Re: [asterisk-users] can we install 10 PCI card on asterisk

[Raj Mathur (राज माथुर)]

On Tuesday 28 Aug 2012, DHAVAL INDRODIYA wrote:
> Thanks for everyone input on this, this was just mine thoughts to put
> 80 PRI line in that.but after reading inputs from everyone i think
> there are some options to achieve it.
> 
> it means i need to put a gateway which convert my SIP calls to PRI
> line and another options is to put
> multiple asterisk boxes and each box have maximum 16 pri lines . now
> which is best choice to work on further. also i need to consider
> hardware sizing too as if gateway is expensive i would go with pri
> cards.
> also if i choose gateway then  also i need to put multiple asterisk
> boxes.

FWIW...

Our largest setup consists of some 2000 SIP users distributed over 2 
boxes (sip concentrators).  The PSTN interface is through another set of 
boxes with up to 24 PRIs per box (dial banks).

Users log into one of the sip concentrators with soft (Qutecom) or hard 
SIP phones.  When they place a call, it's automatically distributed to a 
PRI on one of the dial banks.  The PRI selection is weighted random, 
with individual preference sets being assigned for each group of 
callers.

The biggest issue we faced was figuring out that you can't have more 
than one PSTN provider on a single dial bank -- the timing sources 
interfere with each other and cause call drops.  The current setup 
connects all the PRIs of a single telco to a single dial bank, 
eliminating that problem.  There are currently 3 telcos providing PRIs 
in the largest centre.

The client and we are happily running vanilla Asterisk Debian packages 
with (even though I say so myself) some scripting to die for.  Setup is 
completely stable and is being used to generate some $15M of business 
annually for the client.

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

--
_
-- 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] can we install 10 PCI card on asterisk

[DHAVAL INDRODIYA]

Hey All,

Thanks for everyone input on this, this was just mine thoughts to put 80
PRI line in that.but after reading inputs from everyone i think there are
some options to achieve it.

it means i need to put a gateway which convert my SIP calls to PRI line and
another options is to put
multiple asterisk boxes and each box have maximum 16 pri lines . now which
is best choice to work on further. also i need to consider hardware sizing
too as if gateway is expensive i would go with pri cards.
also if i choose gateway then  also i need to put multiple asterisk boxes.

let me know your thoughts.

thanks
Dhaval

On Mon, Aug 27, 2012 at 10:54 PM, Eric Wieling  wrote:

> Your best bet is a carrier class device from someone like Adtran and
> convert the PRIs to SIP before passing the calls to Asterisk.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
> Sent: Monday, August 27, 2012 8:09 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] can we install 10 PCI card on asterisk
>
> Hi All,
>
> i would like to know if anyone has done or having idea regarding PRI
> terminations in asterisk.
>
> i have a requirement where i need to support 80 PRI in one machine i have
> found a machine which have 10 PCI slots available
>
> now i am thinking of arranging 8port sangoma card in this pci slots so i
> can arrenge 10 card in that.
>
> is it possible to run system like that ? is it good idea , can asterisk
> handle 2400 calls if machine size and RAM is good.
>
> let me know ideas and suggestions.
>
> thanks
> Dhaval
>
>
> --
> _
> -- 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 hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi danny,

Are you talking about modules or sip extensions and dahdi extensions
because its a fresh installation also it doesn't have dahdi interface, I am
just trying to have only ip side.

Regards
On Aug 27, 2012 7:27 PM, "Danny Nicholas"  wrote:

> I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and
> 10 SP2).  My advice would be to try to start the box with as few SIP/DAHDI
> channels as possible to begin with and add as you get things stable.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
> *Sent:* Monday, August 27, 2012 8:52 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
> 12.2
>
> ** **
>
> Hi Patrick,
>
> ** **
>
> With other OS it works like charm. Only with OpenSuse, I am facing this
> issue, since I have a situation to stick with OpenSuse, I am struggling in
> this. 
>
> ** **
>
> Regards. 
>
> On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists <
> asterisk-l...@puzzled.xs4all.nl> wrote:
>
> On 27-08-12 08:25, Gopalakrishnan N wrote:
>
> This is really tuff working with OpenSuse. I am clueless how to sort our
> this.
>
> ** **
>
> Maybe switch to a different distribution? I have used CentOS and RHEL for
> years without any problems and as far as I know both debian and ubuntu
> should work without problems too.
>
> Regards,
> Patrick
>
>
>
>
>
> --
> _
> -- 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
>
--
_
-- 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 community services - Old Mantis instance to be shutdown on Aug 28th, 2012

[Vladimir Mikhelson]

Guys,

Is it possible to leave the Mantis on permanently?

It allows to productively search and work with issues recorded in it. 
Search, convenient straight forward layout, patch download URLs,
everything just works there.

