[asterisk-users] Force different codecs on call base

2010-12-30 Thread Stefan Schmidt
Hello,

what i want to do is to find a way how i can solve the following problem.

we want to offer our customers in the country side also isdn over voip
but we have to use internet connections from another company for this.
This company offers a QoS on this connections but only with 192kbit
bandwith and with the ATM headers a normal g711a call has exactly 103,5
kbit/s so we can only use 1 channel but for isdn we need 2 :(

my idea was if i can find a way that the first call of a peer has g711a
codec (like normally) and if a second call comes in, or has to be placed
for this peer i only offer g726 (40kbit) so i dont have a bandwith issue.

is there a possible way of doing this or would it be easier to use two
peers, one with g711a and one with g726 and just let both only use one
channel?

thanks for your help!

best regards
stefan

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Re: [asterisk-users] Usage Reports

2010-12-30 Thread Duncan Turnbull
Freepbx really needs its own list but it doesn't seem to have one

But - if you have mysql setup and records being logged then the reports should 
show you usage on a daily, weekly, monthly level. Make sure you built asterisk 
with cdrs logged into mysql - its in the addons 

Cheers Duncan

On 30/12/2010, at 8:36 PM, Ben Schorr wrote:

 We’re using FreePBX 2.8 and there is a Reports tab but it doesn’t seem to 
 actually do anything.  Is there some secret/trick to getting a report out of 
 it that will tell us which extensions are placing calls?  I’ve tried every 
 query on the form that I can think of.  Is the reporting disabled by default 
 or ???
  
 Any tips/pointers appreciated.
  
 Ben M. Schorr
 Chief Executive Officer
 __
 Roland Schorr  Tower
 www.rolandschorr.com
 b...@rolandschorr.com
  
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Re: [asterisk-users] GotoIf CALLERID(num)

2010-12-30 Thread Doug Lytle

Joseph wrote:
I see my problem, caller ID is coming in as 7804715665 and I was 
blocking 4715665 I need to enter area code first, or can I use * as 
first digits?


It would be best just to match against the whole number.

Doug


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Re: [asterisk-users] No MOH with parked call

2010-12-30 Thread Steve Davies
On 24 December 2010 15:44, Steve Davies davies...@gmail.com wrote:
 On 23 December 2010 18:01, Steve Davies davies...@gmail.com wrote:
 Hi Again,

 I thought I had this sorted, but it appears that in a clean
 environment I did not in fact fix it. There appears to be a bit of a
 contradiction.

 1) In 1.6.2.x, musiconhold requires DAHDI (which we have)
 2) In 1.6.2.x parked calls get MOH only if res_timing_dahdi is not loaded...

 I am confused. MOH in general terms works just fine, but if I park a
 call with res_timing_dahdi loaded, I get silence after the orbit
 announcement. If I unload res_timing_dahdi, all works fine, but my
 timing sources are less reliable.

 I have backported res_musiconhold.c from 1.8 to 1.6.2.16-rc1, but this
 does not seem to fix things - is the problem elsewhere? Is there a fix
 that I can try, or perhaps backport?

 Further to this, I have been slowly tracing through the codepath for a
 parked call - ast_settimer is called correctly for the MOH generator,
 and seems to set up the DAHDI timer exactly the same way as it does
 for the alternative timing modules, and all of the setup calls return
 success. I don't have a verbose output trace to hand, but the sequence
 seems to be as follows:

 -- Start moh_files_generator successfully using ast_settimer()
   Chunks of MOH file are generated here
 -- Stop moh_files_generator successfully
 -- Play the parking position numbers
 -- Start moh_files_generator using ast_settimer()
 Generator is never called

 Any thoughts where to look next? It is odd that it appears to be
 working immediately before the position is read out, but then fails to
 restart afterwards...


Thanks to Lee pointing me in the right direction, I discovered

https://issues.asterisk.org/view.php?id=18262

Which seems to fix this issue.

