Re: [asterisk-users] Sound problem with format files but not codecs
Le 22/10/2012 04:27, Binan AL Halabi a écrit : Hello, It means that one of clients, is using 'silence suppression' mechanism which sends audio frames that do not contain any samples. Asterisk complains about silence supression and appears these warnings on CLI. If the client turn off the silence suppression the message will disappear. Hi Binan, silence suppression is already turned off Regards // Binan. *Från:* Administrator TOOTAI ad...@tootai.net *Till:* Asterisk-Users asterisk-users@lists.digium.com *Skickat:* söndag, 21 oktober 2012 10:34 *Ämne:* [asterisk-users] Sound problem with format files but not codecs Hi all, on asterisk 1.8.16 [2012-10-20 19:36:17] VERBOSE[743] pbx.c:-- Executing [801@OFFICE-Numbers:2] MusicOnHold(Local/801@OFFICE-Numbers-e54a;2, ) in new stack [2012-10-20 19:36:17] VERBOSE[743] res_musiconhold.c:-- Started music on hold, class 'TOOTAi', on Local/801@OFFICE-Numbers-e54a;2 [2012-10-20 19:36:17] WARNING[742] translate.c: no samples for ulawtolin [2012-10-20 19:36:21] VERBOSE[742] pbx.c: == Spawn extension (from_to-OFFICE, 801, 23) exited non-zero on 'SIP/8081773619-2f28' [2012-10-20 19:36:21] VERBOSE[743] res_musiconhold.c:-- Stopped music on hold on Local/801@OFFICE-Numbers-e54a;2 or asterisk 10.8.0 -- Executing [801@macro-GeneralNumbers:1] Set(SIP/105-0081, CHANNEL(musicclass)=TOOTAi) in new stack -- Executing [801@macro-GeneralNumbers:2] MusicOnHold(SIP/105-0081, ) in new stack -- Started music on hold, class 'TOOTAi', on SIP/105-0081 [2012-10-20 22:48:48] WARNING[22435]: translate.c:343 framein: no samples for g722tolin -- Stopped music on hold on SIP/105-0081 This is when calling extension: exten=801,1,Set(CHANNEL(musicclass)=TOOTAi) exten=801,n,MusicOnHold() exten=801,n,Hangup What does mean those WARNINGS and how to solve this problem? MeetMe, Voicemail or holding a call are working fine. From what I understand, codecs are used in channels and format for handling files. In both cases, two different servers, asterisk is compiled from tar.gz and in menuselect all codecs and formats are activated. Is this a bug? Did I forget something? On a third server I run latest Elastix with an asterisk 1.8.16 version. On this server I have no MusicOnHold at all even during calls. Logs show VERBOSE[19717] res_musiconhold.c:-- Started music on hold, class 'default', on SIP/104-00b3 VERBOSE[19717] res_musiconhold.c:-- Stopped music on hold on SIP/104-00b3 which is MusicOnHold stop immediately. On all servers wav files are installed, even try with original ones delivered with Asterisk. Thanks for any hint -- _ -- 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] monitor application, file name change on attended transfer
2012/10/22 Binan AL Halabi binanalhal...@yahoo.com Hi, You are using b flag in monitor command. This means don't begin recording untill call is bridged. So what you get if you delete this flag ? If I dont use the b flag then I get two separate files just like in the case when B waits till C answers before transfering call, but this is obvious because without b flag the monitor is started right away and when the time of transfer the second monitor is already stopped. It seems that when the channels are bridged after transfer the variables get mixed somehow and it affects the filename from monitor application. I think that when using b option I should not get any files from the second monitor application since that call is never answered. -- z poważaniem Grzegorz Pycia Administrator systemów contact center Thulium sp. z o.o. Na Skarpie 24 lok. 15, 31-910 Kraków tel. 123975301 www.thulium.pl -- _ -- 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] How can read the headers ISDN?
Hello all, My name is Danilo and I have a problem with the ISDN. I hope I have the wrong section. =P I have a CS1000 Nortel central with release 5.50. This central is attached to an Asterisk server with Sangoma PRI ISDN. I need to read the headers of ISDN and comes running from Nortel to Asterisk. How can I read them? Thank you, Danilo. -- -- _ -- 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 1.6.0 disable cdr account logs?
