Re: [asterisk-users] AMD Not Working
Hi,
Thanks for your reply.
I have tried as you suggested.
In "h" extension it is giving Status as AMD_HANGUP.
Below is the log
-- Executing Answer("SIP/sip-874d", "") in new stack
-- Executing AMD("SIP/sip-874d", "") in new stack
-- AMD: SIP/sip-874d (null) (null) (Fmt: 4)
Apr 28 00:53:41 NOTICE[5837]: app_amd.c:134 isAnsweringMachine: AMD using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300]
totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP
-- Executing NoOp("SIP/sip-874d", "Status: AMD_HANGUP Cause: ") in new
stack
vm3*CLI>
Any help is highly appreciated.
Thanks.
On Mon, Apr 27, 2009 at 5:04 PM, Matt Riddell wrote:
> On 27/04/2009 4:22 p.m., Sam Hawkin wrote:
> > Hi,
> >
> > Thanks for your reply.
> >
> > I have tried as you suggested, I does not even come upto NoOp()
> > It hangups after AMD.
> > I have decreased the silence threshold from 256 to 100 and 50.
>
> Try the NoOp in the h extension:
>
> exten => h,1,NoOp(Status: ${AMDSTATUS} Cause: ${AMDCAUSE})
>
> --
> Kind Regards,
>
> Matt Riddell
> Director
> ___
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Re: [asterisk-users] Asterisk & EC2
On Tue Apr 28 2009 09:19:56 GMT+1000 (EST) Eric Chamberlain wrote: > > The original Feodra 8 image came from the Amazon EC2 team, they > optimized it to run in EC2. I chose the Amazon fc8 image, because I'm > not comfortable getting OS images from third-parties. When Amazon > releases new images, I'll update the guide. You might want to > consider installing fc8 then upgrading to a newer release. Thanks for the explanation for using FC8 Eric. It makes sense. You might also look into Ubuntu now. Ubuntu has an official EC2 release now, and they publish their AMIs and maintain them. > To build DAHDI kernel modules, all you have to do is setup your OS of > choice to use a build environment that matches the Amazon kernel you > are using. > > > > Latency is pretty route specific. Amazon has good bandwidth, but I > would avoid proxying media if possible. > > We haven't had any reliability problems with EC2 hosting our Asterisk > real-time application servers. > That's good to know. It's good to know that Asterisk is being used in a production environment in EC2. And thanks for the guideline. Cheers -- Aryan ___ -- 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] music on hold using mms
Thanks. But I heard that mpg123 uses much resources (CPU & memory) of each connection. Is it true? How about using madplay? On 4/28/09, M Hulber wrote: > Didn't do mms but have implemented using Shoutcast. I have instructions > at the link below: > > http://mark.hulber.com/voip/configuration/shoutcast-musiconhold-in-asterisk-16/ > > > Rilawich Ango wrote: >> Hi, >> I follow the >> web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf >> - mohstream.sh , to configure music on hold to play using mms but >> failed. Anyone can play using mms? >> ango >> >> ___ >> -- 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 >> > > -- > > MARK. > > Hulber Technologies > asterisk-ad...@hulber.com > > Read my blog : http://mark.hulber.com > Follow @hulber on Twitter: http://twitter.com/hulber > > > ___ > -- 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
[asterisk-users] POS modems
Hi, If anyone is interested in the low speed modems needed for POS applications (V.22, V.22bis, V.22bisFC and V.29FC) please contact me. I had some spare time while travelling, and finally got the V.22bis code I started a long time ago into a start where its basically functional. I'm now looking for input about exactly what application software expects from these modems, so I can plan the remainder of the code. Steve ___ -- 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] Asterisk & EC2
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Chamberlain Sent: April-27-09 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk & EC2 On Apr 25, 2009, at 10:31 PM, Aryan Ameri wrote: > > The second one, is built on a custom Fedora 8 image. The steps are not > repeatable on any other distro, not even a stock official Fedora 8 > one. Fedora > 8 itself is long EOLed and as such, not something I'd want to use on a > production server. Dahdi compilation as described on that guide > doesn't work > on CentOS or Ubuntu or Debian. Aryan, The original Feodra 8 image came from the Amazon EC2 team, they optimized it to run in EC2. I chose the Amazon fc8 image, because I'm not comfortable getting OS images from third-parties. When Amazon releases new images, I'll update the guide. You might want to consider installing fc8 then upgrading to a newer release. To build DAHDI kernel modules, all you have to do is setup your OS of choice to use a build environment that matches the Amazon kernel you are using. > > > Besides, I asked about anecdotal usage experiences running Asterisk > on EC2. > About whether latency is an issue if extensions are outside the EC2 > availability zone. About reliability of EC2 when used to host a real- > time > application server. Not just an installation guideline. Latency is pretty route specific. Amazon has good bandwidth, but I would avoid proxying media if possible. We haven't had any reliability problems with EC2 hosting our Asterisk real-time application servers. -- Eric Chamberlain, Founder RF.com - http://RF.com/ You do realize that your initials are EC ;) so amazon is your next logical step BTW trying EC2 for 1 week now, and its running better at 1 unit, then most dedicated boxes out there. ___ -- 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] Asterisk & EC2
