date:20050921



Hey ho,

I suppose you are the person from the digium forum :)

The reason i recommended you to use a ramdisk is because i think the
problem with recording to disk is saving 20ms of stream 1, then 20 ms of
stream 2, then 20ms of stream 3 etc etc meaning you write everytime
very small things. (with a lot of seeking).
Our best test results were with:

- buffering the recordings to a ramdisk, then
- on low load (at night) copy the files over the network (easy to shape
the pipe, so that you dont overload anything), This way, the memory
buffer will take care of the 'fragmentation' and not your harddisk.
- on the remove server, do all the mixing / indexing etc. (i really
don't mixing or converting between audio formats on the same server as
asterisk).

If you want to go even freakier, run asterisk (or you complete distro)
from a ramdisk.

Oh, another thing, for the people trying this the performance of hdparm
is not linear with the quality of your calls, tweaking your disks to be
faster will not help for asterisk when you do a copy. (in general).

I thought over your suggestion to use a sniffer to do the recordings,
you might pull it off, but will have to write your own to do so. (or go
to the expensive version of commercial sniffer applications).

Zoa.

-
www.asteriskguru.com


Matt Florell wrote:


On 9/20/05, *Matt Roth* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:

Patrick,

Thank you for your suggestions.

Our initial runs were recording directly to an NFS mount and they
experienced the same problems as recording to the local disk.  In
our final setup, the copy will be done to an NFS mount as long as
it exists, falling back to local disk only when the NFS server is
down.

The theory that we're running on is that any I/O bottlenecks (or
network latencies in the case of NFS) only matter when they are
bound to a call in progress.  In that scenario, the bottleneck
would introduce a latency in Asterisk's handling of the RTP
packets causing call degradation and drops.  By decoupling I/O
from live calls and performing the copies (a very lightweight
operation) in a separate process, we hope to not affect Asterisk's
real-time handling of the RTP packets.

Because of limited access to the test equipment, we were only able
to test up to storing the digital recordings on a RAM disk. Please
shoot holes in this setup if you see any weaknesses.  Better today
than on our go-live date.

Thanks,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


 Hello,

I'm very interested in the specifics of your setup.
How much space is on the RAM disk?
What kind of RAM drive is it?
What format are you recording to?
What codec are the SIP calls being placed over?

We've run into the Avoided deadlock recording issues several times
when trying to do more than 50 concurrent recordings. Changing the
ast_channel_lock loop from 10 to 20 has helped somewhat reduce the
warnings and reduce audio gaps on the recordings, but what is really
needed for more robust recording is a configurable recording buffer
that wouldn't freak out if a 10ms delay occurs.

Good luck and please keep us updated on your progress,

MATT---





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Re: [Asterisk-Users] Cisco Ip phones

2005-09-21 Thread Florian Overkamp


Hi,

Michiel van Baak wrote:

What about the license?? And do you have to buy a license and changing the
phone to sip protocol looks scary :( and time consuming with 100 phones not
all suppliers will do it for you, and does any of you know a supplier in the
netherlands with good pricing neonova is way too expensive 


I got mine from www.centralpoint.nl
As far as I know they only deliver the phones with SCCP
image. But as you can read in my previous mail this is no
problem, simply install chan_sccp.
If you want the phones to run SIP, you have to buy a license
for the SIP image. Centralpoint has them too.


My company is a cisco supplier too, maybe we can arrange some pricing 
strategies together. However, Cisco remains an expensive phone.


Be aware, you cannot really compare delivery from any dutch supplier to 
what you find on Ebay. We only deal in new stock, nothing refurb, and 
yes, they are expensive.


Florian
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Re: [Asterisk-Users] iax2 trunking wackyness

2005-09-21 Thread Matt Riddell

Clive wrote:
 Hi
 
 I was doing some bandwidth testing, and my incomming usage is 
 36% more than my outgoing bandwidth.

In my case the calls come in separately (i.e. untrunked) and get trunked by
the Asterisk machine and sent out.  This causes an imbalance.

Are your calls coming from many to one or one to one?

-- 
Cheers,

Matt Riddell
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[Asterisk-Users] Brand New IPSwitchBoard

2005-09-21 Thread Thorben Jensen

I am proud to announce that I have just released a brand new IPSwitchBoard
totally rewritten from scratch. The new version has been built on the
experiences gained by the previous versions.

Download for FREE: http://ipsoftware.thorben.dk

IPSwitchBoard is totally customizable and will give you, among other things:

Unattended/attended transfers. 
Park calls and retrieve/forward them again. 
Organize all your SIP, IAX extensions (automatically retrieved from
Asterisk). 
Monitor all extensions, queues and Parked Calls. 
Dynamically log extensions in and out of queues. 
Set Do Not Disturb on Extensions and give a reason
Set Call forwarding for extensions
Set Dual call for extensions
Monitor multiple calls on an Extension/Queue
Monitor Extension online status
Totally customizable
Make you own skins with logo's etc.
Make your own buttons - DND, Online, Queue Status, Call Forward, Dual Call,
MWI etc.

IPDesigner is a unique tool for customizing and setting up IPSwitchBoard.
With IPDesigner you can design your own IPSwitchBoard with Company logo and
all the buttons you need for the Operator.

IPSwitchBoard works with .IPS files. These files contain everything needed
for IPSwitchBoard such as bitmaps, server and extension configuration.  You
can build .IPS files with IPDesigner.When you have installed IPSwitchBoard
you can start IPSwitchBoard by double clicking an .IPS file. If you right
click an .IPS file you will have the options:

Open - will open the file in IPSwitchBoard
Edit - will open the file in IPDesigner
Configure - will start the configuration program where you can edit the
server configuration.
Extract Bitmaps - will extract any bitmaps there's embedded in the .IPS
file.

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[Asterisk-Users] Re: HooDaHek 0.6 Released

2005-09-21 Thread Tony Mountifield

In article [EMAIL PROTECTED],
Shawn Rutledge [EMAIL PROTECTED] wrote:
 The eth suffix is oft abused, and oft he who writeth it
 knoweth not the rules.

Such as IBM (or their ad agency), who came up with IBM can helpeth
in a TV advert!!!

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

2005-09-21 Thread trixter http://www.0xdecafbad.com

On Wed, 2005-09-21 at 10:07 +0300, Zoa wrote: 
 The reason i recommended you to use a ramdisk is because i think the
 problem with recording to disk is saving 20ms of stream 1, then 20 ms of
 stream 2, then 20ms of stream 3 etc etc meaning you write everytime
 very small things. (with a lot of seeking).
 Our best test results were with:
 
filesystems are also a consideration with larger scale projects.
Different filesystems add different amounts of overheads on different
types of operations.  Some are faster at moving small files around
others faster with large files.  This adds to the disk latency.
Removing the disk latency itself is a good thing, since that is
typically slower, but to crank out that last little bit of performance
some research into the different filesystems under the specific kernel
that you are using could also be a consideration.  The most obvious
(and easiest to update a running system) is to remove things like atime,
whih with most linux distros is on by default.  This causes a write
operation for the read of a file to update the last time accessed.  A
couple little things can add up to a few percent improvement and
generally make the cost go down.


 - buffering the recordings to a ramdisk, then
 - on low load (at night) copy the files over the network (easy to shape
 the pipe, so that you dont overload anything), 
Or have a seperate network set up (dual nic card for example) where the
2nd network is used just for NFS traffic.  Although NFS generally is
ugly network wise, it is standard and makes things easier.  Just gotta
watch the IO on the system given that the network card itself will cause
cpu cycles to be used, but lets face it cpu is cheap now.  Different
drivers also work differently, and then with the 2.6 series kernels you
can use device polling instead of interupts which can help a little.



 If you want to go even freakier, run asterisk (or you complete distro)
 from a ramdisk.
 