Just add noload=cdr_csv.so to modules.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JR Richardson Sent: Friday, October 19, 2012 5:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.6.0 disable cdr account logs? Hi All, I would like to disable the cdr account logs but in 1.6.0 but the 'accountlogs=no' switch is not available till 1.8 as far as I can tell. Is the any switch I can turn off int he Mkae file for the cdr_csv.so module to disable accountcode logs? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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
[asterisk-users] realtime sip peers status
Dear All, I have successfully setup Asterisk realtime. Now I can create extensions dynamically. But when I put this command on cli mode sip show peers it returns no result. can any one guide me to fix this problem. Thanks-- _ -- 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] realtime sip peers status
On Mon, 2012-10-22 at 19:17 +0500, Control Oye wrote: Dear All, I have successfully setup Asterisk realtime. Now I can create extensions dynamically. But when I put this command on cli mode sip show peers it returns no result. can any one guide me to fix this problem. Thanks -- The extensions you have created will not show up in the cli command of sip show peers until the sip extensions have tried to connect to the asterisk server. Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] realtime sip peers status
On 22 Oct 2012, at 15:21, Ishfaq Malik wrote: On Mon, 2012-10-22 at 19:17 +0500, Control Oye wrote: I have successfully setup Asterisk realtime. Now I can create extensions dynamically. But when I put this command on cli mode sip show peers it returns no result. can any one guide me to fix this problem. The extensions you have created will not show up in the cli command of sip show peers until the sip extensions have tried to connect to the asterisk server. You may also need to cache realtime peers for some of the stats you're probably after. There are plenty of guides online for this. Google is your friend. Steve -- _ -- 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] monitor application, file name change on attended transfer
Grzegorz Pycia wrote:
Hi
I have some problem with monitor application when call i transferred
in
attended mode and the transfer occurs before call is answered.
Here is how it looks:
A calls B(let's assume ${UNIQUEUEID}=1)
exten = _,1,NoOp
seme = n,Set(MONITOR_FILENAME=call-${UNIQUEID})
same =
n,monitor(alaw,/var/spool/asterisk/monitor/${MONITOR_FILENAME},bm)
When B answers the call, files call-1-in* and call1-out* are created.
During The call, B tries to make attended transfer A is put on hold
and
B calls C using the same dialplan logic:
B calls C(let's assume ${UNIQUEUEID}=2)
At the time off invoking monitor application none off the call-2
channels are monitored so the monitor application starts without
errors,
if B waits till C answers, everything is OK monitor starts recording
and
files call-2-in* and call-2-out* are created, When B transfers the
call
call-2 monitor is stopped. And call-2 files contain only the call
between B and C.
But there is problem when B does not wait until C answers the call,
if
transfer is done before C answers the call, the call-2* are not
created
and the call is still recorded to the call-1* files, but when the
transferred call between A and C ends, the call-1* files get renamed
to
call-2* and the MONITOR_EXEC application is called with call-2* file
names as parameters.
This makes it impossible to locate the call record since the file
names
get changed, can someone tell if I should file a BUG report or is it
intended to act like this?
Regards
Are you using Asterisk 1.8 or higher? A good way to mitigate this
would be to use MixMonitor. It applies as an audiohook which should
persist through transfers like the one you described, so you would
just need to set AUDIOHOOK_INHERIT for MixMonitor in order to use it
that way. One difference with this approach though would be that
MixMonitor will automatically mix audio from both ends of the call
into a single recording. That behavior can be worked around starting
with Asterisk 10 by using the r and t options.
I guess it's worth noting that if you aren't using 1.8 or higher
there isn't really any point in filing a bug report since earlier
versions aren't supported anymore.
--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct +1 256 428 6139
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] monitor application, file name change on attended transfer
I'm using latest 1.8, althought I did check and this behaviour is the same
since 1.6.2.11. I will file a bug report about it in 1.8.17.0.
Auto Mixing would not bother me, i will check the Mix monitor.
Regards.