JIRA maybe is convenient for the management and developers.  I just
guess, as somebody must have loved it so it was chosen as a Mantis
replacement.  But for an ordinary user (in my opinion) it is cumbersome
and unfriendly.

Thank you,
Vladimir





On 8/27/2012 9:08 AM, Asterisk Development Team wrote:
> On June 5, 2011, we migrated from Mantis to Jira as the issue tracker
> for Asterisk [1]. We temporarily left Mantis running in read-only mode
> to smooth the transition. At 15 months, temporary has turned into
> semi-permanent. As a part of other infrastructure changes we are making
> to the community services, we will finally shut down Mantis for good.
>
> We will update our DNS servers on the morning of Tuesday, August 28th,
> however it may take a few hours for those changes to propagate.
>
> We have done our best to put redirects in place so that old links to
> Mantis will still work. If you find a link that does not redirect as
> expected, or have any other problems you think may be caused by the
> Mantis shutdown, please report them in the "JIRA Help" project [2]. If
> you would rather report your issue via email, you may contact us at
> asteriskt...@digium.com.
>
> [1]:
> http://lists.digium.com/pipermail/asterisk-announce/2011-June/000324.html
> [2]: https://issues.asterisk.org/jira/browse/JA
>
> Digium's Asterisk Development Team
>
> -- 
> _
> -- 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] One leg in a conference and adjusting stream volume of other leg

[Matthew Jordan]

- Original Message -
> From: "Markus" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Cc: "Matthew Jordan" 
> Sent: Monday, August 27, 2012 1:55:08 PM
> Subject: Re: [asterisk-users] One leg in a conference and adjusting stream 
> volume of other leg
> 
> Hi Matthew,
> 
> Am 27.08.2012 20:08, schrieb Matthew Jordan:
> >>> You can use ConfBridge's DTMF menus to allow a participant to
> >>> change
> >>> their listening volume.  This should only affect the audio that
> >>> the
> >>> participant hears, and not the other participants in the
> >>> conference
> >>> -
> >>> regardless of whether or not the audio originates from the same
> >>> source.
> >>
> >> thanks! I wasn't clear enough in my original mail. What I meant
> >> is:
> >> the
> >> volume of the stream that a user is listening to is adjusted, but
> >> the
> >> volume of the conference itself is not changed! That means, a
> >> conference
> >> is going on, and everyone is listening to the same music at the
> >> same
> >> time, but when the music becomes too loud or annoying, a user can
> >> individually adjust the volume of his music, while the volume of
> >> the
> >> speech of each user, basically the conference itself, remains the
> >> same.
> >
> > Yes, I know.  That's what the DTMF menus in ConfBridge let you do.
> 
> thanks again. If I understand correctly, you are saying that there is
> a
> switch that allows a user to adjust the volume of the "background"
> music
> only, but the incoming speech that is coming in to him from other
> users
> will not get adjusted? That's awesome, but I can't find anything like
> that in the docs.

No - what you stated was "the volume of the stream that a user is
listening to is adjusted, but the volume of the conference is not changed!"

I interpreted that as being the volume of the audio sent to the conference
participant.  That can be manipulated directly in ConfBridge.  However,
that affects all audio sent to that participant, which isn't apparently
what you want.

ConfBridge works by mixing the audio for all channels in the conference
and playing the resulting audio to each participant.  You can affect
each participant, but you can't change that all of the audio is mixed
together first.  If you want to play audio separately to each participant,
than you have to do something outside of the actual conference bridge itself.

> Will your example
> 
> [bridge_user_menu]
> *1=increase_listening_volume
> 1=increase_listening_volume
> *2=decrease_listening_volume
> 2=decrease_listening_volume
> 
> not just decrease/increase the audio of *everything* that is coming
> in
> to the user, i.e. both music and speech? At least that it's how it's
> explained in the documentation, isn't it?

Yes.
 
> "Decreases the caller's listening volume. *Everything* they hear will
> sound quieter."
> 
> What I'm looking for is to adjust the incoming music only, not the
> incoming speech. How is ConfBridge able to separate between these two
> if
> they are going on at the same time?

It doesn't; they are mixed together.

> Done that couple of times, but I still don't see that "feature".
> 
> I think there is still some sort of misunderstanding here. Maybe I'm
> not
> explaining it right...

Yup, that was a misunderstanding.