On Apr 25, 2009, at 10:31 PM, Aryan Ameri wrote: > > The second one, is built on a custom Fedora 8 image. The steps are not > repeatable on any other distro, not even a stock official Fedora 8 > one. Fedora > 8 itself is long EOLed and as such, not something I'd want to use on a > production server. Dahdi compilation as described on that guide > doesn't work > on CentOS or Ubuntu or Debian. Aryan, The original Feodra 8 image came from the Amazon EC2 team, they optimized it to run in EC2. I chose the Amazon fc8 image, because I'm not comfortable getting OS images from third-parties. When Amazon releases new images, I'll update the guide. You might want to consider installing fc8 then upgrading to a newer release. To build DAHDI kernel modules, all you have to do is setup your OS of choice to use a build environment that matches the Amazon kernel you are using. > > > Besides, I asked about anecdotal usage experiences running Asterisk > on EC2. > About whether latency is an issue if extensions are outside the EC2 > availability zone. About reliability of EC2 when used to host a real- > time > application server. Not just an installation guideline. Latency is pretty route specific. Amazon has good bandwidth, but I would avoid proxying media if possible. We haven't had any reliability problems with EC2 hosting our Asterisk real-time application servers. -- Eric Chamberlain, Founder RF.com - http://RF.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
Re: [asterisk-users] Who has the clever Polycom upgrade system?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Darrick Hartman (lists) wrote: > That would be Karl Fife, of the famous Karl Fife experience. > > http://kfife.com/voip/ That's what I'm looking for. Thanks Darrick! Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ9i/KCFu3bIiwtTARAnuQAJsHx/fRb/n6EnEj0pco1eY0wgEcugCcDTTY NrTQOrDBYfVbpqyO6LMkIW0= =GaAB -END PGP SIGNATURE- ___ -- 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] Who has the clever Polycom upgrade system?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Kevin P. Fleming wrote: > It's easy; just don't edit the files that come with the firmware! Hi Kevin. That's the model I currently use. The one I'm interested in is linked in Darrick's post below. It's an interesting approach. Thanks for replying! Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ9i+2CFu3bIiwtTARAj0wAKCP8K5NnEnHawuA5q5k0Aq1bKBe8QCeLBz7 e/xvLf7N0Ofl1Ic7Q9/ZT5A= =yDXt -END PGP SIGNATURE- ___ -- 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] IPv6 support?
On Tue, 2009-04-28 at 09:08 +1200, Andrew Ruthven wrote: > Hey, > > Just wondering if anyone can let me know what the status of IPv6 support > for Asterisk is currently. I see that the branch where development was > happening has gone away. I was trying: > > http://svn.digium.com/svn/asterisk/team/blanchet/v6 > > Has this branched moved to somewhere else? > > Cheers! > Sometime ago i got this status from _the_ guru... Russell replied, referencing 1.6.2... (but other code might get in-the-way) From: Russell Bryant > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Asterisk and IPv6 > Date: Mon, 28 Jan 2008 18:45:42 -0600 > > Hans Witvliet wrote: > > Any progress on IPv6 ? > > Still completely seperate code, or is it already being merged into the > > tree... > > > > Perhaps i overlooked it, but i couldn't find any reference in: > > http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co > > There has been progress. It is not yet merged into the main tree, though. I > would expect it to go in within the first few releases of 1.6 ... > ___ -- 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] Who has the clever Polycom upgrade system?
Barry L. Kline wrote: > I remember someone wrote a great document concerning Polycom server > provisioning that provided a way to ensure that updates to the firmware > did not overwrite customizations. I'll be damned if I can remember > where I saw it. It may have been discussed during a VUC session or may > have been on this list. It's easy; just don't edit the files that come with the firmware! Instead, make all your modifications by copying the sections you want to change into one (or more) new files, and then in the very first files the phones download (.cfg) ensure that your customized files are listed *before* the standard sip.cfg. This will make your settings take precedence over the standard settings. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- 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] Who has the clever Polycom upgrade system?
Barry L. Kline wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > I remember someone wrote a great document concerning Polycom server > provisioning that provided a way to ensure that updates to the firmware > did not overwrite customizations. I'll be damned if I can remember > where I saw it. It may have been discussed during a VUC session or may > have been on this list. > > Either way, I'm unable to google my way to it. Can anyone point me in > the right direction? That would be Karl Fife, of the famous Karl Fife experience. http://kfife.com/voip/ Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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
[asterisk-users] Who has the clever Polycom upgrade system?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I remember someone wrote a great document concerning Polycom server provisioning that provided a way to ensure that updates to the firmware did not overwrite customizations. I'll be damned if I can remember where I saw it. It may have been discussed during a VUC session or may have been on this list. Either way, I'm unable to google my way to it. Can anyone point me in the right direction? Thanks! Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD4DBQFJ9iSWCFu3bIiwtTARAlH2AJjcCtRPi9dyqwY0p2AqCZelgskIAKCVeuSV ++7hraanUhxNBF2RvIMmRg== =Hg/L -END PGP SIGNATURE- ___ -- 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
[asterisk-users] Where I get free VoiP-in numbers?