Many thanks,
Steve

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Re: [asterisk-users] call is not going to Voicemail with 1,n

2010-12-30 Thread Joseph

Thank you William,

It looks nice but I like the previous one you suggested, it looks simpler and 
it easier to follow the logic :-)

--
Joseph


On 12/30/10 01:40, William Stillwell wrote:

Also, a more fancy approach

[macro-dialvm]

exten = s,1,NoOp(${ExTEN}|${MACRO_EXTEN}|${ARG1})
exten = s,n,Dial(SIP/${ARG1},25,t)
exten = s,n,NoOp(${ARG1})
exten = s,n,NoOp(${DIALSTATUS})
exten = s,n,GotoIf($[${DIALSTATUS} = BUSY]?BUSY)
exten = s,n,GotoIf($[${DIALSTATUS} = NOANSWER]?NOANSWER)
exten = s,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?CHANUNAVAIL)
exten = s,n,VoiceMail(${ARG1},a)
exten = s,n,MacroExit()

exten = s,n(BUSY),Set(CDR(userfield)=DIAL-BUSY)
exten = s,n,NoO${MACRO_EXTEN})
exten = s,n,NoOp(${ARG1})
exten = s,n,ResetCDR(w)
exten = s,n,VoiceMail(${ARG1},b)
exten = s,n,MacroExit()

exten = s,n(NOANSWER),Set(CDR(userfield)=DIAL-NOANSWER)
exten = s,n,NoOp(${MACRO_EXTEN})
exten = s,n,NoOp(${ARG1})
exten = s,n,ResetCDR(w)
exten = s,n,VoiceMail(${ARG1},u)
exten = s,n,MacroExit()

exten = s,n(CHANUNAVAIL),Set(CDR(userfield)=DIAL-UNAVIL)
exten = s,n,NoOp(${MACRO_EXTEN})
exten = s,n,NoOp(${ARG1})
exten = s,n,ResetCDR(w)
exten = s,n,VoiceMail(${ARG1},uj)
exten = s,n,MacroExit()

exten = s,BUSY+101,Set(CDR(userfield)=DIAL-BSY-NOMBX)
exten = s,n,NoOp(${MACRO_EXTEN})
exten = s,n,NoOp(${ARG1})
exten = s,n,ResetCDR(w)
exten = s,n,NoOp(Mailbox Not found)
exten = s,n,Goto(CHANUNAVAIL+101)

exten = s,NOANSWER+101,Set(CDR(userfield)=DIAL-NA-NOMBX)
exten = s,n,NoOp(${MACRO_EXTEN})
exten = s,n,NoOp(${ARG1})
exten = s,n,NoOp(MailBox Not found)
exten = s,n,Goto(CHANUNAVAIL+101)

exten = s,CHANUNAVAIL+101,Playback(/home/asterisk/gen/themailbox)
exten = s,n,NoOp(${MACRO_EXTEN})
exten = s,n,NoOp(${ARG1})
exten = s,n,SayDigits(${MACRO_EXTEN})
exten = s,n,Playback(/home/asterisk/gen/doesnotexist)
exten = s,n,MacroExit()



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Wednesday, December 29, 2010 11:56 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] call is not going to Voicemail with 1,n

I've tried to simplified the dial plan and use n instead of numbers but
I've noticed it is not executing my voicemail if I substitute number with
n

In the example below when the call is not answered, it does not go to
voicemail; call just hangup.

exten = 1,1,Playback(transfer)
exten = 1,n,Dial(${sales_support}IAX2/iaxy-322,20,jrw)
exten = 1,103,Voicemail(11,b)
exten = 1,104,Hangup()
exten = 1,n,Voicemail(11,b) ; Right to voicemail
exten = 1,n,Hangup()

Here is the transcript:

-- Executing [...@office-open:1] Playback(SIP/pstn-5665-00be,
transfer) in new stack
-- SIP/pstn-5665-00be Playing 'transfer' (language 'en')
-- Executing [...@office-open:2] Dial(SIP/pstn-5665-00be,
SIP/11IAX2/iaxy-322|20|jrw) in new stack
-- Called 11
-- Called iaxy-322
-- Call accepted by 10.0.0.108 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxy-322-8406 is busy
-- Hungup 'IAX2/iaxy-322-8406'
-- SIP/11-00bf is ringing
-- Nobody picked up in 2 ms
  == Auto fallthrough, channel 'SIP/pstn-5665-00be' status is
'NOANSWER'


However, if I number the dial plan in the old fashion way and don't answer
the phone it goes to voicemail just fine:

exten = 1,1,Playback(transfer)
exten = 1,2,Dial(${sales_support}IAX2/iaxy-322,20,jrw)
exten = 1,103,Voicemail(11,b)
exten = 1,104,Hangup()
exten = 1,3,Voicemail(11,b) ; Right to voicemail
exten = 1,4,Hangup()

--
Joseph


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[asterisk-users] VUC; Friday December 31st - 2010: The Year in VoIP

2010-12-30 Thread Michael Graves
On this weeks VUC call we will collectively be our own guests. That is,
we'd like to know what was the big issue that impacted YOU in 2010? All
opinions welcome.