You could probably use ChanSpy to whisper the music to each individual
participant.  Something like this:

[conference]

exten => s,1,NoOp()
same => n,Set(GLOBAL(CONF_CHANNEL_NAME=${CHANNEL}))
same => n,Originate(Local/start_music@conference,exten,conference,moh,1)
same => n,ConfBridge(1)

exten => moh,1,NoOp()
same => n,MusicOnHold()

exten => start_music,1,NoOp()
same => n,Answer()
same => n,ChanSpy(${CONF_CHANNEL_NAME},w)
same => n,Hangup()

You may not want to use something more elegant than a global variable to
cache the name of the channel going into the conference or at least provide
some synchronization around it so that two channels entering the conference
don't step on each other, but this should point you in the correct direction.

As Johan mentioned, the trick to manipulating the volume on the Local channel
streaming the music is best handled externally through AMI.  You can use
the Redirect AMI command to manipulate the channel into other dialplan
extensions that change the volume, then Redirect them back into the moh
extension.  You could trigger that by using dialplan_exec menu actions
from the ConfBridge participant, and raise UserEvents that signal what
action the user wants to take.




--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  

Re: [asterisk-users] the lenght of the uri affects on dialplan?

[Matthew J. Roth]

Rafael Visser wrote:

> I replaced for the following sip.conf
>
> [general] 
> context=default ; Default context for incoming calls 
> allowguest=no ; Allow or reject guest calls -sin password- (default is yes) 
> allowoverlap=no ; Disable overlap dialing support. (Default is yes) 
> udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds 
> to all) 
> tcpenable=yes ; Enable server for incoming TCP connections (default is no) 
> tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to 
> all interfaces) 
> srvlookup=yes ; Enable DNS SRV lookups on outbound calls 
> relaxdtmf=yes 
> dtmfmode=inband 
> ;rfc2833compensate=yes 
>
> [sip.ericsson] 
> ;cambios allowguest hosts 
> allowguest=no ; Allow or reject guest calls -sin password- (default is yes) 
> type=friend 
> calllimit=200 
> fromuser=ivr1 
> dtmfmode=inband 
> username=administrador 
> context=incoming-sip-ericsson 
> host=10.146.9.70 
> host=ericsson 
> host=MSSASU1.MYDOMAIN.COM.PY 
> port=5060 
> disallow=all 
> allow=alaw 
> allow=gsm 
> allow=ulaw 
> qualify=yes 
> insecure=no
>
>
> Debug with long hostname (B is considered as an '*') 
>  
> <--- SIP read from TCP:10.146.9.70:6240 ---> 
> ...
> No matching peer for '971200152' from '10.146.9.70:6240' 
>
> Short hostname on switch 
> === 
> <--- SIP read from UDP:10.146.9.70:5060 ---> 
> ...
> Found peer 'sip.ericsson' for '971200152' from 10.146.9.70:5060 


The device sending SIP is using a different protocol (TCP vs. UDP) and
port (6240 vs. 5060) in your two examples.  You have Asterisk
configured to listen for UDP and TCP connections, so the protocol
isn't the problem.

However, the first example fails to find a matching peer because the
"sip.ericsson" SIP entity is defined with "port=5060" and
"insecure=no".  Try changing the insecure option to "insecure=port".
This should resolve your problem by allowing the peer to be matched
by IP address regardless of the port number.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

--
_
-- 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] One leg in a conference and adjusting stream volume of other leg

[Markus]

Hi Matthew,

Am 27.08.2012 20:08, schrieb Matthew Jordan:

You can use ConfBridge's DTMF menus to allow a participant to
change
their listening volume.  This should only affect the audio that the
participant hears, and not the other participants in the conference
-
regardless of whether or not the audio originates from the same
source.


thanks! I wasn't clear enough in my original mail. What I meant is:
the
volume of the stream that a user is listening to is adjusted, but the
volume of the conference itself is not changed! That means, a
conference
is going on, and everyone is listening to the same music at the same
time, but when the music becomes too loud or annoying, a user can
individually adjust the volume of his music, while the volume of the
speech of each user, basically the conference itself, remains the
same.


Yes, I know.  That's what the DTMF menus in ConfBridge let you do.


thanks again. If I understand correctly, you are saying that there is a 
switch that allows a user to adjust the volume of the "background" music 
only, but the incoming speech that is coming in to him from other users 
will not get adjusted? That's awesome, but I can't find anything like 
that in the docs.


Will your example

[bridge_user_menu]
*1=increase_listening_volume
1=increase_listening_volume
*2=decrease_listening_volume
2=decrease_listening_volume

not just decrease/increase the audio of *everything* that is coming in 
to the user, i.e. both music and speech? At least that it's how it's 
explained in the documentation, isn't it?


"Decreases the caller's listening volume. *Everything* they hear will 
sound quieter."


What I'm looking for is to adjust the incoming music only, not the 
incoming speech. How is ConfBridge able to separate between these two if 
they are going on at the same time?




Again, that is what the menus in ConfBridge do.  Please read the ConfBridge
documentation on the Asterisk wiki.

https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10


Done that couple of times, but I still don't see that "feature".