Hi list, Anyone knows how to get free VoiP-in numbers from USA or Canada, I have found some links for example sipnumber.com but it does not run. Also I want to know how to configure it in my asterisk server. Thanks in advance. Regards ___ -- 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
[asterisk-users] IPv6 support?
Hey, Just wondering if anyone can let me know what the status of IPv6 support for Asterisk is currently. I see that the branch where development was happening has gone away. I was trying: http://svn.digium.com/svn/asterisk/team/blanchet/v6 Has this branched moved to somewhere else? Cheers! -- Andrew Ruthven, Wellington, New Zealand At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz GPG fpr: 34CA 12A3 C6F8 B156 72C2 D0D7 D286 CE0C 0C62 B791 signature.asc Description: This is a digitally signed message part ___ -- 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] Packet2packet bridging while in sip.conf canreinvite=no
jonas kellens wrote:
> I have put canreinvite=no for all my internal SIP-clients in sip.conf
> because I want Asterisk to be in the middle of the RTP-stream so he can
> provide MusiconHold and so...
>
> Now, what the Asterisk CLI tells me when I make a call from my one
> internal SIP-phone to another internal SIP-phone is :
>
> Verbosity is at least 25
> == Spawn extension (intern, 51, 1) exited non-zero on 'SIP/BT201-088f93e0'
> -- Executing [...@intern:1] Dial("SIP/GXP1200-088f93e0",
> "SIP/BT201|30") in new stack
> -- Called BT201
> -- SIP/BT201-088faa00 is ringing
> -- SIP/BT201-088faa00 answered SIP/GXP1200-088f93e0
> *-- Packet2Packet bridging SIP/GXP1200-088f93e0 and SIP/BT201-088faa00*
> == Spawn extension (intern, 52, 1) exited non-zero on
> 'SIP/GXP1200-088f93e0'
>
> Why is there this native bridging ? Does this mean that Asterisk is no
> longer in the middle of it ?
It is important to note that Packet2Packet bridging is not the same as native
bridging. With native bridging, the audio flows outside of Asterisk between the
endpoints. With P2P bridging, the audio comes into the RTP layer of Asterisk
but
does not pass through the Asterisk core. This allows for Asterisk to intercept
DTMF or play warning files to the bridged parties.
>
> Also : there is no audio at all ! Just when I put down the phone there's
> the DTMF-signal that the line is cancelled...
SIP debug would probably help.
>
> Everything worked well before I edited musiconhold.conf and
> features.conf (to create a park extension).
Looking at your musiconhold.conf file, it looks very much like the sample
musiconhold.conf file. I doubt that your changes there would have affected
anything. If you say that the problems started when you edited features.conf,
then I would suggest that you start undoing the changes you made one-by-one to
see if you can find what change it was that caused the problem to occur.
[sample configs snipped]
>
> Do you need extra info ??
> What setting can I have set in musiconhold.conf or features.conf to
> affect the audiostream between my clients ???
There is nothing you can set in musiconhold.conf to control the media stream.
With SIP, the signalling still goes through Asterisk even if the media does
not.
Even if Asterisk is not in the media path, the endpoints can still signal to
Asterisk to play MOH to the other side. Asterisk can accomplish this through
reinvites.
Also, there is nothing you can set in features.conf to control the media
stream.
Settings pertaining to the media stream are channel-driver-specific and are
thus
configured in each particular channel driver's configuration file. As you have
already discovered, the setting which forces media onto Asterisk during a SIP
call is the canreinvite setting.
Mark Michelson
>
> Before I could call all my clients, I had musiconhold when putting 'on
> hold' and I was just figuring out how parked calls worked...
>
> Thanks for the help !
>
> Jonas Kellens.
>
>
>
>
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Re: [asterisk-users] Change Termination of Read Command
Daniel Hazelbaker wrote:
> On Apr 27, 2009, at 10:29 AM, Danny Nicholas wrote:
>
>> Greetings all,
>>This is a “just-for-fun” question. I was reading
>> the support forum and a fellow there wanted Read() to stop on *
>> instead of #. I thought that changing app_read.c would resolve this
>>
>> current
>> if (tmp[x-1] == '#') {
>> tmp[x-1] = '\0';
>> break;
>>
>> new
>> }if (tmp[x-1] == '*') {
>> tmp[x-1] = '\0';
>> break;
>> }
>>
>> He applied and recompiled, but no joy. Any ideas why?