22 paź 2012 17:22, Jonathan Rose jr...@digium.com napisał(a):
Grzegorz Pycia wrote:
Hi
I have some problem with monitor application when call i transferred
in
attended mode and the transfer occurs before call is answered.
Here is how it looks:
A calls B(let's assume ${UNIQUEUEID}=1)
exten = _,1,NoOp
seme = n,Set(MONITOR_FILENAME=call-${UNIQUEID})
same =
n,monitor(alaw,/var/spool/asterisk/monitor/${MONITOR_FILENAME},bm)
When B answers the call, files call-1-in* and call1-out* are created.
During The call, B tries to make attended transfer A is put on hold
and
B calls C using the same dialplan logic:
B calls C(let's assume ${UNIQUEUEID}=2)
At the time off invoking monitor application none off the call-2
channels are monitored so the monitor application starts without
errors,
if B waits till C answers, everything is OK monitor starts recording
and
files call-2-in* and call-2-out* are created, When B transfers the
call
call-2 monitor is stopped. And call-2 files contain only the call
between B and C.
But there is problem when B does not wait until C answers the call,
if
transfer is done before C answers the call, the call-2* are not
created
and the call is still recorded to the call-1* files, but when the
transferred call between A and C ends, the call-1* files get renamed
to
call-2* and the MONITOR_EXEC application is called with call-2* file
names as parameters.
This makes it impossible to locate the call record since the file
names
get changed, can someone tell if I should file a BUG report or is it
intended to act like this?
Regards
Are you using Asterisk 1.8 or higher? A good way to mitigate this
would be to use MixMonitor. It applies as an audiohook which should
persist through transfers like the one you described, so you would
just need to set AUDIOHOOK_INHERIT for MixMonitor in order to use it
that way. One difference with this approach though would be that
MixMonitor will automatically mix audio from both ends of the call
into a single recording. That behavior can be worked around starting
with Asterisk 10 by using the r and t options.
I guess it's worth noting that if you aren't using 1.8 or higher
there isn't really any point in filing a bug report since earlier
versions aren't supported anymore.
--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct +1 256 428 6139
Check us out at: http://digium.com http://asterisk.org
--
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Re: [asterisk-users] Agents in more than one queue at once
In general there is no guaarantee as which call will connect; each queue is independent AFAIK. Lenz- big fan :) And I'm sure this topic is of interest to you... I'll admit, I had a feeling that it's random would be the response to my original question. I remember reading the app_queue code a while back and getting the impression that the logic was something like- * Loop through the list of all queued calls in the system, one by one. If the current call is 'next' in its respective queue (as defined by the queue scheduling algo), then we need to find an agent to take this call... * Loop through the list of agents who are members of this queue. If the current agent is available to take a call, then send the call to the available agent. This logic leaves the above mentioned oversight where at no point is it ever considered that an agent may be a member of more than one queue. Even with a fair scheduling algo applied to each queue, this bug causes queues with large numbers of waiting calls and/or large numbers of available agents to starve other queues. Effectively, this bug makes skills-based routing impossible, because unique skills can not exist. Additionally, regardless of how many callers are in queue, this bug causes hold times to increase - significantly, in our case. This is an important oversight in my opinion because it is the only way that skills-based routing can be implemented within the framework of app_queue. Without fixing this issue, Asterisk can not claim to have a reliable method of implementing skills-based routing. *DEVELOPERS* - If I took a crack at fixing this issue, what general tips do you have for me to make it most likely that my solution can be integrated into HEAD? I believe I can justify spending some time at work to deal with this, but not without at least a decent chance that the work will be integrated into mainline (assuming it doesn't suck, of course :) Alex Forster -- _ -- 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 1.6.0 disable cdr account logs?