I think there is still some sort of misunderstanding here. Maybe I'm not 
explaining it right...


Thanks!
Markus


--
_
-- 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] One leg in a conference and adjusting stream volume of other leg

[Johan Wilfer]

2012-08-27 19:48, Markus skrev:

Hi Matthew,

Am 27.08.2012 15:41, schrieb Matthew Jordan:

When they adjust the volume of the stream, if effects only their
stream,
and not the volume of the stream of the other callers.

In short: All callers at all times are *always* in the same
conference,
but each caller is able to increase or decrease the volume of "their"
MP3 stream individually.


You can use ConfBridge's DTMF menus to allow a participant to change
their listening volume.  This should only affect the audio that the
participant hears, and not the other participants in the conference -
regardless of whether or not the audio originates from the same source.


thanks! I wasn't clear enough in my original mail. What I meant is: the
volume of the stream that a user is listening to is adjusted, but the
volume of the conference itself is not changed! That means, a conference
is going on, and everyone is listening to the same music at the same
time, but when the music becomes too loud or annoying, a user can
individually adjust the volume of his music, while the volume of the
speech of each user, basically the conference itself, remains the same.

I think what I'm looking for is to inject the MP3 stream into only the
"listening" direction of each user, and allow its volume to get adjusted
via DTMF. And at the same time, each user is in the same conference.

Even more: I would like to be able to feed each user a *different*
volume-adjustable MP3 stream, but all of the users are still in the same
conference (not hearing each others MP3 stream, only their voice!).

I've researched high and low and came up with the following pointers:

- Dial with the G flag
- ChanSpy, whispering
- VOLUME()
- MOH connected to a local channel
- Queue that loops indefinitely

But I don't know yet how to put it all together.

I found some hints in the right direction here:

"Playing audio to one channel only":
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg245811.html

"Meetme with background music" (last post)
http://fonality.com/trixbox/forums/trixbox-forums/help/meetme-background-music


"Background music during a call"
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg254252.html


Does anyone have the right solution and is available to create a
dialplan for me for cash? Please get in touch!



I would do it like this:

1. Use Meetme or Confbridge and use functionality to jump out of the 
conference if DTMF is pressed (X-flag in meetme, I expect similar exists 
in confbridge).


2. Call AGI, Log to DB, etc - whatever - and return to the conference.

3. Have a external program that manipulates the channel playing the 
music. For example this could be done by ChannelRedirect AMI to special 
dialplan extensions that lower and raises the volum. You can use 
System()-app in asterisk, or AGI for example. Then use AMI in the script.



The music on hold could be implemented as a Local channel.

a. Look at Originate-app, or Originate AMI command. One side of the call 
are connected to a context/extension/priority (for example: Meetme 
here). And the other end you dial Local/extension@context (for example: 
Here you play music).


b. Prepare some extensions that lower/raises volume (look at func_volume)

Good luck!

/Johan

--
_
-- 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] One leg in a conference and adjusting stream volume of other leg

[Matthew Jordan]

- Original Message -
> From: "Markus" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Cc: "Matthew Jordan" 
> Sent: Monday, August 27, 2012 12:48:53 PM
> Subject: Re: [asterisk-users] One leg in a conference and adjusting stream 
> volume of other leg
> 
> > You can use ConfBridge's DTMF menus to allow a participant to
> > change
> > their listening volume.  This should only affect the audio that the
> > participant hears, and not the other participants in the conference
> > -
> > regardless of whether or not the audio originates from the same
> > source.
> 
> thanks! I wasn't clear enough in my original mail. What I meant is:
> the
> volume of the stream that a user is listening to is adjusted, but the
> volume of the conference itself is not changed! That means, a
> conference
> is going on, and everyone is listening to the same music at the same
> time, but when the music becomes too loud or annoying, a user can
> individually adjust the volume of his music, while the volume of the
> speech of each user, basically the conference itself, remains the
> same.

Yes, I know.  That's what the DTMF menus in ConfBridge let you do.
 
> I think what I'm looking for is to inject the MP3 stream into only
> the
> "listening" direction of each user, and allow its volume to get
> adjusted
> via DTMF. And at the same time, each user is in the same conference.

Again, that is what the menus in ConfBridge do.  Please read the ConfBridge
documentation on the Asterisk wiki.

https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10

> Even more: I would like to be able to feed each user a *different*
> volume-adjustable MP3 stream, but all of the users are still in the
> same
> conference (not hearing each others MP3 stream, only their voice!).

Feeding different MP3 streams from outside sources to different participants
is not possible in ConfBridge, as the bridging layer mixes the audio from the
various channels and sends it to all participants.  Manipulating the volume for
each participant is possible.