>
> Without knowing where in the file this came from I can't say for sure,
> but that code looks to me like the code that would run after the digits
> are received and is stripping off the # character at the end, if it is
> there. Further up (or somewhere else entirely) there is probably a spot
> that actually terminates the read command when # is pressed.
>
> Daniel
>
Daniel is correct in his analysis. If you want app_read to terminate on a '*'
instead of a '#' then you will need to change the ast_readstring call inside of
ast_app_getdata (which is called from read_exec in app_read). This will have
the
side effect of making other situations use a * instead of a # as well (like
entering voicemail mailbox and entering an agent password).
Mark Michelson
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[asterisk-users] Packet2packet bridging while in sip.conf canreinvite=no
I have put canreinvite=no for all my internal SIP-clients in sip.conf
because I want Asterisk to be in the middle of the RTP-stream so he can
provide MusiconHold and so...
Now, what the Asterisk CLI tells me when I make a call from my one
internal SIP-phone to another internal SIP-phone is :
Verbosity is at least 25
== Spawn extension (intern, 51, 1) exited non-zero on
'SIP/BT201-088f93e0'
-- Executing [...@intern:1] Dial("SIP/GXP1200-088f93e0", "SIP/BT201|
30") in new stack
-- Called BT201
-- SIP/BT201-088faa00 is ringing
-- SIP/BT201-088faa00 answered SIP/GXP1200-088f93e0
-- Packet2Packet bridging SIP/GXP1200-088f93e0 and
SIP/BT201-088faa00
== Spawn extension (intern, 52, 1) exited non-zero on
'SIP/GXP1200-088f93e0'
Why is there this native bridging ? Does this mean that Asterisk is no
longer in the middle of it ?
Also : there is no audio at all ! Just when I put down the phone there's
the DTMF-signal that the line is cancelled...
Everything worked well before I edited musiconhold.conf and
features.conf (to create a park extension).
My sip.conf :
[r...@asterisk asterisk]# cat sip.conf
[general]
context=default
port=5060
bindaddr=192.168.4.248
srvlookup=yes
disallow=all
allow=alaw
allow=gsm
allow=ulaw
language=be
[BT201]
type=friend
context=intern
host=dynamic
username=BT201
secret=testpaswoord
canreinvite=no
callerid=Jonas Kellens <52>
qualify=yes
[GXP1200]
type=friend
context=intern
host=dynamic
username=GXP1200
secret=testpaswoord
canreinvite=no
callerid=callerid? <51>
qualify=yes
[GXP2020]
type=friend
context=intern
host=dynamic
username=GXP2020
secret=testpaswoord
canreinvite=no
callerid=Kristof Teirlinck <50>
qualify=yes
Musiconhold.conf :
[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes
Features.conf :
; Sample Call Features (parking, transfer, etc) configuration
;
[general]
parkext => 90 ; What extension to dial to park
parkpos => 91-95; What extensions to park calls on. These needs
to be
; numeric, as Asterisk starts from the start
position
; and increments with one for the next parked
call.
context => parkedcalls ; Which context parked calls are in
parkingtime => 60 ; Number of seconds a call can be parked for
; (default is 45 seconds)
;courtesytone = beep; Sound file to play to the parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
;parkedplay = caller; Who to play the courtesy tone to when picking
up
a parked call
; one of: parked, caller, both (default is
caller)
;parkedcalltransfers = caller ; Enables or disables DTMF based
transfers when picking up a parked call.
; one of: callee, caller, both, no
(default is both)
;parkedcallreparking = caller ; Enables or disables DTMF based
one-touch parking when picking up a parked call.
; one of: callee, caller, both, no
(default is no)
;parkedcallhangup = caller ; Enables or disables DTMF based hangups
when picking up a parked call.
; one of: callee, caller, both, no
(default is no)
;parkedcallrecording = caller ; Enables or disables DTMF based
one-touch recording when picking up a parked call.
; one of: callee, caller, both, no
(default is no)
;adsipark = yes ; if you want ADSI parking announcements
;findslot => next ; Continue to the 'next' free parking space.
; Defaults to 'first' available
parkedmusicclass=default; This is the MOH class to use for the parked
channel
; as long as the class is not set on the
channel directly
; using Set(CHANNEL(musicclass)=whatever) in
the dialplan
;transferdigittimeout => 3 ; Number of seconds to wait between digits
when transferring a call
; (default is 3 seconds)
;xfersound = beep ; to indicate an attended transfer is complete
;xferfailsound = beeperr; to indicate a failed transfer
pickupexten = *8; Configure the pickup extension. (default is
*8)
;featuredigittimeout = 1000 ; Max time (ms) between digits for
; feature activation (default is 1000 ms)
;atxfernoanswertimeout = 15 ; Timeout for answer on attended transfer
default is 15 seconds.
Do you need extra info ??