When you say ramdisk here I assume you mean using conventional ram, its
cheap yes but its volatile, do you have any plans for failure of the
system or ram?  Or is the data integrity itself not as critical?  The
reason that people like hard drives is because most of the time if the
system goes down for any reason the data is still intact.  


 I thought over your suggestion to use a sniffer to do the recordings,
 you might pull it off, but will have to write your own to do so. (or go
 to the expensive version of commercial sniffer applications).
 
isnt vomit free?  It was a voip sniffer that worked with some codecs
many years ago (I wanna say mid-late 90s but I may be thinking of
another back then). http://vomit.xtdnet.nl/ does G.711 only.

The bigger prIoblem that I see is that sniffers dont always get all the
traffic that is on a network particularly when the network has more
traffic on it.  While this generally isnt a concern and I would like to
think that even a poorly configured network could allow for 512 calls,
it is a factor to implement this type of a solution.

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording



Also when you do things over the network, disable your onboard network
card, and go for some more expensive network card.
In our tests with small packets, we could increase the throughput with a
factor 2. (related to cpu load).

Zoa.

--
www.asteriskguru.com

trixter http://www.0xdecafbad.com wrote:


On Wed, 2005-09-21 at 10:07 +0300, Zoa wrote:



The reason i recommended you to use a ramdisk is because i think the
problem with recording to disk is saving 20ms of stream 1, then 20 ms of
stream 2, then 20ms of stream 3 etc etc meaning you write everytime
very small things. (with a lot of seeking).
Our best test results were with:




filesystems are also a consideration with larger scale projects.
Different filesystems add different amounts of overheads on different
types of operations.  Some are faster at moving small files around
others faster with large files.  This adds to the disk latency.
Removing the disk latency itself is a good thing, since that is
typically slower, but to crank out that last little bit of performance
some research into the different filesystems under the specific kernel
that you are using could also be a consideration.  The most obvious
(and easiest to update a running system) is to remove things like atime,
whih with most linux distros is on by default.  This causes a write
operation for the read of a file to update the last time accessed.  A
couple little things can add up to a few percent improvement and
generally make the cost go down.





- buffering the recordings to a ramdisk, then
- on low load (at night) copy the files over the network (easy to shape
the pipe, so that you dont overload anything),



Or have a seperate network set up (dual nic card for example) where the
2nd network is used just for NFS traffic.  Although NFS generally is
ugly network wise, it is standard and makes things easier.  Just gotta
watch the IO on the system given that the network card itself will cause
cpu cycles to be used, but lets face it cpu is cheap now.  Different
drivers also work differently, and then with the 2.6 series kernels you
can use device polling instead of interupts which can help a little.






If you want to go even freakier, run asterisk (or you complete distro)
from a ramdisk.




When you say ramdisk here I assume you mean using conventional ram, its
cheap yes but its volatile, do you have any plans for failure of the
system or ram?  Or is the data integrity itself not as critical?  The
reason that people like hard drives is because most of the time if the
system goes down for any reason the data is still intact.





I thought over your suggestion to use a sniffer to do the recordings,
you might pull it off, but will have to write your own to do so. (or go
to the expensive version of commercial sniffer applications).




isnt vomit free?  It was a voip sniffer that worked with some codecs
many years ago (I wanna say mid-late 90s but I may be thinking of
another back then). http://vomit.xtdnet.nl/ does G.711 only.

The bigger prIoblem that I see is that sniffers dont always get all the
traffic that is on a network particularly when the network has more
traffic on it.  While this generally isnt a concern and I would like to
think that even a poorly configured network could allow for 512 calls,
it is a factor to implement this type of a solution.





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[Asterisk-Users] DID problem with calls from analog to ISDN

2005-09-21 Thread Christian Gansberger

hi all,

I have this anoying problem with Direct in Dial, when someone calls
from an extern analog phone to my isdn number, Asterisk is not waiting
for the DID numbers the caller dials after the main-number.

Version: Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n, with junghanns quadbri

zapata.conf:

switchtype = euroisdn
signalling = bri_cpe
faxdetect=incoming
pridialplan = unknown
prilocaldialplan = unknown
echocancel=yes
echotraining = 100
callerid=asreceived
usecallerid=yes
overlapdial=yes
immediate=no
;---
group = 1
context=isdn-buero
channel = 1-2
channel = 4-5

group = 2
context=isdn-res
channel = 7-8
channel = 10-11

zaptel.conf:

loadzone=nl
defaultzone=nl
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
span=2,1,3,ccs,ami
bchan=4-5
dchan=6
span=3,1,3,ccs,ami
bchan=7-8
dchan=9
span=4,1,3,ccs,ami
bchan=10-11
dchan=12


To me it looks like immediate=no doesn't do anything.
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[Asterisk-Users] Call getting disconnected in queue

2005-09-21 Thread Rajkumar S


Hi,

I have a small call center with 4 Zap lines and 4 agents. Agents login 
using sip phones with AgentCallbackLogin. I occasionally gets a 
complaint that when customers call the call center, after the initial 
greeting is over the call gets cut after playing the thank you message. 
I started investigating and found that that happens when the call gets 
transferred to an agent who is making an outbound call (either calling 
customers or logging out). The debug logs of one such conversation is 
given below:


As you can read below, the call gets fwd to agent 1005 at SIP/1004. But 
he is trying to log off at the same time, and call gets disconnected.


Any help to fix this will be very much appreciated.

regards,

raj

   -- Executing Answer(Zap/2-1, ) in new stack
-- Executing Goto(Zap/2-1, MainMenu|s|1) in new stack
-- Goto (MainMenu,s,1)
-- Executing BackGround(Zap/2-1, Welcome) in new stack
-- Playing 'Welcome' (language 'en')
-- Playing 'agent-incorrect' (language 'en')
  == Spawn extension (from-sip, 2002, 1) exited non-zero on 'SIP/1004-cc2f'
-- Executing Queue(Zap/2-1, callcenter|tT|||300) in new stack
-- Started music on hold, class 'default', on Zap/2-1
-- Stopped music on hold on Zap/2-1
-- Playing 'queue-youarenext' (language 'en')
-- Executing AgentCallbackLogin(SIP/1004-e376, |l) in new stack
-- Playing 'agent-user' (language 'en')
-- Told Zap/2-1 in callcenter their queue position (which was 1)
-- Playing 'queue-thankyou' (language 'en')
-- Started music on hold, class 'default', on Zap/2-1
-- outgoing agentcall, to agent '1005', on 'Local/[EMAIL PROTECTED],1'
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1004) in new 
stack
Sep 21 11:56:39 ERROR[12633]: chan_sip.c:1615 update_user_counter: Call 
from user '1004' rejected due to usage limit of 1

-- Couldn't call 1004
  == Everyone is busy/congested at this time
-- Called Agent/1005
-- Playing 'agent-incorrect' (language 'en')
  == Spawn extension (from-sip, 2002, 1) exited non-zero on 'SIP/1004-e376'
-- Timeout on Local/[EMAIL PROTECTED],2
  == CDR updated on Local/[EMAIL PROTECTED],2
-- Executing BackGround(Local/[EMAIL PROTECTED],2, vm-goodbye) 
in new stack

-- Playing 'vm-goodbye' (language 'en')
-- Agent/1005 answered Zap/2-1
-- Stopped music on hold on Zap/2-1
-- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack
  == Spawn extension (from-sip, t, 2) exited non-zero on 
'Local/[EMAIL PROTECTED],2'
monitor executing ( nice -n 19 soxmix 
/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-in.wav 
/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-out.wav 
/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010.wav 
  rm -f 
/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-* 
) 

  == Spawn extension (MainMenu, s, 2) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'

sip.conf entry for the phone is

[1004]
host=dynamic
type=friend
dtmfmode=RFC2833
username=1004
secret=password
context =  from-sip
disallow=all
allow=speex
allow=gsm
incominglimit=1

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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

2005-09-21 Thread trixter http://www.0xdecafbad.com

On Wed, 2005-09-21 at 11:11 +0300, Zoa wrote:
 Also when you do things over the network, disable your onboard network
 card, and go for some more expensive network card.
 In our tests with small packets, we could increase the throughput with a
 factor 2. (related to cpu load).