While feeding different MP3 streams to different users is not feasible,
you can stream different MOH classes to different users by setting the
music_on_hold_class user profile option for different ConfBridge users.

You can also stream audio to a single channel using the playback menu option,
but that's for relatively short messages, and not for "MOH" type situations.

> I've researched high and low and came up with the following pointers:
> 
> - Dial with the G flag

I'm not sure why transferring to a particular dialplan context would be
needed here.  If you wanted to bounce out to the dialplan while in a
ConfBridge, that's possible using the dialplan_exec menu option.

> - ChanSpy, whispering

You could always bounce out to the dialplan and execute a ChanSpy on
the channel in the bridge, but that feels overly complex for what
you're attempting to do.

> - VOLUME()

This does the same thing under the hood that the ConfBridge listening/talking
volume menu options do.

> - MOH connected to a local channel

This would stream some music into the Bridge to all participants.  You may
want this to stream some MP3 file into the conference; on the other hand,
it doesn't allow for the same granularity that the music_on_hold_class
bridge profile option provides.

> - Queue that loops indefinitely

This just scares me.

> 
> But I don't know yet how to put it all together.
>
> I found some hints in the right direction here:
> 
> "Playing audio to one channel only":
> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg245811.html
>
> "Meetme with background music" (last post)
> http://fonality.com/trixbox/forums/trixbox-forums/help/meetme-background-music
> 
> "Background music during a call"
> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg254252.html
> 

Those are all MeetMe specific.  You can certainly use MeetMe in Asterisk 10 and
put together a solution using what those forum/e-mails suggest; or you can
use the tools provided in ConfBridge.

> Does anyone have the right solution and is available to create a
> dialplan for me for cash? Please get in touch!
> 

If you don't want to do the work yourself, contact the asterisk-biz list.

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One leg in a conference and adjusting stream volume of other leg

[Markus]

Hi Matthew,

Am 27.08.2012 15:41, schrieb Matthew Jordan:

When they adjust the volume of the stream, if effects only their
stream,
and not the volume of the stream of the other callers.

In short: All callers at all times are *always* in the same
conference,
but each caller is able to increase or decrease the volume of "their"
MP3 stream individually.


You can use ConfBridge's DTMF menus to allow a participant to change
their listening volume.  This should only affect the audio that the
participant hears, and not the other participants in the conference -
regardless of whether or not the audio originates from the same source.


thanks! I wasn't clear enough in my original mail. What I meant is: the 
volume of the stream that a user is listening to is adjusted, but the 
volume of the conference itself is not changed! That means, a conference 
is going on, and everyone is listening to the same music at the same 
time, but when the music becomes too loud or annoying, a user can 
individually adjust the volume of his music, while the volume of the 
speech of each user, basically the conference itself, remains the same.


I think what I'm looking for is to inject the MP3 stream into only the 
"listening" direction of each user, and allow its volume to get adjusted 
via DTMF. And at the same time, each user is in the same conference.


Even more: I would like to be able to feed each user a *different* 
volume-adjustable MP3 stream, but all of the users are still in the same 
conference (not hearing each others MP3 stream, only their voice!).


I've researched high and low and came up with the following pointers:

- Dial with the G flag
- ChanSpy, whispering
- VOLUME()
- MOH connected to a local channel
- Queue that loops indefinitely

But I don't know yet how to put it all together.

I found some hints in the right direction here:

"Playing audio to one channel only":
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg245811.html

"Meetme with background music" (last post)
http://fonality.com/trixbox/forums/trixbox-forums/help/meetme-background-music

"Background music during a call"
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg254252.html


Does anyone have the right solution and is available to create a 
dialplan for me for cash? Please get in touch!



Thank you!
Markus





--
_
-- 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] can we install 10 PCI card on asterisk

[Eric Wieling]

Your best bet is a carrier class device from someone like Adtran and convert 
the PRIs to SIP before passing the calls to Asterisk. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
Sent: Monday, August 27, 2012 8:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] can we install 10 PCI card on asterisk

Hi All,

i would like to know if anyone has done or having idea regarding PRI 
terminations in asterisk.

i have a requirement where i need to support 80 PRI in one machine i have found 
a machine which have 10 PCI slots available 

now i am thinking of arranging 8port sangoma card in this pci slots so i can 
arrenge 10 card in that.

is it possible to run system like that ? is it good idea , can asterisk handle 
2400 calls if machine size and RAM is good.

let me know ideas and suggestions.

thanks
Dhaval  


--
_
-- 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] Getting hold status via AMI ?

[Brian Camp]

Hi,

Is there any way to tell via the AMI or console if a given SIP channel is
hold?   ChanIsAvail in the dialplan appears to have a 'hold' status, but
AMI and CLI commands tend to return 'in use', which is the same state as a
regular active call.