What setting can I have set in musiconhold.conf or features.conf to
affect the audiostream between my clients ???
Before I could call all my clients, I had musiconhold when putting 'on
hold' and I was just figuring out how parked calls worked...
Thanks for the help !
Jonas Kellens.
Re: [asterisk-users] Change Termination of Read Command
On Mon, 27 Apr 2009, Danny Nicholas wrote: > This is a "just-for-fun" question. I was reading the > support forum and a fellow there wanted Read() to stop on * instead of > #. I thought that changing app_read.c would resolve this Any chance "features" is getting in your way? 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk & EC2
I followed the Ronald Lewis instructions and was able to get EC2 to run Asterisk. I was able to use IAX2 so I'm not sure what you are saying. You should also be able to build dahdi but of course you won't have any physical devices in the machine. I think for meet-me dahdi provides a software timer. I have not tested it enough to know about issues with quality but the first reference uses a particular kernel to help avoid local timing issues with the VM. Aryan Ameri wrote: > On Sun Apr 26 2009 02:48:13 GMT+1000 (EST) Kai-Uwe Jensen > > wrote: > >> There's a boat-load of articles on the web with step-by-step guidance. >> The first I became aware of was >> http://ronaldlewis.com/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ >> >> , another good one is http://voxilla.com/2009/2/13/asterisk-amazon-ec2-1178 >> >> Google is your friend. >> > > > Thanks. Google has not been my friend, that's why I'm asking this mailing > list. > > The first guide that you link to, is by a guy who obviously doesn't know why > dahdi/zaptel are important, and completely ignores it, which means, no IAX2 > or > meet-me. > > The second one, is built on a custom Fedora 8 image. The steps are not > repeatable on any other distro, not even a stock official Fedora 8 one. > Fedora > 8 itself is long EOLed and as such, not something I'd want to use on a > production server. Dahdi compilation as described on that guide doesn't work > on CentOS or Ubuntu or Debian. > > Besides, I asked about anecdotal usage experiences running Asterisk on EC2. > About whether latency is an issue if extensions are outside the EC2 > availability zone. About reliability of EC2 when used to host a real-time > application server. Not just an installation guideline. > > It seems like no one is using Asterisk on EC2 for a production environment. > > Cheers > -- MARK. Hulber Technologies asterisk-ad...@hulber.com Read my blog : http://mark.hulber.com Follow @hulber on Twitter: http://twitter.com/hulber ___ -- 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] Record in mp3
On Friday 24 April 2009 18:35:16 Atis Lezdins wrote: > > Secondarily, MPEG audio compression takes a lot of CPU. Until the last > > few years, desktop CPUs weren't even capable of doing realtime MPEG audio > > compression, which is necessary if you're going to have the recording > > ready by the time the audio input is terminated. Above and beyond that, > > even modern CPUs are limited in how many concurrent streams can be > > MPEG-compressed, which may cause problems if you're encoding multiple > > channels to MP3 at the same time. > > Well, actually it's lot of CPU for encoding 44kHz stream. I wonder how > it would scale to encode 8kHz.. We currently do a daily routine to > compress all ulaw files to mp3 at night time, and it takes ~6 hours of > processing on 1 CPU (no parallel processing). > > Regarding legal reasons, can't it be linked with lame within > asterisk-addons? You're confusing licensing issues with patent issues. No, linking it to something else does not solve patent issues. -- Tilghman ___ -- 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] Error, Clue to what?
I've seen that message when then endpoint is not available. Cary Fitch wrote: > [Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer > '3516533812' is now UNREACHABLE! Last qualify: 86 > [Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke: > Peer '3516533812' is now Reachable. (98ms / 2000ms) > [Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > [Apr 26 12:51:20] WARNING[32281]: app_dial.c:1242 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > [Apr 26 12:52:56] WARNING[32284]: app_dial.c:1242 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > We got here to 351 land > [Apr 26 13:01:01] WARNING[32288]: app_dial.c:1242 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > [Apr 26 14:10:01] WARNING[32294]: app_dial.c:1242 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > We got here to VOICEPULSE land > We got here to VOICEPULSE land > [Apr 26 14:31:20] NOTICE[32157]: chan_iax2.c: __iax2_poke_noanswer: Peer > 'brandy' is now UNREACHABLE! Time: 85 > [Apr 26 14:31:30] NOTICE[32163]: chan_iax2.c:7967 socket_process: Peer > 'brandy' is now REACHABLE! Time: 117 > [Apr 26 15:06:07] WARNING[32300]: app_dial.c:1242 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > [Apr 26 16:18:16] WARNING[32309]: app_dial.c:1242 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > We got here to IT IS NOT NECESSARY land > We got here to VOICEPULSE land > We got here to VOICEPULSE land > We got here to VOICEPULSE land > We got here to VOICEPULSE land > We got here to VOICEPULSE land > We got here to VOICEPULSE land > [Apr 26 17:42:41] WARNING[32324]: app_dial.c:1242 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > [Apr 26 18:18:50] WARNING[32329]: app_dial.c:1242 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > > > What is the (likely) cause of the above errors? > > It happens with little channel usage at the time. I understand that the > peers were not reachable, is the dial exec full message Asterisk's message > that it couldn't communicate with those peers? > > Thanks, > > Cary Fitch > > > > > > > ___ > -- 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 > -- MARK. Hulber Technologies asterisk-ad...@hulber.com Read my blog : http://mark.hulber.com Follow @hulber on Twitter: http://twitter.com/hulber ___ -- 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] 1.6.1: menuselect has problems with x86_64 ??