I wonder how much of that is a poorly written driver and not the card
itself.  I have seen some fairly poor drivers performance wise.  :/ 


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording



True...  But i tried several brands of cards, and several drivers, the
dual nic gigabit intel card was a lot better than all the other
combinations i tried.

zoa

trixter http://www.0xdecafbad.com wrote:


On Wed, 2005-09-21 at 11:11 +0300, Zoa wrote:



Also when you do things over the network, disable your onboard network
card, and go for some more expensive network card.
In our tests with small packets, we could increase the throughput with a
factor 2. (related to cpu load).




I wonder how much of that is a poorly written driver and not the card
itself.  I have seen some fairly poor drivers performance wise.  :/






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[Asterisk-Users] Integrating * with Mitel SX2000 Lite

2005-09-21 Thread John Mylchreest

Hello there.

I have noticed you were trying to place an asterisk box infront of an
SX2000.
I am trying to do this also, but no matter what I try the connection to
the SX2000 causes a major alarm.

Did you get this working?
If so could I please see what signalling you used to achieve this?

Thanks,
John
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[Asterisk-Users] Intermitant delays on call setup.

2005-09-21 Thread Morgan Gilroy

We are seeing this weird problem, it seems to happen at random periods
throughout the day from a few minuets to a up to an hour.

[Phone A] --SIP-- [Asterisk] --SIP-- [Phone B]
Both phones are snom 360's.
Asterisk is Stable 1.0.9
Pretty simple config, just a dial direct to each other like
Dial(SIP/phoneA,30,t)
Running Gentoo linux

When we make a call during one of the problem periods, from [Phone A]
extension to [Phone B] there is up to 2 seconds delay before A hears B.

Looking at a packet trace we see the SIP invites coming in and the calls
being setup ok, but on the call to B we see the RTP from B to * and A to
* but nothing being sent from * to either of the phones, then after
about a second we see a while bunch of RTP packets being sent out all in
one go to both phones. It does this 'bunching' a couple of times then
settles down to normal.

Id say it was some sort of timing problem or load problem, but during
these times conferencing etc works ok and there is no appreciable load
on the server or network.

Anyone have any ideas?
Thanks.

Morgan Gilroy.
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[Asterisk-Users] How to retrieve voicemail from an IP phone?

2005-09-21 Thread Ryan Pagquil


Hi,
  How can I retrieve those voicemails using my ip phone? and how 
will i confiugre it on asterisk?


Please help I'm very new in asterisk.

Thanks,

--
Ryan Pagquil
Infodyne Inc. - PhilOnline.com
3603 Antel Global Corporate Center
Doña Julia Vargas Ave.
Ortigas Center Pasig City
Tel: 687-0715
Web: www.philonline.com

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Re: [Asterisk-Users] How to retrieve voicemail from an IP phone?


How can I retrieve those voicemails using my ip phone? and how 
 will i confiugre it on asterisk?
 
 Please help I'm very new in asterisk.

Add something like this in your extensions.conf file:

; Voicemail access (prompts for exten and password)
exten = 3998,1,Wait,1
exten = 3998,2,VoicemailMain
exten = 3998,3,Hangup

; Voicemail access (does not prompt for anything)
exten = 3999,1,Wait,1
exten = 3999,2,VoicemailMain(s${CALLERIDNUM})
exten = 3999,3,Hangup



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Re: [Asterisk-Users] How to retrieve voicemail from an IP phone?

2005-09-21 Thread Ryan Pagquil


Hi Rich,
   Does the user need to dial his extension just to retrieve the 
voicemails or he will dial other number to access those voicemails?
In the config does it mean that when a user dial 3998 he will be able to 
retrieve those voicemails? So it means that every users must have a 
mailbox number for which they will retrive their voicemails? I'm really 
a newbie. =)


Thanks fo the help,
--ryan

Rich Adamson wrote:

  How can I retrieve those voicemails using my ip phone? and how 
will i confiugre it on asterisk?


Please help I'm very new in asterisk.
   



Add something like this in your extensions.conf file:

; Voicemail access (prompts for exten and password)
exten = 3998,1,Wait,1
exten = 3998,2,VoicemailMain
exten = 3998,3,Hangup

; Voicemail access (does not prompt for anything)
exten = 3999,1,Wait,1
exten = 3999,2,VoicemailMain(s${CALLERIDNUM})
exten = 3999,3,Hangup



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Tel: 687-0715
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Re: [Asterisk-Users] How to retrieve voicemail from an IP phone?

2005-09-21 Thread Rudolf Ladyzhenskii


Hi,

You need a single extension to call voicemail. I am using 100.
extensions.conf
exten =100,1,VoiceMailMain(${CALLERIDNUM})
exten =100,2,Hangup()

Now, if you simply call VoiceMailMain() without parameters, voicemail system 
will ask you to enter the number of mailbox you want to access. This is 
useful if you want to read any mailbox from any phone.
However, if you specify a parameter like I did, voicemail will automatically 
go into mailbox for the extension you have called from. There is a little 
trick to get it work, though. Normally caller ID is a name like Joe Smith
You will have to specify caller ID per user like that: (sip.conf for 
example)

[user1]
callerid=Joe Smith 101

This will present asterisk with a way to get both name and extension number.

Rudolf

- Original Message - 
From: Ryan Pagquil [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Wednesday, September 21, 2005 8:58 PM
Subject: Re: [Asterisk-Users] How to retrieve voicemail from an IP phone?



Hi Rich,
   Does the user need to dial his extension just to retrieve the 
voicemails or he will dial other number to access those voicemails?
In the config does it mean that when a user dial 3998 he will be able to 
retrieve those voicemails? So it means that every users must have a 
mailbox number for which they will retrive their voicemails? I'm really a 
newbie. =)


Thanks fo the help,
--ryan

Rich Adamson wrote:

  How can I retrieve those voicemails using my ip phone? and how 
will i confiugre it on asterisk?


Please help I'm very new in asterisk.



Add something like this in your extensions.conf file:

; Voicemail access (prompts for exten and password)
exten = 3998,1,Wait,1
exten = 3998,2,VoicemailMain
exten = 3998,3,Hangup

; Voicemail access (does not prompt for anything)
exten = 3999,1,Wait,1
exten = 3999,2,VoicemailMain(s${CALLERIDNUM})
exten = 3999,3,Hangup



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--
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Infodyne Inc. - PhilOnline.com
3603 Antel Global Corporate Center
Doña Julia Vargas Ave.
Ortigas Center Pasig City
Tel: 687-0715
Web: www.philonline.com

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Re: [Asterisk-Users] How to retrieve voicemail from an IP phone?

2005-09-21 Thread Pisac

exten = 3999,2,VoicemailMain(s${CALLERIDNUM})
if you extension is 104, then it will be converted inside asterisk to:
exten = 3999,2,VoicemailMain(s104)

and that will give to you access to mailbox 104 without passwordprompt (s=skip) and you can retreive messages.

${CALLERIDNUM} is extension (caller id)number of caller, and caller gets his own mailbox.
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Re: [Asterisk-Users] iax2 trunking wackyness

2005-09-21 Thread Clive

On 21 Sep 2005 at 19:48, Matt Riddell wrote:

 Clive wrote:
  Hi
  
  I was doing some bandwidth testing, and my incomming usage is 
  36% more than my outgoing bandwidth.
 
 In my case the calls come in separately (i.e. untrunked) and get trunked by
 the Asterisk machine and sent out.  This causes an imbalance.
 