Thanks

-Brian
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 1.8.15 distintive ringtone for internal calls

[Phil Frost]

On 08/27/2012 01:02 PM, motty.cruz wrote:

Hello, would like to have distintive ringtone for internal calls, google
gave me blurr answer.

My extensions are 46**, any calls made within 46** I want to ring
differently than external calls.


Assuming you are using SIP handsets, distinctive ring isn't directly an 
Asterisk feature. Rather, you must consult the documentation for your 
endpoints, and determine how to signal them to use a different ringer. 
Usually it involves a SIP header. For an example, I use snom 870 
handets, and I have this in my dialplan:


SIPAddHeader(Alert-Info: 
;info=sales;x-line-id=0)


and I've also configured the handsets to associate "sales" with "Ringer 3".


--
_
-- 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 1.8.15 distintive ringtone for internal calls

[motty.cruz]

Hello, would like to have distintive ringtone for internal calls, google
gave me blurr answer. 

My extensions are 46**, any calls made within 46** I want to ring
differently than external calls. 

Thanks in advance. 



--
_
-- 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 community services - Old Mantis instance to be shutdown on Aug 28th, 2012

[Leif Madsen]

On 27/08/12 10:08 AM, Asterisk Development Team wrote:

As a part of other infrastructure changes we are making
to the community services, we will finally shut down Mantis for good.


Huzzuh!

Does this mean http://issues.asterisk.org will now go directly to JIRA?

Leif.

--
Leif Madsen
http://www.oreilly.com/catalog/asterisk

--
_
-- 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 community services - Old Mantis instance to be shutdown on Aug 28th, 2012

[Asterisk Development Team]

On June 5, 2011, we migrated from Mantis to Jira as the issue tracker
for Asterisk [1]. We temporarily left Mantis running in read-only mode
to smooth the transition. At 15 months, temporary has turned into
semi-permanent. As a part of other infrastructure changes we are making
to the community services, we will finally shut down Mantis for good.

We will update our DNS servers on the morning of Tuesday, August 28th,
however it may take a few hours for those changes to propagate.

We have done our best to put redirects in place so that old links to
Mantis will still work. If you find a link that does not redirect as
expected, or have any other problems you think may be caused by the
Mantis shutdown, please report them in the "JIRA Help" project [2]. If
you would rather report your issue via email, you may contact us at
asteriskt...@digium.com.

[1]: 
http://lists.digium.com/pipermail/asterisk-announce/2011-June/000324.html

[2]: https://issues.asterisk.org/jira/browse/JA

Digium's Asterisk Development Team

--
_
-- 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 hangs while starting in OpenSuse 12.2

[Danny Nicholas]

I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and 10
SP2).  My advice would be to try to start the box with as few SIP/DAHDI
channels as possible to begin with and add as you get things stable.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan
N
Sent: Monday, August 27, 2012 8:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

 

Hi Patrick,

 

With other OS it works like charm. Only with OpenSuse, I am facing this
issue, since I have a situation to stick with OpenSuse, I am struggling in
this. 

 

Regards. 

On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists
 wrote:

On 27-08-12 08:25, Gopalakrishnan N wrote:

This is really tuff working with OpenSuse. I am clueless how to sort our
this.

 

Maybe switch to a different distribution? I have used CentOS and RHEL for
years without any problems and as far as I know both debian and ubuntu
should work without problems too.

Regards,
Patrick





--
_
-- 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 hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi Patrick,

With other OS it works like charm. Only with OpenSuse, I am facing this
issue, since I have a situation to stick with OpenSuse, I am struggling in
this.

Regards.

On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 27-08-12 08:25, Gopalakrishnan N wrote:
>
>> This is really tuff working with OpenSuse. I am clueless how to sort our
>> this.
>>
>
> Maybe switch to a different distribution? I have used CentOS and RHEL for
> years without any problems and as far as I know both debian and ubuntu
> should work without problems too.
>
> Regards,
> Patrick
>
>
>
>
> --
> __**__**_
> -- 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] One leg in a conference and adjusting stream volume of other leg

[Matthew Jordan]

- Original Message -
> From: "Markus" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Sunday, August 26, 2012 6:43:31 PM
> Subject: [asterisk-users] One leg in a conference and adjusting stream
> volume of other leg
> 
> Hi all,
> 



> A SIP caller dials into to my Asterisk 10. He will automatically
> listen
> to a specific MP3 stream.

As you're using Asterisk 10, I'm going to assume you're using ConfBridge.



> When they adjust the volume of the stream, if effects only their
> stream,
> and not the volume of the stream of the other callers.
>
> In short: All callers at all times are *always* in the same
> conference,
> but each caller is able to increase or decrease the volume of "their"
> MP3 stream individually.