Just add noload=cdr_csv.so to modules.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JR Richardson Sent: Friday, October 19, 2012 5:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.6.0 disable cdr account logs? Hi All, I would like to disable the cdr account logs but in 1.6.0 but the 'accountlogs=no' switch is not available till 1.8 as far as I can tell. Is the any switch I can turn off int he Mkae file for the cdr_csv.so module to disable accountcode logs? Correct me if I'm wrong, if I noload=cdr_csv.so, won't that disable all csv CDR's. I still want the Master CSV file with account code, what I don't want is a seperate CSV CDR for each accountcode generated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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] Agents in more than one queue at once
Il 22/10/2012 18:44, Alex Forster ha scritto: *DEVELOPERS* - If I took a crack at fixing this issue, what general tips do you have for me to make it most likely that my solution can be integrated into HEAD? I believe I can justify spending some time at work to deal with this, but not without at least a decent chance that the work will be integrated into mainline (assuming it doesn't suck, of course:) Nice to hear you are willing to work on it. I suggest you to ask on asterisk-dev ;) Cheers, Niccolò -- http://www.linuxsystems.it -- _ -- 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] asterisk and mp3 on 1.4.43
If I am using asterisk (server) and then asterisk on client (sound port) and I want to get the best MP3 sound I can get - how can I do that with ulaw codec and wav file conversion. I used gst-launch to convert my MP3 to WAV (16K and mono) then playing over ulaw to the other client. I know mono will not sound as good but I am trying to get the best sound I can. Is there a better way to do this? THanks, Jerry -- _ -- 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 1.6.0 disable cdr account logs?
My bad. I sent Igor to the boneyard to fetch 1.6.0.28 and it appears to me that by commenting out lines 309-312 and doing a fresh make you eliminate the extra files (or make them empty). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JR Richardson Sent: Monday, October 22, 2012 11:58 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.6.0 disable cdr account logs? Just add noload=cdr_csv.so to modules.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JR Richardson Sent: Friday, October 19, 2012 5:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.6.0 disable cdr account logs? Hi All, I would like to disable the cdr account logs but in 1.6.0 but the 'accountlogs=no' switch is not available till 1.8 as far as I can tell. Is the any switch I can turn off int he Mkae file for the cdr_csv.so module to disable accountcode logs? Correct me if I'm wrong, if I noload=cdr_csv.so, won't that disable all csv CDR's. I still want the Master CSV file with account code, what I don't want is a seperate CSV CDR for each accountcode generated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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 and mp3 on 1.4.43
Your best bet in 1.4.X is going to be to use a LAME/SOX combination to convert your MP3's to wav/ulaw files. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, October 22, 2012 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk and mp3 on 1.4.43 If I am using asterisk (server) and then asterisk on client (sound port) and I want to get the best MP3 sound I can get - how can I do that with ulaw codec and wav file conversion. I used gst-launch to convert my MP3 to WAV (16K and mono) then playing over ulaw to the other client. I know mono will not sound as good but I am trying to get the best sound I can. Is there a better way to do this? THanks, Jerry -- _ -- 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
[asterisk-users] Voicemail to text for Asterisk
A customer has asked us to provide that feature. I know there are a few methods and products out there, but I haven't paid attention in a while. It is for about 300 users, and we'll consider open as well as paid-for products. We would prefer to pay for supported products as the cost will be passed on to the customer and they are willing to pay for quality. Do not want any complex scripting screwing around with third parties and such. Your ideas welcome. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Voicemail to text for Asterisk
Unless I missed something, there isn't anything out there that is as cheap or reliable as human translation in this case. If I did miss it, I know somebody will correct me. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Monday, October 22, 2012 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voicemail to text for Asterisk A customer has asked us to provide that feature. I know there are a few methods and products out there, but I haven't paid attention in a while. It is for about 300 users, and we'll consider open as well as paid-for products. We would prefer to pay for supported products as the cost will be passed on to the customer and they are willing to pay for quality. Do not want any complex scripting screwing around with third parties and such. Your ideas welcome. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Voicemail to text for Asterisk