Here are a few things to get you thinking in advance:

- Apple's Antenna-gate
- Asterisk 1.8 Launches
- Amazon EC2 as a DOS platform
- Cisco launched UMI video conference device
- More HDVoice capable phones
- Skype Outage
- VoIP on mobile devices
- or perhaps something more personal.

Come one, come all. Bring your story.

Connect details at http://vuc.me

Michael Graves
--
Michael Graves
mgravesatmstvp.com
http://www.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
Twitter mjgraves




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Re: [asterisk-users] call is not going to Voicemail with 1,n

2010-12-30 Thread Joseph

On 12/30/10 01:38, William Stillwell wrote:


Try this instead:

exten = 1,1,Playback(transfer)
exten = 1,n,Dial(${sales_support}IAX2/iaxy-322,20,jrw)
exten = 1,n,Voicemail(11,b)
exten = 1,n,Hangup()
exten = 1,n+101,Voicemail(11,b)
exten = 1,n,Hangup()

which will result in a DP looking like this:

exten = 1,1,Playback(transfer)
exten = 1,2,Dial(${sales_support}IAX2/iaxy-322,20,jrw)
exten = 1,3,Voicemail(11,b)
exten = 1,4,Hangup()
exten = 1,105,Voicemail(11,b)
exten = 1,106,Hangup()


I was analyzing it and I think in the above:
exten = 1,n+101,Voicemail(11,b)
is not needed at all as it does not take any effect. The (j) in the dial tell is to jump 
but there is no 103 so it falls through to 3 which is voicemail.
so actually j not needed nor are the:

exten = 1,n+101,Voicemail(11,b)
exten = 1,n,Hangup()

--
Joseph

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Re: [asterisk-users] VUC: VUC; Friday December 31st - 2010: The Year in VoIP

2010-12-30 Thread Michael Graves
Oh, yes. There are some contest, too. Another Gigaset DE380 IP-R desk
phone, and a DIDX marketing package are up for grabs.

Details as always at http://vuc.me

On Thu, 30 Dec 2010 12:29:13 -0600, Michael Graves wrote:

On this weeks VUC call we will collectively be our own guests. That is,
we'd like to know what was the big issue that impacted YOU in 2010? All
opinions welcome.

Here are a few things to get you thinking in advance:

- Apple's Antenna-gate
- Asterisk 1.8 Launches
- Amazon EC2 as a DOS platform
- Cisco launched UMI video conference device
- More HDVoice capable phones
- Skype Outage
- VoIP on mobile devices
- or perhaps something more personal.

Come one, come all. Bring your story.

Connect details at http://vuc.me

Michael Graves
--
Michael Graves
mgravesatmstvp.com
http://www.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
Twitter mjgraves



-- 

http://vuc.me  - Fridays 12 Noon ET : http://vuc.me/next

IRC freenode.net #vuc or web: http://vuc.me/irc
Twitter: follow 'voipusers' and 'vucmailinglist' for the RSS feed of this list

Thanks to Voxeo for their support and to all the participants

Unsubscribe email: voip-users-conference-unsubscr...@googlegroups.com


--
Michael Graves
mgravesatmstvp.com
http://www.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
Twitter mjgraves




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[asterisk-users] Find media and sip endpoints IP address durring h extension

2010-12-30 Thread Bryant Zimmerman
How can I get the media and sip endpoints IP address durring h 
extension?

I need to write these to my CEL logs.

Any ideas?

Thanks
Bryant


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Re: [asterisk-users] Find media and sip endpoints IP address durringh extension

2010-12-30 Thread Danny Nicholas
  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Thursday, December 30, 2010 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Find media and sip endpoints IP address durringh
extension

 

How can I get the media and sip endpoints IP address durring h extension?