 Are your calls coming from many to one or one to one?
 
 -- 
 Cheers,
 
 Matt Riddell

Hi

My setup is:   telco-asterisk(voip)-asterisk{ITSP}telco

so there should be an almost balanced transmit and receive rate on 
the voip leg.
My suspicion is that perhaps the packets are not getting trunked on 
the ITSP side.

regards
Clive




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[Asterisk-Users] IAX2 registration

2005-09-21 Thread Christoph Eicke

Hi!

I have the following setup:
PSTN1  Asterisk1 --- IAX2 --- Asterisk2  PSTN2
As you can see, two Asterisk machines are connected via IAX2. There are users 
connected to each Asterisk machine over a local LAN. Each of these users in 
both LANs should be able to either use PSTN1 or PSTN2, depending on which 
extension they dial, no problem here.
My question now is, does each Asterisk server need to register with the other 
Asterisk machine, or is it sufficient if only one Asterisk registers with the 
other one? Will the other one then know about the connection and be able to 
make phone calls over that one?

Thanks,
Christoph
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[Asterisk-Users] First release of the Asteriskguru Operator Panel



I'm proud to announce the first version (early alpha) of the
asteriskguru operator panel, (finally!)

Its available for download on : http://www.asteriskguru.com/tools/

No documentation is available yet. (working on that, it will be
available later on the same url).

Features:
--

- Support for multiple servers .
- Optimized for speed (and big installations with a large number of
peers / calls).
- Monitor IAX2 and SIP peers status.
- Monitor multiple channels per peer.
- Monitor ZAP channels status.
- Monitor Queues, Agents and Parked calls.
- Assigning peers to Queue with Dragdrop and removing them with context
menu.
- Originating calls between the peers with DragDrop.
- Transfering channels between the peers with DragDrop.
- Transfering channels to a ZAP channel(external lines).
- Full peer info on tooltip(hint)
- Full queue info on tooltip
- Queue membership per peer on tooltip
- Hangup channels.
- Automatic monitoring of calls.
- Queue usage chart
- Customizable colors / panel size..
- Customizable view of the peers (e.g. all, active, registered and
favorites.)
- ...

Please post all comments, features requests etc on
http://www.asteriskguru.com/board/viewforum.php?f=1

Zoa.


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Re: [Asterisk-Users] Call Forward - 7940 Asterisk - Help

2005-09-21 Thread Tom Rymes

We use 7940s here via SIP, and CFwdAll on the phones works just fine (firmware 7.5 and 7.4). AFAICT, the only way that user would receive incoming calls on his/her mobile is if incoming calls are sent to his/her desk phone, which would then forward those calls to the mobile. If your user was logged into an incoming call queue that used the RingAll strategy, that user would indeed receive all incoming calls on his/her mobile. Make sure that they have logged out of the queue first!There are obviously a number of other reasons that s/he might be receiving incoming calls, but that is my first guess. Are you using a custom dialplan, [EMAIL PROTECTED]/AMP, or something else?TomOn Sep 16, 2005, at 5:00 AM, c waddy wrote:I am looking for a simple way to forward calls unconditionally with Asterisk. We are running an Asterisk system with 10 extensions using SIP. One of our users leaves the office regulary, when she is out, she needs to be able to forward unconditionally to her mobile or collegue. I am trying to keep it as simple as possible, we use Cisco 7940's, they have a call forward option, when she uses it, "all" our incoming calls go to her mobile? Not just the calls to her extension. My Question: Does Call Forward on the Cisco Phones and Asterisk work? If so do I need to implement something into the dial plan. I have read on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding Is there an updated way to do this? I thought *21* was hard coded into Asterisk? If the Cisco phones wont work, i would like her to simply dial *21*mobile number#, any suggestions on this? Thanks.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [Asterisk-Users] How to retrieve voicemail from an IP phone?


 Does the user need to dial his extension just to retrieve the 
 voicemails or he will dial other number to access those voicemails?

If a user dials 3998, they will be prompted to enter their mailbox
number (extension) and password. If they dial 3999, they will not be
prompted as the callerid parameter passes the extension number to the
voicemail app.

 In the config does it mean that when a user dial 3998 he will be able to 
 retrieve those voicemails? So it means that every users must have a 
 mailbox number for which they will retrive their voicemails? I'm really 
 a newbie. =)

Every valid extension in your extensions.conf file that expects to have
voicemail must have a voicemail box entered in voicemail.conf. So, if you
define extension 299 in extensions.conf, then define 299 in voicemail.conf.

If that user picks up their phone and dials 3999, he will be placed
directly into his voicemail box without any prompts.


 Thanks fo the help,
 --ryan
 
 Rich Adamson wrote:
 
How can I retrieve those voicemails using my ip phone? and how 
 will i confiugre it on asterisk?
 
 Please help I'm very new in asterisk.
 
 
 
 Add something like this in your extensions.conf file:
 
 ; Voicemail access (prompts for exten and password)
 exten = 3998,1,Wait,1
 exten = 3998,2,VoicemailMain
 exten = 3998,3,Hangup
 
 ; Voicemail access (does not prompt for anything)
 exten = 3999,1,Wait,1
 exten = 3999,2,VoicemailMain(s${CALLERIDNUM})
 exten = 3999,3,Hangup
 
 
 
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 -- 
 Ryan Pagquil
 Infodyne Inc. - PhilOnline.com
 3603 Antel Global Corporate Center
 Doña Julia Vargas Ave.
 Ortigas Center Pasig City
 Tel: 687-0715
 Web: www.philonline.com
 
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---End of Original Message-


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Re: [Asterisk-Users] Asterisk PBX

2005-09-21 Thread Tom Rymes


On Wed, 21 Sep 2005, kapil dhawan wrote:



Hi List

I am very new to Asterisk but have been alloted a job to replace my  
traditional PBX with it. Kindly provide me some useful info (PDF's  
etc) to setup Asterisk with FXO and FXS both.


I have to cater some 60 users with 10 simultaneous calls.

Regards


Go to asteriskathome.sourceforge.net and download [EMAIL PROTECTED] Be  
*VERY*** *VERY* *VERY* careful with the CD  
you burn, though, as when booted it will erase your hard drive and  
install CentOS ***WITHOUT WARNING


Other than that, it is a very good way to replace a PBX for an office  
of that size. Of course, you will need to select phones, server  
hardware, PSTN interconnect hardware, etc. as well. Considering how  
important phones are to the average business, you might want to  
consider hiring a consultant (might I recommend cough cough me?)  
to help get you up to speed. It's just a thought, but getting help  
from someone who has already done this might keep you from making a  
few expensive mistakes (ie: buying equipment that is over/ 
underpowered, unreliable, low quality, etc.)


If you choose to do it yourself, [EMAIL PROTECTED], voip-info.org, and this list  
will be your best tools.


Tom
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Re: [Asterisk-Users] Asterisk PBX

2005-09-21 Thread Francesco Peeters

On Wed, September 21, 2005 15:14, Tom Rymes said:
 On Wed, 21 Sep 2005, kapil dhawan wrote:
SNIP
 If you choose to do it yourself, [EMAIL PROTECTED], voip-info.org, and this 
 list
 will be your best tools.

 Tom

I'd like to add Google to that shortlist:

Searchphrase + site:voip-info.org
or
Searchphrase + site:lists.difium.com

will help you quickly search the wiki and list archives...

Good luck!


-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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Re: [Asterisk-Users] Asterisk PBX

2005-09-21 Thread Francesco Peeters

On Wed, September 21, 2005 15:19, Francesco Peeters said:
 On Wed, September 21, 2005 15:14, Tom Rymes said:
 On Wed, 21 Sep 2005, kapil dhawan wrote:
 SNIP
 If you choose to do it yourself, [EMAIL PROTECTED], voip-info.org, and this 
 list
 will be your best tools.

