhcp server with a tftp server. On linux, that would be next-server ip/host in the subnet section. TFTP server on windows. On Wed, 7 Jun 2006, Mateo Meier wrote: Hello Guys, I just got my new Cisco 7940* IP Phone Unfortunately I can't find out how to setup this phone. Am I really that stupid ;-) ? I've read the Cisco Manuel, but everything I tried did not help anything. 1. When I turn on the phone it will display Configuring VLANConfiguring IP.. This message will not disappear. 2. I can see that the phone has a local IP. I can also access the IP over my LAN with http (only http, telnet does not work) 3 My Main menu will this show Configuring VLANConfiguring IP.. But if I click on settings, network settings it will show me the local IP of the phone Now, my question, what do I do wrong ? how can I get that phone installed with a sip image ? I tried to unlock the phone with **# but that does not do anything. Also, there is no unlock function in the phone menu (phone settings) This is a new Cisco phone, no sip image on it. Thank you for the help Matt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Opinions of Sphinx?
Hi, I would like to get some Voice Recon working with a small asterisk server I have (10 endpoints). We would have no more than 3 calls at the MOST up trying to do recon, so volume is not a problem. Anyone have a link to the download AND a HOWTO for Sphinx /w Asterisk? Regards, -Eric -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Sunday, June 12, 2005 8:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Opinions of Sphinx? Hello, Very true, we actually convert the 8k 8bit wav files to 16 bit files and normalize the audio through SoX for sphinx. We spent a couple weeks developing a custom limited vocabulary and we do the recognition in 2 phases, the first phase is the quick and dirty, we then analyze the results of that and if our internal score is high enough to recognize the words we finalize that recording(about 90%), if not we put the non-recognized ones in a second batch to be analyzed at a higher level of analysis by sphinx. we then analyze and score those(about 50% finalize). so we are left with about 95% of our total recordings that finalize correctly with 5% that need to be manually listened to for confirmation. This was a rather complicated process to build with many stumbles along the way. The hardest part is the building of the vocabulary and the tuning of the analysis of the conversions. Here's the batch launching script, although that is a very small part of the whole that you need for this all to work: http://astguiclient.sourceforge.net/experimental_code/sphinx2_pltest.pl MATT--- -Original Message- From: Race Vanderdecken [mailto:[EMAIL PROTECTED] Sent: Saturday, June 11, 2005 9:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Opinions of Sphinx? Curious, which codec are you using with Sphinx? The smaller the bandwidth, generally, the harder it is to do recognition. Sphinx4 is a tool built on JAVA (one moment while I clear my throat and spit to get the coffee taste out of my mouth.) Write me offline; I am curious about doing batch reco for several projects, if you don't want to answer here. Race the tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Saturday, June 11, 2005 9:04 PM To: 'Brian Roy'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Opinions of Sphinx? We use batch sphinx to analyze recordings at night. We attempted real-time sphinx, but it is way too slow and resource-intensive to use for realtime on Asterisk with more than a couple lines at once(and that's at the poor quality settings). We have not tried sphinx4, but I wouldn't imagine that it would be that much improved in speed over version 3. MATT--- -Original Message- From: Brian Roy [mailto:[EMAIL PROTECTED] Sent: Saturday, June 11, 2005 6:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Opinions of Sphinx? On 5/31/05, Alistair Cunningham [EMAIL PROTECTED] wrote: Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Alistair, Well, it's been a couple weeks and no answers on the list. That isn't encouraging, but I'm hoping to accomplish some of the same thing. Have you made any progress on your own with this? Let me(us) know... Thanks, -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice recognition application - VoIP/Open Source
Hi all, Anyone knows of any Voice Recognition applications which use VoIP? Preferably open source. I am basically trying to build a Voice Call Router, something to recognize a spoken name and then transfer the caller to the right partys extension? TIA. -Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 SIP registration???