I need to write these to my CEL logs.

Any ideas?

Thanks
Bryant

Possibly from the SIP headers.  Probably have to capture this information
before the hangup occurs (DeadAGI might be able to extract this).

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[asterisk-users] Users of CEL Please comment on Bug

2010-12-30 Thread Bryant Zimmerman
If you are using CEL in asterisk 1.8 can you please look at the issue 
tracker and comment.
On how this might effect you.

https://issues.asterisk.org/view.php?id=18559

Thanks
Bryant
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[asterisk-users] Base memory usage

2010-12-30 Thread Larry Wimble


Asterisk gurus

I just installed asterisk 1.8.1.1 along with FreePBX on a fairly small 
VPS (512mb standard, 512mb burst).  I note that the asterisk process 
is using about 209mb of memory just doing nothing (not configured to do 
anything yet)


In contrast to this, my 1.6.1.2 installation from a little over a year 
ago uses only 40mb and it's fully configured and running with about 4 
months of uptime (2 trunks, 4 channels, 3 DIDs, and 4 extensions.)


Any ideas on how I can get the memory consumption down on my new 
installation, or is it time to downgrade to the older version?


Thanks,
Larry






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Re: [asterisk-users] Base memory usage

2010-12-30 Thread Faisal Hanif

Hi,

1-TuneUp your setting in /etc/dafult/asterisk
2-Stop l;oadng all not required modules by adding noload = 
modulename.so  lines to /etc/asterisk/modules.conf


Regards,

Faisal

On 12/31/2010 7:59 AM, Larry Wimble wrote:


Asterisk gurus

I just installed asterisk 1.8.1.1 along with FreePBX on a fairly small 
VPS (512mb standard, 512mb burst).  I note that the asterisk process 
is using about 209mb of memory just doing nothing (not configured to 
do anything yet)


In contrast to this, my 1.6.1.2 installation from a little over a year 
ago uses only 40mb and it's fully configured and running with about 4 
months of uptime (2 trunks, 4 channels, 3 DIDs, and 4 extensions.)


Any ideas on how I can get the memory consumption down on my new 
installation, or is it time to downgrade to the older version?


Thanks,
Larry






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Re: [asterisk-users] Base memory usage

2010-12-30 Thread Jeremy Kister

On 12/30/2010 9:59 PM, Larry Wimble wrote:

I just installed asterisk 1.8.1.1 along with FreePBX on a fairly small
VPS (512mb standard, 512mb burst).  I note that the asterisk process
is using about 209mb of memory just doing nothing (not configured to do
anything yet)


I'm running 1.8.1 rc1 + some patches (nothing to do with memory) and i'm 
at 42MB resident (73 virt/8shared)


I've got just about everything turned on via menuselect, but then i have 
a bunch of modules turned off via modules.conf


I doubt that's your issue, but if you're interested to see my 
modules.conf, it's temporarily at http://jeremy.kister.net/tmp/modules.conf


--

Jeremy Kister
http://jeremy.kister.net./

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Re: [asterisk-users] Usage Reports

2010-12-30 Thread Ben Schorr
I think we installed it with all of the defaults - but maybe MYSQL isn't
there.  I'll check on that, thanks!

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com http://www.rolandschorr.com/ 
b...@rolandschorr.com mailto:b...@rolandschorr.com 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Duncan
Turnbull
Sent: Thursday, December 30, 2010 2:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Usage Reports

 

Freepbx really needs its own list but it doesn't seem to have one

 

But - if you have mysql setup and records being logged then the reports
should show you usage on a daily, weekly, monthly level. Make sure you
built asterisk with cdrs logged into mysql - its in the addons 

 

Cheers Duncan

 

On 30/12/2010, at 8:36 PM, Ben Schorr wrote:





We're using FreePBX 2.8 and there is a Reports tab but it doesn't seem
to actually do anything.  Is there some secret/trick to getting a report
out of it that will tell us which extensions are placing calls?  I've
tried every query on the form that I can think of.  Is the reporting
disabled by default or ???

 

Any tips/pointers appreciated.

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com http://www.rolandschorr.com/ 
b...@rolandschorr.com mailto:b...@rolandschorr.com 

 

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