 Tom

 I'd like to add Google to that shortlist:

 Searchphrase + site:voip-info.org
 or
 Searchphrase + site:lists.difium.com

 will help you quickly search the wiki and list archives...

 Good luck!

Oops! Typo!

Searchphrase + site:lists.digium.com

is the correct syntax...

Sorry!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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Re: [Asterisk-Users] Asterisk PBX

2005-09-21 Thread Nathan Pralle


Searchphrase + site:lists.difium.com


The above is good when searching for information on Joe Diffie -- 
Otherwise, you'll want:


Searchphrase + site:lists.digium.com

:)

Nathan


--
-
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Re: [Asterisk-Users] Asterisk PBX

2005-09-21 Thread Francesco Peeters

On Wed, September 21, 2005 15:30, Nathan Pralle said:
 Searchphrase + site:lists.difium.com

 The above is good when searching for information on Joe Diffie --
 Otherwise, you'll want:

 Searchphrase + site:lists.digium.com

 :)

 Nathan


G

I already corrected myself... I canna help them list servers take so long!
 ;-)

---FP
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Re: [Asterisk-Users] iax2 trunking wackyness

2005-09-21 Thread Andrew Kohlsmith

On Wednesday 21 September 2005 07:27, Clive wrote:
 My setup is:   telco-asterisk(voip)-asterisk{ITSP}telco

Are both your asterisk boxes peered to each other?  IIRC trunking ONLY works 
between peers.

-A.
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[Asterisk-Users] Need a test number in NE US that I can call

2005-09-21 Thread Paul


Does anybody here have a verizon land line with caller ID name feature?

I need to call someone to verify the name displayed for a Maine(207) 
number. My LEC claims to have fixed a typo 2 weeks ago. I still don't 
see it changed when calling vonage, voicepulse and broadvoice voip 
DID's. I check those every day.


Email me offlist with a phone number to call if you can help. Thanks.

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[Asterisk-Users] add 0 (zero) to incoming callerID - how?

2005-09-21 Thread Tomasz Chmielewski


I have an asterisk box and SIP / IAX2 phones.

To call out, users have to add 0 (zero) before a real telephone number.

That means, that if they want to call someone that has a number 123456,
they have to call 0-123456.

Simple, right?

This has a serious drawback though - when someone calls us from the
number 123456, we see the callerID 123456, and we're unable to use the
callback/redial feature in the telephone (because the phone doesn't know
that it should add 0 before the number).


So the idea is to manipulate the incoming callerID number, and to add a
0 before it.

This way the telephone user will be able to callback/redial.

How can I manipulate the incoming callerID number (and add 0 before it)?


--
Tomek
http://wpkg.org

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[Asterisk-Users] Packetization period for CODECs

2005-09-21 Thread kurt x

Is it possible in * to set the Packetization period.  For example:  If
I want G711 to be at
10ms.  Is that possible in *?

Thanks,

Kurt
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[Asterisk-Users] permit syntax question

2005-09-21 Thread Min Qiu

Hi,

I'm newbie, please forgive me if this question was asked...

The demo conf has permit=0.0.0.0/0.0.0.0, is the second 
0.0.0.0 is a network mask?  That is, if I need to permit
a /24 prefix, I should do permit=192.168.1.0/255.255.255.0?

Can I do permit=192.168.1.0/24 instead?

If I have multiple prefixes, what is the proper way to list
them?  Should I do permit=192.168.1.0/24,192.168.100.0/24 ?

Thanks a lot,

Min
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[Asterisk-Users] Using *0 to flash an external trunk on bridged channel

2005-09-21 Thread hugolivude

Anybody had success using *0 to flashes an external trunk on bridged channel?

The Asterisk vertical service activation codes section of the Wiki
(http://www.voip-info.org/tiki-index.php?page=Asterisk+vertical+service+activation+codes)
indicates that *0 can be used to flash an external trunk on bridged
channel.  Nothing seems to happen when I use it though.

Do I need to do something special in extensions.conf or perhaps
features.conf to get this to work?

Thanks,
Hugh
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[Asterisk-Users] Ask for config files of Nortell Meridian Op 11 Asterisk for PRI

2005-09-21 Thread Alvaro Parres

Hi list, any one can let me his config files for interconecting a Meridian Op 11 and Asterisk
via a E1 PRI CARD. 

Actually i need the nortell config part, becouse my client nortell provider doesn't know
how to config the PRI card at his part.

Thanks all.


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[Asterisk-Users] maximum concurrent ZAP channels .... max conf ports ...

2005-09-21 Thread Vamsi Pottangi


Hi All,

Is it possible to go beyond 250 concurrent ZAP channels with some
tweaking or workaround ? Meetme uses zap channels, so we could have a
max of 250 conference ports. Is it possible to higher this ?

An Asterisk system can only handle a max. of 250 concurrent ZAP channels. This is due to the design limit (255) within the ZAP channel driver.


Thanks,
~Vamsi
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[Asterisk-Users] HELP: E1 ChannelBank and UniCall

2005-09-21 Thread Paradise Dove

has anybody succeeded  in connecting an E1 CB to asterisk using R2
Digital signalling and Unicall?

any help will be appreciated,
Paradise Dove
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[Asterisk-Users] Cellphones and Asterisk Bluetooth

2005-09-21 Thread chawki hammoud

Hi All:

Is there a script that I can download so I can make
Asterisk send calls through a cellphone?

Regards;
Chawki Hammoud



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Re: [Asterisk-Users] add 0 (zero) to incoming callerID - how?

2005-09-21 Thread Nathan Pralle


Pretty easy stuff.

Something like this:

exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,3,SetCIDNum(0${CALLERIDNUM})
exten = s,4,Dial(Local/105)

Nathan

Tomasz Chmielewski wrote:

I have an asterisk box and SIP / IAX2 phones.

To call out, users have to add 0 (zero) before a real telephone number.

That means, that if they want to call someone that has a number 123456,
they have to call 0-123456.

Simple, right?

This has a serious drawback though - when someone calls us from the
number 123456, we see the callerID 123456, and we're unable to use the
callback/redial feature in the telephone (because the phone doesn't know
that it should add 0 before the number).


So the idea is to manipulate the incoming callerID number, and to add a
0 before it.

This way the telephone user will be able to callback/redial.

How can I manipulate the incoming callerID number (and add 0 before it)?




--
-
Nathan E. Pralle
Give the Director a Serpent Deflector
www.nathanpralle.com
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Re: [Asterisk-Users] iax2 trunking wackyness

2005-09-21 Thread Matt Riddell

Andrew Kohlsmith wrote:
 On Wednesday 21 September 2005 07:27, Clive wrote:
 
My setup is:   telco-asterisk(voip)-asterisk{ITSP}telco
 
 
 Are both your asterisk boxes peered to each other?  IIRC trunking ONLY works 
 between peers.

If you do iax2 show peers in the console, it should show a (T) for trunked
connections.

-- 
Cheers,

Matt Riddell
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[Asterisk-Users] ISDN-forwarding to intern without cost?

2005-09-21 Thread Oliver Rath


Hello *,

I have a little problem here with no idea for solving any more:

I have two cards (AVM-B1 with capi_cm, works fine for callback with 
asterisk and HFC in NT-Mode with zaptel driver, works fine for dialing 
out).


Now i want to forward incoming calls from avm-capi to hfc-zap, but there 
must be same mistake in my config I cannot find. If I make an inbound 
call (e.g. from mobile), a free-tone comes, but the phone doesnt ring.