Looks like you have sip.conf set up to expect registrations for tycisco since it has a D for dynamic. You can either set up the 7960 to register with asterisk and use something like this in sip.conf: [tycisco] type=friend username= someusername secret= somesecret insecure=no mailbox=757 host=dynamic callerid= or just not have the 7960 register and specify its IP address using the host= line instead. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of List Receiver Sent: Wednesday, April 20, 2005 11:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 SIP registration??? So, here's my quandary: 1) Asterisk running CVS HEAD as of a couple days ago 2) Cisco 7960 SIP phones in a different subnet than the Asterisk server 3) NAT/Firewall device between 7960's and * I can initiate a call from the 7960's just fine. They can call out using our Broadvoice account and access any of the vmail stuff on *. When calling in from the outside world and dialing one of their extensions, however, I always get a this user is on the phone message. The console spits out this nugget: == CDR updated on SIP/4252780761-933d -- Executing Macro(SIP/4252780761-933d, stdsip|tycisco|101) in new stack -- Executing Dial(SIP/4252780761-933d, SIP/tycisco) in new stack Apr 20 08:14:59 NOTICE[32728]: app_dial.c:973 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) A showing of the sip peers: sip show peers Name/username HostDyn Nat ACL Mask Port Status rickcisco/cisco2 (Unspecified)D N 255.255.255.255 0UNKNOWN tycisco/cisco1 (Unspecified)D N 255.255.255.255 0UNKNOWN sip.broadvoice.com/425278 147.135.4.128 255.255.255.255 5060 OK (127 ms) 3 sip peers [1 online , 2 offline] I'm sure the reason I can't call to an extension is that they are appearing offline. How can I remedy this, however? I'm an * newbie, so go easy on me. :^) Thanks, Ty Christensen MCP, MCSP, MCSB Master Mind Productions Inc. www.mastermindpro.com http://www.mastermindpro.com/ (425) 378-7724 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbie - want to use asterisk as an internal PBX
Sounds like you have a context mis-match. Please posts your sip.conf and extensions.conf files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mak kwak Sent: Monday, April 04, 2005 10:56 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] newbie - want to use asterisk as an internal PBX Hallo. At the begining I would like to use asterisk as a VoIP server for some internal extensions inside one building without connection to external world. I planning to use kphone as soft phones. I tried to use configureation description that is described in http://asterisk.net.au/tutorial/1/ I'm running RH7.3, compiled and installed asterisk successfuly, compiled kphone. I set up all scripts according to the link above. /usr/sbin/asterisk -vvvgc : seems to be starging OK. When I run kphone I am able to login: trace from asterisk console: == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 Asterisk Ready. *CLI -- Registered SIP 'kphone' at 10.1.3.154 port 5062 expires 180 Problems start now. According to extensions.conf: ; echo test, to make sure your phone works. exten = 600,1,Playback(demo-echotest) ; Let them know what's going on exten = 600,2,Echo ; Do the echo test exten = 600,3,Playback(demo-echodone) ; Let them know it's over exten = 600,4,Goto(s,6) ; Start over I GUESS that when I dial 600, I should be able to hear echo when I'm talking, but unfortunatelly I cannot even dial the number. I tried many ways (10.1.3.154 - is my asterisk pbx): 600 [EMAIL PROTECTED] sip:[EMAIL PROTECTED] all the dials above produce asterisk log message: Apr 4 16:54:25 NOTICE[27916]: pbx.c:1329 pbx_extension_helper: Cannot find extension context 'voipmenu' and kphone says: call failed: not found My sound card works fine. I do not know what can I do more. Have You got any ideas. Greetings Jeste pracodawc? Szukasz pracownika? Zamie ogoszenie w Praca.wp.pl! Internet to skuteczne narzdzie rekrutacyjne! http://klik.wp.pl/?adr=http%3A%2F%2Fpraca.wp.pl%2Fzamiesc.htmlsid=345 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