You can use ConfBridge's DTMF menus to allow a participant to change
their listening volume.  This should only affect the audio that the
participant hears, and not the other participants in the conference -
regardless of whether or not the audio originates from the same source.

[bridge_user_menu]
*1=increase_listening_volume
1=increase_listening_volume
*2=decrease_listening_volume
2=decrease_listening_volume

For more information on setting up DTMF menus and associating them with
bridge users, see the ConfBridge article on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10

> If I'm right the MP3 stream cannot come from inside conference
> (MeetMe
> or ConfBridge with MOH) because there is no functionality to control
> the
> volume individually. 



No, that's fine.  ConfBridge allows you to control the listening/speaking
volume of each participant.  See above.



> 
> I don't know where to start. Queue? Local channel? ...
> 

If you wanted to stream your sound from a source other than MOH, using a
Local channel may be appropriate.  I'm not sure how a Queue would help here.

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] can we install 10 PCI card on asterisk

[Shaun Ruffell]

On Mon, Aug 27, 2012 at 06:10:23PM +0530, Raj Mathur (राज माथुर) wrote:
> On Monday 27 Aug 2012, DHAVAL INDRODIYA wrote:
> > i would like to know if anyone has done or having idea regarding PRI
> > terminations in asterisk.
> > 
> > i have a requirement where i need to support 80 PRI in one machine i
> > have found a machine which have 10 PCI slots available
> > 
> > now i am thinking of arranging 8port sangoma card in this pci slots
> > so i can arrenge 10 card in that.
> 
> Last I checked, the highest channel number DAHDI supported was 1023, 
> limiting you to some 34 E1 PRIs.

Since DAHDI-Linux 2.5.0 there is not a hard coded limit on the
number of channels [1] or spans [2] that DAHDI can support. The
number of pseudo channels is limited by a module parameter,
'max_pseudo_channels', which currently defaults to 512 [3].

[1] http://svnview.digium.com/svn/dahdi?view=revision&revision=9609
[2] http://svnview.digium.com/svn/dahdi?view=revision&revision=9598
[3] http://svnview.digium.com/svn/dahdi?view=revision&revision=9610

> > is it possible to run system like that ? is it good idea , can
> > asterisk handle 2400 calls if machine size and RAM is good.
> 
> We've faced stability issues with more than 500 simultaneous calls on a 
> single high-powered server, with no transcoding.  However, that's 
> probably more a limitation of our own architecture and application than 
> a hard Asterisk limit.

I too would be very surprised if DHAVAL is able to run anything
close to 80 spans in a single server without modifying the drivers
to a) optimize for increased throughput versus minimal latency
(DAHDI currently favors low latency between bridged channels) and b)
Rework how conferencing / bridging is done in the drivers so that
only channels in conference with one another are checked for audio
mixing. I believe the limiting factor would be CPU cycles and not
memory.

As an aside, the most I've run in a single server are 24 T1s. But I
was not optimizing for density either.

Cheers,
Shaun

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] can we install 10 PCI card on asterisk

[Patrick Lists]

On 27-08-12 14:08, DHAVAL INDRODIYA wrote:

Hi All,

i would like to know if anyone has done or having idea regarding PRI
terminations in asterisk.

i have a requirement where i need to support 80 PRI in one machine i
have found a machine which have 10 PCI slots available

now i am thinking of arranging 8port sangoma card in this pci slots so i
can arrenge 10 card in that.

is it possible to run system like that ? is it good idea , can asterisk
handle 2400 calls if machine size and RAM is good.



I don't think Asterisk can handle that many DAHDI channels and I have 
never heard of an Asterisk box with more that 16 PRI's.


Taking a step back, do you really want to put all your eggs in one 
basket? what if the box fails? That's 2400 channels going down and 
unavailable until you fix it. That will cost a log of money and get you 
angry clients. It makes more sense to spread the lot across different 
servers. Besides that, is your telco willing to provide you with 80x PRI 
or will they insist on aggregating it to several E3 links or something 
higher (STM-1)?


If you really insist on going down this route have a look at FreeSWITCH 
or look at something like Cisco, Alcatel, Telco Systems etc.


Regards,
Patrick


--
_
-- 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] can we install 10 PCI card on asterisk

[Mike]

On 12-08-27 09:08 AM, DHAVAL INDRODIYA wrote:

Hi All,

i would like to know if anyone has done or having idea regarding PRI 
terminations in asterisk.


i have a requirement where i need to support 80 PRI in one machine i 
have found a machine which have 10 PCI slots available
That sounds like having too many eggs in one basket to me.  If you have 
a problem with that single machine, you're going to disrupt 2400 calls 
when you do. A more distributed approach with multiple machines running 
Asterisk and multiple standalone PRI gateways will be much more fault 
tolerant if designed correctly.