My hardware:
-
extern-NTBA - AVM-CAPI - Asterisk - HFC-ZAP(NT) - Phone(ISDN)

capi.conf:
-
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

; mode: ptmp (point-to-multipoint) or ptp (point-to-point)
isdnmode=ptmp
; allow incoming calls to this list of MSNs, * == any
context=capi-in
incomingmsn=*
; capi controller number
controller=1
; dialout group
group=1
; enable/disable software dtmf detection, recommended for AVM cards
softdtmf=1
; accountcode to use in CDRs
accountcode=
-
zapata.conf:
-
[channels]

language=de
switchtype = euroisdn
signalling = bri_net_ptmp
pridialplan = dynamic
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00

echocancel=yes
echotraining = 100
echocancelwhenbridged=yes

immediate=no
overlapdial=yes
group=2
context=hfc-intern
channel = 1-2
-
extensions.conf
-
[general]
static=yes
writeprotect=no

[globals]
IAXINFO=guest

; sprungpunkt definiert in capi.conf
[capi-in]
exten = s,1,Dial(Zap/1,60,tT)
exten = s,2,Hangup
.
.
.
[hfc-intern]

exten = _42.,1,Dial(CAPI/contr1/${EXTEN:2}/123456,60,tT)
exten = _42.,2,Hangup
--


Any ideas?

TfH

Oliver


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Re: [Asterisk-Users] add 0 (zero) to incoming callerID - how?

2005-09-21 Thread Matt Riddell

Tomasz Chmielewski wrote:
 How can I manipulate the incoming callerID number (and add 0 before it)?

exten = s,1,Answer()
exten = s,2,SetCIDNum(0${CIDNUM})
exten = s,3,...

-- 
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Matt Riddell
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[Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?

2005-09-21 Thread Brian McEntire

Hello everyone. I'm new to Asterisk but got some basic functionality
going last night and I'm just giddy to have my own PBX ;-)

Sorry if these are silly questions:
 My Asterisk server has the TDM22B (2 FXO, 2 FXS) interface. I
have a very basic PSTN line coming in from the phone company, I tried
to get the most no-frills line possible (didn't pay for caller ID,
voice mail, etc.). I know I can set up voicemail on * on this line. Can
I also get caller ID by virtue of running Asterisk or is that
information stripped out by the phone co before it gets to my box?

Thanks for any advice. I hope to get up to speed on Asterisk and be able to contribute back to the list in time. :)
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[Asterisk-Users] Does Asterisk know if the trunks are busy?

2005-09-21 Thread Steven

I am planning on dabbling with some VOIP providers.
I was thinking of Teliax first.

My thinking is that the first LD call would go to teliax and the second 
(etc.) calls would go out to the PSTN.

I could then verify bandwidth and quality to decide when to add more trunks 
and to Internet connections.

I have been doing some concept testing with FWD for toll free calls, but I 
am using 393 as a trunk access code.

Question:

Will Asterisk know that the one Teliax circuit is in use and use a different 
trunk?
How would I make the dialplan to use a different trunk if the Teliax one is 
busy?

Currently I have:
[outrt-003-dial9]
include = outrt-003-dial9-custom
exten = _9.,1,Macro(hoodahek,${ARG1})
exten = _9.,2,Macro(dialout-trunk,1,${EXTEN:1},)   ;or could be 
Dial(Zap/g1/${EXTEN}) ;exten = _9.,3,Macro(outisbusy) ; No available 
circuits

;Since this is a PRI group, I am not sure how it's internals work. I do not 
see it say channel 1 is busy, channel 2 is busy using channel 3 in the logs.

would I just change it to:
[outrt-003-dial9]
include = outrt-003-dial9-custom
exten = _9.,1,Macro(hoodahek,${ARG1})
exten = _9.,2,Macro(dialout-trunk,4,${EXTEN:1},)   ;or could be 
Dial(IAX2/Teliax1${EXTEN}) ; Will it skip this if it is in use or down?
exten = _9.,3,Macro(dialout-trunk,1,${EXTEN:1},)   ;or could be 
Dial(Zap/g1/${EXTEN})
exten = _9.,4,Macro(outisbusy) ; No available circuits
?

-- 
-- 
Steven

May you have the peace and freedom that come from abandoning all hope of 
having a better past.
----  ---  - - -   -- -   -   --  - - - --- - --   - 
 - --- - - -- -  -- --   -   -- 



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Re: [Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?


Brian McEntire wrote:
  My Asterisk server has the TDM22B (2 FXO, 2 FXS) interface. I have a 
very basic PSTN line coming in from the phone company, I tried to get 
the most no-frills line possible (didn't pay for caller ID, voice mail, 
etc.). I know I can set up voicemail on * on this line. Can I also get 
caller ID by virtue of running Asterisk or is that information stripped 
out by the phone co before it gets to my box?


You need the caller ID service from your telco to get caller ID into 
Asterisk.


Kevin
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Re: [Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?

2005-09-21 Thread Nathan Pralle

etc.). I know I can set up voicemail on * on this line. Can I also get 
caller ID by virtue of running Asterisk or is that information stripped 
out by the phone co before it gets to my box?


That information gets stripped by the telco if you don't pay to have it.

I find it well worth the $8 or so a month I pay to be able to avoid 
calls. :)


Nathan
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Re: [Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?

2005-09-21 Thread John Novack




Brian McEntire wrote:

Hello everyone. I'm new to Asterisk but got some basic functionality 
going last night and I'm just giddy to have my own PBX  ;-)


Sorry if these are silly questions:
  My Asterisk server has the TDM22B (2 FXO, 2 FXS) interface. I have a 
very basic PSTN line coming in from the phone company, I tried to get 
the most no-frills line possible (didn't pay for caller ID, voice 
mail, etc.). I know I can set up voicemail on * on this line. Can I 
also get caller ID by virtue of running Asterisk or is that 
information stripped out by the phone co before it gets to my box?


You will need to pay your provider for that information. It is not sent 
if you don't subscribe


Watch out for some other pitfalls with your FXO-PSTN interface.
Currently there is no dial tone detection, so Asterisk will begin 
dialing as soon as it goes off hook, and you may find it misdials.
There is some question as to which versions of Asterisk allow waits to 
be inserted at the beginning of the dial string, so you may find that 
does or does not mask the problem of mis dialing.
Asterisk also may have a problem knowing when a call is terminated by 
the caller on the PSTN line,


Good luck

John Novack
|
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[Asterisk-Users] qualify=yes

2005-09-21 Thread Manny A. Wise









I was just wondering if I can leave qualify=yes set on all
my trunks??? Originally I had it only setup in one.



Is nice to see the ms reported back..



Is a bad idea? Is that hammering the other servers? Or mine?



Thanks





Manny A. Wise

EL OBELISCO, Inc.

www.calltheus.com

TollFree: (800)230-0106

Tampa: (813)283-0265

Miami: (786)347-5725

NewYork State:
(631)492-3212

Washington Sate: (360)469-0317

FAX: 1.512.597.1779

FWD: 68346








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Re: [Asterisk-Users] Does Asterisk know if the trunks are busy?


Steven wrote:
How would I make the dialplan to use a different trunk if the Teliax one is 
busy?


This is something I'm testing right now.  This is what I use to keep 
track of how many channels are in use on each T1 and failover to 
something else if there is a problem(?).  I'll be using this for T1 and 
for VOIP.  Here's a sample:


exten = _*1XX,1,GotoIf($[ ${GROUP_COUNT(g1)}  22 ]?g2)
exten = _*1XX,n,Set(GROUP()=g1)
exten = _*1XX,n,NoOp(${GROUP_COUNT(g1)})
exten = _*1XX,n,Dial(Zap/g1/${EXTEN:1}|20)
exten = _*1XX,n,NoOp(${DIALSTATUS})
exten = _*1XX,n,GotoIf($[ ${DIALSTATUS} : CONGESTION ]?g2)
exten = _*1XX,n,Hangup

exten = _*1XX,n(g2),GotoIf($[ ${GROUP_COUNT(g2)}  22 ]?g3)
exten = _*1XX,n,NoOp(${GROUP_COUNT(g1)})
exten = _*1XX,n,Set(GROUP()=g2)
exten = _*1XX,n,NoOp(${GROUP_COUNT(g2)})
exten = _*1XX,n,Dial(Zap/g2/${EXTEN:1}|20)
exten = _*1XX,n,GotoIf($[ ${DIALSTATUS} : CONGESTION ]?g3)
exten = _*1XX,n,Hangup

... etc.  The NoOps can be taken out.  It is just a visual for debugging.