I checked out the 190660 trunk and went all the way through make without a problem. Linux asterisk.hulber.com 2.6.18-128.1.6.el5 #1 SMP Tue Mar 24 12:05:57 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux -- Output through generating input for menuselect: [r...@asterisk trunk]# ./configure checking build system type... x86_64-unknown-linux-gnu checking host system type... x86_64-unknown-linux-gnu checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking how to run the C preprocessor... gcc -E checking for grep that handles long lines and -e... /bin/grep checking for egrep... /bin/grep -E checking for AIX... no checking for ANSI C header files... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking minix/config.h usability... no checking minix/config.h presence... no checking for minix/config.h... no checking whether it is safe to define __EXTENSIONS__... yes checking for uname... /bin/uname checking for gcc... (cached) gcc checking whether we are using the GNU C compiler... (cached) yes checking whether gcc accepts -g... (cached) yes checking for gcc option to accept ISO C89... (cached) none needed checking for g++... g++ checking whether we are using the GNU C++ compiler... yes checking whether g++ accepts -g... yes checking how to run the C preprocessor... gcc -E checking how to run the C++ preprocessor... g++ -E checking for a sed that does not truncate output... /bin/sed checking for egrep... grep -E checking for ld used by gcc... /usr/bin/ld checking if the linker (/usr/bin/ld) is GNU ld... yes checking for gawk... gawk checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for ranlib... ranlib checking for GNU make... make checking for strip... /usr/bin/strip checking for ar... /usr/bin/ar checking for grep... (cached) /bin/grep checking for find... /usr/bin/find checking for compress... : checking for basename... /bin/basename checking for id... /usr/bin/id checking for dirname... /usr/bin/dirname checking for sh... /bin/sh checking for ln... /bin/ln checking for dot... : checking for wget... /usr/bin/wget checking for curl... /usr/bin/curl checking for rubber... : checking for kpsewhich... : checking for xmlstarlet... : checking for soxmix... soxmix checking for md5... no checking for md5sum... md5sum checking for the pthreads library -lpthreads... no checking whether pthreads work without any flags... no checking whether pthreads work with -Kthread... no checking whether pthreads work with -kthread... no checking for the pthreads library -llthread... no checking whether pthreads work with -pthread... yes checking for joinable pthread attribute... PTHREAD_CREATE_JOINABLE checking if more special flags are required for pthreads... no checking for working alloca.h... yes checking for alloca... yes checking for dirent.h that defines DIR... yes checking for library containing opendir... none required checking for ANSI C header files... (cached) yes checking for sys/wait.h that is POSIX.1 compatible... yes checking arpa/inet.h usability... yes checking arpa/inet.h presence... yes checking for arpa/inet.h... yes checking fcntl.h usability... yes checking fcntl.h presence... yes checking for fcntl.h... yes checking for inttypes.h... (cached) yes checking libintl.h usability... yes checking libintl.h presence... yes checking for libintl.h... yes checking limits.h usability... yes checking limits.h presence... yes checking for limits.h... yes checking locale.h usability... yes checking locale.h presence... yes checking for locale.h... yes checking malloc.h usability... yes checking malloc.h presence... yes checking for malloc.h... yes checking netdb.h usability... yes checking netdb.h presence... yes checking for netdb.h... yes checking netinet/in.h usability... yes checking netinet/in.h presence... yes checking for netinet/in.h... yes checking stddef.h usability... yes checking stddef.h presence... yes checking for stddef.h... yes checking for stdint.h... (cached) yes checking for stdlib.h... (cached) yes checking for string.h... (cached) yes checking for strings.h... (cached) yes checking sys/file.h usability... yes checking sys/file.h presence... yes checking for sys/file.h... yes checking sys/ioctl.h usability... yes checking sys/ioctl.h presence... yes checking for sys/ioctl.h... yes checking sys/param.h usability... yes checking sys/param.h presence... yes checking for s
Re: [asterisk-users] 1.6.1: menuselect has problems with x86_64 ??