--
Looking for (employment|contract) work in the
Internet industry, preferrably working remotely.
Building / Supporting the net since 2400 baud was
the hot thing. Ask for a resume! ispbuil...@gmail.com


--
_
-- 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] can we install 10 PCI card on asterisk

[Raj Mathur (राज माथुर)]

On Monday 27 Aug 2012, DHAVAL INDRODIYA wrote:
> i would like to know if anyone has done or having idea regarding PRI
> terminations in asterisk.
> 
> i have a requirement where i need to support 80 PRI in one machine i
> have found a machine which have 10 PCI slots available
> 
> now i am thinking of arranging 8port sangoma card in this pci slots
> so i can arrenge 10 card in that.

Last I checked, the highest channel number DAHDI supported was 1023, 
limiting you to some 34 E1 PRIs.

> is it possible to run system like that ? is it good idea , can
> asterisk handle 2400 calls if machine size and RAM is good.

We've faced stability issues with more than 500 simultaneous calls on a 
single high-powered server, with no transcoding.  However, that's 
probably more a limitation of our own architecture and application than 
a hard Asterisk limit.

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

--
_
-- 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] can we install 10 PCI card on asterisk

[Mitul Limbani]

Not a good idea :)

Asterisk at max we have seen supported 16 PRIs on single machine.

Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422




On Mon, Aug 27, 2012 at 5:38 PM, DHAVAL INDRODIYA
wrote:

> Hi All,
>
> i would like to know if anyone has done or having idea regarding PRI
> terminations in asterisk.
>
> i have a requirement where i need to support 80 PRI in one machine i have
> found a machine which have 10 PCI slots available
>
> now i am thinking of arranging 8port sangoma card in this pci slots so i
> can arrenge 10 card in that.
>
> is it possible to run system like that ? is it good idea , can asterisk
> handle 2400 calls if machine size and RAM is good.
>
> let me know ideas and suggestions.
>
> thanks
> Dhaval
>
> --
> _
> -- 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] can we install 10 PCI card on asterisk

[DHAVAL INDRODIYA]

Hi All,

i would like to know if anyone has done or having idea regarding PRI
terminations in asterisk.

i have a requirement where i need to support 80 PRI in one machine i have
found a machine which have 10 PCI slots available

now i am thinking of arranging 8port sangoma card in this pci slots so i
can arrenge 10 card in that.

is it possible to run system like that ? is it good idea , can asterisk
handle 2400 calls if machine size and RAM is good.

let me know ideas and suggestions.

thanks
Dhaval
--
_
-- 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 tries reinvite when incompatible codecs on call legs

[Frederic Van Espen]

On Sat, 2012-08-18 at 10:55 +0200, Frederic Van Espen wrote:
> Hi,
> 
> I just ran into what seems to be an issue on re-invites. I'm not sure if
> it's a bug or as designed, so I thought I'd ask the question.
> 
> Here's my setup:
> - Asterisk 1.8.13.0
> - Phone A: Polycom ip331, only allowed to use ulaw, canreinvite=yes
> - Phone B: Polycom ip330, only allowed to use alaw, canreinvite=yes
> 
> Phone A calls the extension of phone B.
> 
> After the normal call setup asterisk tries the reinvite:
> - To phone B it sends an SDP which asks alaw and connection information
> of phone A
> - To phone A it sends and SDP with only the connection information of
> phone B
> 
> Phone A responds to this with a 488 Not acceptable here.
> 
> Asterisk ACK's and sends another set of reinvites to both phones with
> the correct codecs and connection information (the asterisk box itself).
> 
> Both phones reply 200 OK. Asterisk ACK's and immediately sends BYE to
> both phones.
> 
> Is this normal behaviour? I would expect one of two things:
> - Asterisk intelligently does not even try reinvite because of the
> incompatible codecs.
> - Asterisk tries the reinvite anyway, but does not end the call if it
> fails.
> 
> To be sure, I also tested this on asterisk 1.8.15.0 and
> certified-asterisk-1.8.11-cert4. Both have the same result.
> 
> Any help would be very appreciated!
> 
> Cheers,
> 
> Frederic Van Espen
> 

No one else encountered this or has any suggestions on what I might try?
Looks like I will be creating a bug report then.

Regards,

Frederic


--
_
-- 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 hangs while starting in OpenSuse 12.2

[Patrick Lists]

On 27-08-12 08:25, Gopalakrishnan N wrote:

This is really tuff working with OpenSuse. I am clueless how to sort our
this.


Maybe switch to a different distribution? I have used CentOS and RHEL 
for years without any problems and as far as I know both debian and 
ubuntu should work without problems too.


Regards,
Patrick



--
_
-- 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