Kevin
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[Asterisk-Users] Macro exists if an application returned -1

2005-09-21 Thread Armin Schindler

Hi all,

I use asterisk cvs-head from last week and was wondering why my macros are 
not working like I expected.

A macro will be aborted if one application returns error (-1), but some 
applications (like ChanIsAvail() in my case) do return -1.

So when I want to use ChanIsAvail() in a macro, it does not work if the 
channel is not available, because the macro is not executed on n+101.
The macro just aborts in that case.

Is there any solution for using ChanIsAvail() in a macro?

Thanks,
Armin

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Re: [Asterisk-Users] sipuras 841 bad sound

2005-09-21 Thread Jason Williams

Make sure you have turned off VAD as asterisk does not support Silence supperssion.

Jason
On 9/21/05, Juan Jose Comellas [EMAIL PROTECTED] wrote:
Have you tried upgrading the firmware? I had several problems with theoutbound volume of these phones until I upgraded them.
On Tuesday 20 September 2005 20:46, Anton Krall wrote: Hi Guys! I have a problems with some sipuras 841 and asterisk 1.0.9. Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with
steve's unicall. Everything compiled fine and in fact I can make and receive calls but I have a problem with bad sound when the sipuras call the outside E1's lines. I can listen to the caller without problems but they heard me with a choppy
sound as if you were losing frames or cutting off. Calls between internal sipuras sound good (eventhough the speaker and headset sound comes and goes, for example, when you start talking, seems as if the sipuras takes a
few seconds to catch up with you on volume so the remote user listen to you as if the first words and the last were at low volume and the conversation in the middle sound good, any had that problem?)
So, internal calls sound good between 841's but sound volume is weird at the start and end of a sentence. Calling the outside lines via E1's, I can listen to people without problems but they heard me as choppy or cut off.
Anybody had issues like this? Is it asterisk or the phones or what? Hope you can help Guys, Im really banging my head against the wall here. ___
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[Asterisk-Users] Problem with Queues

I am getting this on the console once people call in 

   -- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing Macro(Local/[EMAIL PROTECTED],2,
sipline|3044) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/3044|20|t) in new stack
-- Called 3044

-- SIP/3044-6a6e is ringing

-- Agent/1001 is ringing

-- SIP/3044-6a6e answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 answered, waiting for
'#' to acknowledge

-- Started music on hold, class 'default', on
Local/[EMAIL PROTECTED],2

-- Stopped music on hold on
Local/[EMAIL PROTECTED],2


Now, I answer, press # on my phone to acknowledge the
call and it goes back again.Am I doing something
wrong?





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Re: [Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?

2005-09-21 Thread Brian McEntire

Okay, thanks all for the feedback!

I accidentally sent this response to one person who replied rather than to the list so I'm going post it here too:

I didn't go into full details in my message. Part of the
reason of setting up Asterisk is because I recently signed up with VOIP
service and I'm extremely happy with it. I'm only keeping the PSTN line
for emergencies and because my local number currently isn't portable.



Perhaps I'll be able to make * play a message to incoming PSTN callers
announcing my new VOIP number before ringing the inside extensions?
This could help speed the transition to almost purely VOIP, and at some
later date I could dump incoming PSTN calls to VM or just reject them
altogether.



I do have lots of services on the VOIP line, callerID included, and so I should be able to use call filtering on that line.

- - -

Thanks for the tip about dial tone sensing. I made a couple test calls
last night and things went smoothly, but if it's an intermittent
problem, I'll watch for it. I'm running version 1.2 beta from CVS.
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[Asterisk-Users] Addendum to Problem with Queues question

Here is the full transaction

-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing Macro(Local/[EMAIL PROTECTED],2,
sipline|3044) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/3044|20|t) in new stack
-- Called 3044

-- SIP/3044-ea92 is ringing

-- Agent/1001 is ringing

-- SIP/3044-ea92 answered Local/[EMAIL PROTECTED],2

-- Local/[EMAIL PROTECTED],1 answered, waiting for
'#' to acknowledge

-- Started music on hold, class 'default', on
Local/[EMAIL PROTECTED],2

-- Unable to find extension '' in context
'crystal-sip'
-- Playing 'pbx-invalid' (language 'en')

Sep 21 10:30:30 WARNING[52987]: file.c:550
ast_readaudio_callback: Failed to write frame
-- Stopped music on hold on
Local/[EMAIL PROTECTED],2

Sep 21 10:30:30 WARNING[52987]: res_features.c:450
ast_bridge_call: Bridge failed on channels
Local/[EMAIL PROTECTED],2 and SIP/3044-ea92
  == Spawn extension (macro-sipline, s, 1) exited
non-zero

Why doesn't ast_bridge_call do it's thing 





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[Asterisk-Users] HOWTO: A simple AGI application to modify incomi ng CallerID on the fly using SQL Server and not UnixODBC

2005-09-21 Thread Colin Anderson

Requirements: 

1. http://sourceforge.net/projects/odbcsock

2. The SQL server must be in the same subnet as your * server

Howto:

1. Install ODBCSocketServer on your SQL server and verify connection with
the included VB COM app from a Windows box.

2. Decide how you want to transform the Caller ID. In my case, I want to do
a lookup of the CallerID number in SQL server and prepend the Caller ID with
a job number which is a unique ID we assign to each customer.

3. In the case of a large database with lots of fields it's a good idea to
create a view in SQL server that has *only* the records you want, then you
can filter from there. Note the view name.

4. Create an ODBC system DSN on the SQL server that points to your SQL
server DB

5. Modify the following PHP script to your taste. Ensure your PHP.ini in
/etc has error and warning suppression ON or else the AGI will return
invalid characters:

#!/usr/bin/php -q
?php

 class ODBCSocketServer {
var $sHostName; //name of the host to connect to
var $nPort; //port to connect to
var $sConnectionString; //connection string to use

//function to parse the SQL 
 function ExecSQL($sSQL) {

$fToOpen = fsockopen($this-sHostName, $this-nPort,
$errno, $errstr, 30);
if (!$fToOpen)
{
//contruct error string to return
$sReturn = ?xml
version=\1.0\?\r\nresult
state=\failure\\r\nerror$errstr/error\r\n/result\r\n;
}
else
{
//construct XML to send
//search and replace HTML chars in SQL first
$sSQL = HTMLSpecialChars($sSQL);
$sSend = ?xml
version=\1.0\?\r\nrequest\r\nconnectionstring$this-sConnectionString
/connectionstring\r\nsql$sSQL/sql\r\n/request\r\n;
//write request 
fputs($fToOpen, $sSend);
//now read response
while (!feof($fToOpen))
{
$sReturn = $sReturn .
fgets($fToOpen, 128);
}
fclose($fToOpen);
}
return $sReturn;
}
 }//class

//Here is the code that uses this class.  First we create the class
$oTest = new ODBCSocketServer;

//Set the Hostname, port, and connection string
  $oTest-sHostName = 192.168.1.17;
$oTest-nPort = 9628;
$oTest-sConnectionString = DSN=intranet;UID=sa;PWD=12345;;

//It is bad practice to use the SA account; in sane installations you use a
non-priviledge elevated user
//now exec the SQL
$sResult = $oTest-ExecSQL(SELECT * FROM AsteriskCallerID where
homephonecd like '.$argv[1].'); 

//the $argv[x] variable array contains any arguments you pass to the script.
The array is 
// $argv[0] the script itself, $argv[1] the first argument, $argv[2] the
second, etc.