On Mon, Oct 22, 2012 at 2:16 PM, Carlos Alvarez car...@televolve.comwrote: A customer has asked us to provide that feature. I know there are a few methods and products out there, but I haven't paid attention in a while. It is for about 300 users, and we'll consider open as well as paid-for products. We would prefer to pay for supported products as the cost will be passed on to the customer and they are willing to pay for quality. Do not want any complex scripting screwing around with third parties and such. Your ideas welcome. All automated solutions -- paid or free -- are terrible. The technology simply does not exist at this point at a level that is acceptable to most customers. If quality is paramount, you are better off doing the transcription in-house with a human. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Voicemail to text for Asterisk
On Mon, Oct 22, 2012 at 12:40 PM, Christopher Harrington ch...@acsdi.comwrote: All automated solutions -- paid or free -- are terrible. The technology simply does not exist at this point at a level that is acceptable to most customers. If quality is paramount, you are better off doing the transcription in-house with a human. In-house transcriptions are definitely out of the question, but any experience with outsourced solutions would be useful. As far as I can tell the current service is automated, and as awful as Google Voice, yet they find it useful. Their existing carrier uses Broadsoft and I'm not sure if they have that built in. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Voicemail to text for Asterisk
Carlos I have tried several solutions and non of them have been worth the money. I have worked with transcription companies and they are the best but they are expensive. If you do find something that works let the groups know as there are a few of us out here that are looking for that holy grail of speech to text. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Carlos Alvarez car...@televolve.com Sent: Monday, October 22, 2012 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail to text for Asterisk On Mon, Oct 22, 2012 at 12:40 PM, Christopher Harrington ch...@acsdi.com wrote: All automated solutions -- paid or free -- are terrible. The technology simply does not exist at this point at a level that is acceptable to most customers. If quality is paramount, you are better off doing the transcription in-house with a human. In-house transcriptions are definitely out of the question, but any experience with outsourced solutions would be useful. As far as I can tell the current service is automated, and as awful as Google Voice, yet they find it useful. Their existing carrier uses Broadsoft and I'm not sure if they have that built in. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Voicemail to text for Asterisk
On Mon, 22 Oct 2012 12:47:51 -0700 Carlos Alvarez car...@televolve.com wrote: In-house transcriptions are definitely out of the question, but any experience with outsourced solutions would be useful. As far as I can tell the current service is automated, and as awful as Google Voice, yet they find it useful. Their existing carrier uses Broadsoft and I'm not sure if they have that built in. Voice recognition for asterisk based on Google speech API is already available[1], the problem with this service is that it's limited to 20-30 seconds of speech data, which isn't suitable for transcripting voicemails. If you are able to find a reliable way of chopping speech samples in segments no bigger than 20 seconds based on silence detection, so words wont be cut in half, you might come up with something very similar to Google Voice transcription service. But I would recommend against using this into production since google haven't yet defined the terms of service for speech recognition, and its more or less a hack for now. [1] http://zaf.github.com/asterisk-speech-recog/ Lefteris Zafiris -- _ -- 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] Call Presence for Offhook/Onhook Only
We have a customer with a dozen phones and they want nearly all of them to ring.Unfortunately this causes a firestorm of call presence notifications that overwhelm something on their network. Any existing calls get gappy audio for a few milliseconds when a new call comes in and when someone picks it up due to all the state changes between ringing and not ringing. They have a T-1 dedicated to voice so it isn't a bandwidth issue per se. We've been through a handful of routers and QOS settings but nothing has worked. Turning off the busy lamps fixes the problem but of course that isn't really a long term solution. Really I don't think anyone cares about the busy lamps for ringing. They just want to know when someone is on the phone. Is there any way short of hacking code that we can make notifications ignore changes involving ringing and just report inuse/notinuse? We are using 1.8.x if that matters. Chris -- - Chris Owen- Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] Voicemail to text for Asterisk
On Mon, Oct 22, 2012 at 3:05 PM, Lefteris Zafiris zaf@gmail.com wrote: If you are able to find a reliable way of chopping speech samples in segments no bigger than 20 seconds based on silence detection, so words wont be cut in half, you might come up with something very similar to Google Voice transcription service. Unfortunately Google's transcription is vastly improved by its context comprehension (for instance, understanding that the word phone is likely to be followed by words like call or number) and chopping up the audio, even between words, will reduce that context data for the transcriber. Good luck, anyway. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Call Presence for Offhook/Onhook Only