Without having tried it I notice the output is x86-64 and not x86_64. Could there be a typo somewhere? sean darcy wrote: > 1.6.1 svn 190575: > > CC="cc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect > CONFIGURE_SILENT="--silent" menuselect > make[1]: Entering directory > `/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect' > gcc -m64 -march=native -mtune=native -floop-interchange > -floop-strip-mine -floop-block -c -o menuselect_stub.o menuselect_stub.c > gcc -o menuselect menuselect.o strcompat.o menuselect_stub.o mxml/libmxml.a > /usr/bin/ld: i386 architecture of input file `menuselect.o' is > incompatible with i386:x86-64 output > /usr/bin/ld: i386 architecture of input file `strcompat.o' is > incompatible with i386:x86-64 output > > sean > > > ___ > -- 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 > -- MARK. Hulber Technologies asterisk-ad...@hulber.com Read my blog : http://mark.hulber.com Follow @hulber on Twitter: http://twitter.com/hulber ___ -- 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] music on hold using mms
Didn't do mms but have implemented using Shoutcast. I have instructions at the link below: http://mark.hulber.com/voip/configuration/shoutcast-musiconhold-in-asterisk-16/ Rilawich Ango wrote: > Hi, > I follow the > web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf > - mohstream.sh , to configure music on hold to play using mms but > failed. Anyone can play using mms? > ango > > ___ > -- 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 > -- MARK. Hulber Technologies asterisk-ad...@hulber.com Read my blog : http://mark.hulber.com Follow @hulber on Twitter: http://twitter.com/hulber ___ -- 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
[asterisk-users] SIP infrastructure
O boy. SIP infrastructure is so flexible that basically nobody gets it right. :-) You could easily have 20 different SIP network elements (/servers /services). Even more. And we get at least 5 new SIP-RFCs per day. They're all trying to fix things which the previous specifications didn't address. :-) Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Change Termination of Read Command
On Apr 27, 2009, at 10:29 AM, Danny Nicholas wrote:
Greetings all,
This is a “just-for-fun” question. I was
reading the support forum and a fellow there wanted Read() to stop
on * instead of #. I thought that changing app_read.c would resolve
this
current
if (tmp[x-1] == '#') {
tmp[x-1] = '\0';
break;
new
}if (tmp[x-1] == '*') {
tmp[x-1] = '\0';
break;
}
He applied and recompiled, but no joy. Any ideas why?
Without knowing where in the file this came from I can't say for sure,
but that code looks to me like the code that would run after the
digits are received and is stripping off the # character at the end,
if it is there. Further up (or somewhere else entirely) there is
probably a spot that actually terminates the read command when # is
pressed.
Daniel
Danny Nicholas
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[asterisk-users] Change Termination of Read Command
Greetings all,
This is a "just-for-fun" question. I was reading the
support forum and a fellow there wanted Read() to stop on * instead of #. I
thought that changing app_read.c would resolve this
current
if (tmp[x-1] == '#') {
tmp[x-1] = '\0';
break;
new
}if (tmp[x-1] == '*') {
tmp[x-1] = '\0';
break;
}
He applied and recompiled, but no joy. Any ideas why?
Danny Nicholas
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Re: [asterisk-users] No format for saving voicemail?
cbbs...@hotmail.com schrieb: > All; > I just came accross this problem, and I am a bit stumped. I am using > Asterisk 1.4.23.1 and am using Asterisk Realtime Static for voicemail. I have > not had a problem before, but now when someone tries to leave a vm, I get the > error "No format for saving voicemail?" and Asterisk hangs up the call. > According to the docs, Asterisk should default to wav49|gsm|wav. Clearly it > is not defaulting at all. I added the MySQL column "format", but still no > joy. As a temporary fix, I moved back to voicemail.conf and added "format" > there. Has anyone seen this before? Any insight at all would be greatly > appreciated. Are you really using static Realtime (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static) or real Realtime? In static Realtime you don't need a column for the format parameter (or any other parameter). In real Realtime you do. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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
[asterisk-users] No format for saving voicemail?
All; I just came accross this problem, and I am a bit stumped. I am using Asterisk 1.4.23.1 and am using Asterisk Realtime Static for voicemail. I have not had a problem before, but now when someone tries to leave a vm, I get the error "No format for saving voicemail?" and Asterisk hangs up the call. According to the docs, Asterisk should default to wav49|gsm|wav. Clearly it is not defaulting at all. I added the MySQL column "format", but still no joy. As a temporary fix, I moved back to voicemail.conf and added "format" there. Has anyone seen this before? Any insight at all would be greatly appreciated. Thanks _ Windows Live™ Hotmail®:…more than just e-mail. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_more_042009___ -- 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] Digium fax force T38?
On Mon, 27 Apr 2009 00:33:44 +1200 Michael wrote: > I can't with Digium fax, and it always fails at the point it decides > to switch to T38. Have you tried dedicating the line to fax only, no "detection"? I tried using it, but for me it apparently fails the codec switch: WARNING[3862]: frame.c:214 __ast_smoother_feed: Smoother was working on 4 format frames, now trying to feed 64? [Apr 14 21:50:46] ERROR[3862]: res_fax.c:910 generic_fax_exec: channel 'DAHDI/1-1' fax session '3' failure, reason: 'Failed to feed the smoother' But if I just initiate the fax on that channel for whatever comes, it works. http://forums.digium.com/viewtopic.php?p=128681#128681 Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- 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] Digium fax force T38?
Michael wrote: > Is it possible to force T38 for all invocations ReceiveFAX() ? It already does that. > I can't with Digium fax, and it always fails at the point it decides to > switch > to T38. You've posted two or three messages about this, but haven't included any information we could use to help you. At a minimum, we need the exact versions of Asterisk and FFA you tested with, and a complete console log (including verbosity >= 10, debug >= 10 and 'sip set debug on') to see what is happening. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- 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] Can't dial out until I dial in once
On Sat, 25 Apr 2009 00:01:44 -0400 Michael van der Stoop wrote: > I call in once from a cell phone, which is > successful then I can call out with out issue. It's a bug. Maybe this one? http://bugs.digium.com/print_bug_page.php?bug_id=14577 Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- 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
[asterisk-users] Diference between volume of mp3 and wav files
Hi, I have some files in mp3 in my Asterisk but when I play it the volume is lo= wer than wav files. Both the files (wav and mp3) are encoded with the same = amplitude. In anothers players the audio volume of these files are equal. Can I fix this diference between volume of mp3 and wav file? Thanks Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.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
[asterisk-users] Going to AMOOCON?
Hi, If you are going to AMOOCON through Berlin Sunday evening and could use a ride to Rostock, please feel free to email me. If you are are going to be there I look forward to meeting you. I will be leaving early Wednesday morning for Berlin as well. Reserve now and avoid the rush :) ___ -- 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] Outgoing Queues
Ok, and thats exactly what I mean monitoring outbound groups, so you can have realtime info for monitoring. And as with queues have the ability to reset the statics for monitoring porpouses. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Atis Lez Sent: lunes, 27 de abril de 2009 08:25 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Outgoing Queues > Shouldnt the member has the statics per queue? > > I mean, I have 2 queues test1 and test2, with member 1001 for example for > both queues, if I make a call to queue test1 and the member 1001 answers the > call, the statics for the member is up in both queues, (has taken 1 call .), > this should be per queue basis dont you think? > Yes it's so, unless You have enabled shared_lastcall, in which case lastcall and call counter is shared across queues in order to acquire fair call distribution strategy. You shouldn't use queue data for statistics, there's queue_log for that. This is purely monitoring info which can get lost during restarts/reloads. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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 Se certificó que el correo entrante no contiene virus. Comprobada por AVG - www.avg.es Versión: 8.5.287 / Base de datos de virus: 270.12.4/2081 - Fecha de la versión: 04/26/09 09:44:00 ___ -- 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] AMD Not Working
On 27/04/2009 4:22 p.m., Sam Hawkin wrote:
> Hi,
>
> Thanks for your reply.
>
> I have tried as you suggested, I does not even come upto NoOp()
> It hangups after AMD.
> I have decreased the silence threshold from 256 to 100 and 50.
Try the NoOp in the h extension:
exten => h,1,NoOp(Status: ${AMDSTATUS} Cause: ${AMDCAUSE})
--
Kind Regards,
Matt Riddell
Director
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Re: [asterisk-users] Outgoing Queues
> Shouldn’t the member has the statics per queue? > > I mean, I have 2 queues test1 and test2, with member 1001 for example for > both queues, if I make a call to queue test1 and the member 1001 answers the > call, the statics for the member is up in both queues, (has taken 1 call….), > this should be per queue basis don’t you think? > Yes it's so, unless You have enabled shared_lastcall, in which case lastcall and call counter is shared across queues in order to acquire fair call distribution strategy. You shouldn't use queue data for statistics, there's queue_log for that. This is purely monitoring info which can get lost during restarts/reloads. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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
[asterisk-users] music on hold using mms
Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango ___ -- 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] Outgoing Queues
We use something like that in QueueMetrics to track outgoing calls for call-centers: http://forum.queuemetrics.com/index.php?topic=261.0 thanks l. 2009/4/25 Sebastian > Anyone thought about something like outgoing queues? > > I mean, having same info that has for inbound queues but for outbound > calls, and grouping members there. > > For example, before using dial application put an app outqueue that get all > the statics. > > Talked time, member status, last call, completed calls, failed calls, reset > statics, and maybe some more. So its possible to get more control and has > more data for example via AMI. > > > > Another comment about queues. > > Shouldn’t the member has the statics per queue? > > I mean, I have 2 queues test1 and test2, with member 1001 for example for > both queues, if I make a call to queue test1 and the member 1001 answers the > call, the statics for the member is up in both queues, (has taken 1 call….), > this should be per queue basis don’t you think? > > > > > > Thanks! > > > > ___ > -- 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 > -- Loway - 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