You could do a simple PHP/Perl script to query hints and ring only the not-in-use phones. Or more simply that that do a ChanIsAvail() against the list and ring the returned array. If I do ChanIsAvail(line1/line2/line3/line4/line5) and 1 and 3 are in use, it returns an array with 2/4/5 and I can Dial the array. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Owen Sent: Monday, October 22, 2012 3:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call Presence for Offhook/Onhook Only We have a customer with a dozen phones and they want nearly all of them to ring.Unfortunately this causes a firestorm of call presence notifications that overwhelm something on their network. Any existing calls get gappy audio for a few milliseconds when a new call comes in and when someone picks it up due to all the state changes between ringing and not ringing. They have a T-1 dedicated to voice so it isn't a bandwidth issue per se. We've been through a handful of routers and QOS settings but nothing has worked. Turning off the busy lamps fixes the problem but of course that isn't really a long term solution. Really I don't think anyone cares about the busy lamps for ringing. They just want to know when someone is on the phone. Is there any way short of hacking code that we can make notifications ignore changes involving ringing and just report inuse/notinuse? We are using 1.8.x if that matters. Chris -- - Chris Owen- Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- 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] Call Presence for Offhook/Onhook Only
Check the notifyringing option in sip.conf -Original Message- From: Chris Owen ow...@hubris.net Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 22 Oct 2012 15:17:27 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Call Presence for Offhook/Onhook Only -- _ -- 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] Voicemail to text for Asterisk
On 22/10/2012 at 16:02 -0400, Bryant Zimmerman wrote: Carlos I have tried several solutions and non of them have been worth the money. I have worked with transcription companies and they are the best but they are expensive. If you do find something that works let the groups know as there are a few of us out here that are looking for that holy grail of speech to text. There is no holy grail yet, speech technology deployment requires a close cooperation between the speech technology provider and the users. It's not plug and play but after some joint efforts automated transcriptions must be useful. If anyone wants to experiment with CMUSphinx-based automated solution to transcribe voicemails, drop me a note. The results could be pretty interesting. 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] Voicemail to text for Asterisk
On Mon, Oct 22, 2012 at 2:43 PM, Nickolay V. Shmyrev nshmy...@nexiwave.comwrote: There is no holy grail yet, speech technology deployment requires a close cooperation between the speech technology provider and the users. It's not plug and play but after some joint efforts automated transcriptions must be useful. If anyone wants to experiment with CMUSphinx-based automated solution to transcribe voicemails, drop me a note. The results could be pretty interesting. We will probably give it a try, though not sure when. Probably in about a month. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Call Presence for Offhook/Onhook Only
On Oct 22, 2012, at 4:25 PM, Danny Nicholas da...@debsinc.com wrote: You could do a simple PHP/Perl script to query hints and ring only the not-in-use phones. Or more simply that that do a ChanIsAvail() against the list and ring the returned array. If I do ChanIsAvail(line1/line2/line3/line4/line5) and 1 and 3 are in use, it returns an array with 2/4/5 and I can Dial the array. I don't think the problem isn't the phones that are in using getting the notifications so much as just the shear number of phones getting them. If we only ring a few phones the problem goes away even if the phone you are on is one of the ones getting the notifications. Chris -- - Chris Owen- Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] Call Presence for Offhook/Onhook Only
On Oct 22, 2012, at 4:11 PM, isr...@gmail.com wrote: Check the notifyringing option in sip.conf Interesting. Looks like exactly what I want other than it looks like it is a global only setting? I'll play with it tonight but any idea if this is still global only? Chris -- - Chris Owen- Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] Call Presence for Offhook/Onhook Only
On Oct 22, 2012, at 4:11 PM, isr...@gmail.com wrote: Check the notifyringing option in sip.conf Looks like this really doesn't do what I had hoped: ;notifyringing = no ; Control whether subscriptions already INUSE get sent ; RINGING when another call is sent (default: yes) Chris -- - Chris Owen- Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] Call drop weirdness
I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get a data dump to see what actually happens. Some times there is a bit of artifacting which takes place just prior to the drop, but mostly nothing: it just drops. I've checked and rechecked firewall settings. Bandwidth consumption on the Inet link varies, but the dropped audio happens even on off-peak times. I'm considering giving the Asterisk box a public IP on one IF and bypassing the FW to rule out NAT weirdness. Any thoughts on things to look at would be greatly appreciated. Kind Regards, Chris -- _ -- 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