$p = xml_parser_create();
xml_parse_into_struct($p, $sResult, $vals, $index);

//xml_parse_into_struct takes the returned XML and parses it out into a
variable array

xml_parser_free($p); //clean up

//the print statement returns your SQL data to Asterisk using the SET
VARIABLE statement. 
//$vals[x][value] returns the contents of the variable based on it's
variable index, you will
//have to play with the index to determine which one has the data you want
to return

print SET VARIABLE LANDMARKCID \.$vals[4][value];
print  .$vals[2][value].:\;

?

6. Rename this script to transformcallerid.agi and drop it into
/var/lib/asterisk/agi-bin,  chmod 755 it.

7. Modify your dialplan with the AGI script called in a distinct context and
that context returns control to the original context when done.

[my-inbound-context-with-did]

'Assumption here is that you have several DID's in this context but should
be able to work without a DID

exten = 3078,1,SetVar(CURRENTEXTEN=3078)'Set a variable with the current
extension being processed
exten = 3078,2,Goto(Transform-CallerID,s,1))
exten = 3078,3,DoRegualarDialplanStuffHere

[Transform-CallerID]

exten = s,1,SetVar(CIDPREFIX=${CALLERIDNUM:3:3})
exten = s,2,SetVar(CIDSUFFIX=${CALLERIDNUM:6:10})
exten = s,3,agi(transformcallerid.agi|${CIDPREFIX}-${CIDSUFFIX}) 'We keep
phone numbers in XXX- format
exten = s,4,NoOp(${LANDMARKCID}) 'Display the returned variable for
debugging purposes
exten = s,5,Gotoif($[${LANDMARKCID} =  : ]?6:7) 'If the AGI didn't fond
a match in the database
exten = s,6,SetVar(LANDMARKCID=UNKNOWN:) 'Prepend the caller ID with
UNKNOWN:
exten = s,7,SetCallerID(${LANDMARKCID}${CALLERIDNUM:3:10}) 'Otherwise

Re: [Asterisk-Users] ISDN-forwarding to intern without cost?

2005-09-21 Thread Joerg Lauer


Hi,

I think I had the same problem and I think the error was that the dial 
statement had to be:


exten = s,1,Dial(Zap/1/,60,tT)

I may remeber wrong, though.

Btw: It may be a better idea to use Zap/g2/,60,tT), this way both 
B-Channels of the HFC card may be used.


Joerg
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[Asterisk-Users] is possible connect?

2005-09-21 Thread Pablo Allietti

hi all i have this structure.


Box.(te110p)Pbx(e1)4 analogic lines to outside


is poosible connect asterisk to get outside lines? because i can call
any extension in my pbx with xten but i cant get outside lines. the
asterisk tellme all circuits are busy when i send the number 9 to get
the line. i remove all in my extensions.conf and have this

[EMAIL PROTECTED] asterisk]# cat extensions.conf
[GLobals]
;   RECEPTIONIST=Zap/1
JOHN=SIP/203
MARY=SIP/202
LOCALTRUNK=Zap/1

[incoming]
exten = s,1,Answer()
exten = s,2,Background(current-movies)
exten = s,3,Hangup()
exten = 1,1,Playback(movie1)
exten = 1,2,Goto(incoming,s,1)
exten = 2,1,Playback(movie2)
exten = 2,2,Goto(incoming,s,1)
exten = 0,1,Dial(${RECEPTIONIST})

[internal]
;   ignorepat = 9
exten = _1XX,1,Dial(${LOCALTRUNK}/${EXTEN})
exten = _1XX,2,Voicemail(u${EXTEN})
exten = _,1,Dial(${LOCALTRUNK}/${EXTEN})
exten = _,2,Playback(invalid)
exten = _,3,Hangup


my extensions in my pbx start with the number 1 and i need for example
to call 9,856 is a service number in my country.

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Re: [Asterisk-Users] Macro exists if an application returned -1

2005-09-21 Thread Thorsten Lockert


On Sep 21, 2005, at 8:27 , Armin Schindler wrote:

Is there any solution for using ChanIsAvail() in a macro?


Yes.  Fix app_chanisavail.c such that it says if (ast_goto_if_exists 
(... instead of if (!ast_goto_if_exists(  Somone bungled the  
conversion to use ast_goto_if_exists -- this may or may not be  
incorrect in other places as well.


Thorsten

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[Asterisk-Users] Help with asterisk-oh323 driver

2005-09-21 Thread Fernando Herrera






  
  DV,
  
  Have you solved 
  this? I am facing the same problem. I am running Asterisk 1.0.9 and outgoing 
  TCS does not show the 
  receiveRTPAudioTelephonyEventCapability.
  
  Kind regards, 
  
  
  Fernando 
  Herrera
  
  
  
  
  
Hi all,

Sorry if this has been answered previously, but I have not had any
luck trying to find it.

I am trying to connect my Asterisk server (1.0 stable, Fedora Core 2,
kernel 2.6.8-1.521) to connect to a gateway that can only support
H323. I have installed the asterisk-oh323 channel driver (version
0.6.3b) using Open H323 1.13.5 (patched as per asterisk-oh323's
instructions) and PWLIB 1.6.6. This is all working fine for very basic
call setup and tear down, from any of my SCCP, SIP, H323 or POTS
(X100P card) phones.

NB: The gateway only handles signalling, so all media will flow
between the endpoints and the gateway will handle signalling to the
receiving gateway, as such (excuse the dodgy diagram :) ):

-[Gateway]---
|  |
(H323)(H323 or MGCP/ISUP)
|  |
   V V
[Asterisk]---(RTP)--[Terminating gateway]
   |
(Signalling + RTP)
   |
(Zaptel/SIP/H323/SCCP phones)


There are some requirements for me to connect to this switch:

1. I must support H245 tunneling and faststart (working fine)
2. I must dynamically negotiate the codecs (i.e. send multiple codecs
as part of the faststart and the softswitch will decide which of the
codecs to use based on the terminating gateway's capabilities). The
codec picked will be passed back in the return faststart from the
gateway.
3. It must support RFC2833 for OOB DTMF.

The problems I am facing are that my faststart in my setup messages
only ever has one codec, regardless of what I have set in the [codecs]
section of oh323.conf, and even if I specify userInputMode=RFC2833 in
oh323.conf my TCS does not include the capability
receiveRTPAudioTelephonyEventCapability hence RFC2833 is never
neogitated. I'm sure this is just a minor tweak of the source code,
but not being an expert in C I am having problems figuring out what
needs to be done and where.

Any help on this matter would be appreciated.

Cheers
DV

  
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Re: [Asterisk-Users] Problem with Queues


Crystal Stream, Incorporated wrote:
I am getting this on the console once people call in 


   -- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing Macro(Local/[EMAIL PROTECTED],2,
sipline|3044) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/3044|20|t) in new stack
-- Called 3044

-- SIP/3044-6a6e is ringing

-- Agent/1001 is ringing

-- SIP/3044-6a6e answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 answered, waiting for
'#' to acknowledge

-- Started music on hold, class 'default', on
Local/[EMAIL PROTECTED],2

-- Stopped music on hold on
Local/[EMAIL PROTECTED],2


Now, I answer, press # on my phone to acknowledge the
call and it goes back again.Am I doing something
wrong?


You mean, it hangs up and calls you back again?  Sounds like you have 
option h on Queue and have # set to hangup in features.conf.



Kevin
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[Asterisk-Users] ODBC Voicemail WEB Retrieval V1.1

2005-09-21 Thread pbx

Hi All.

After some input, I created a V1.1 version of my ODBC VM retrieval from
the ODBC_Storage

It now uses either Mysql or unixODBC drivers to connect to the database

I didn't have php compiled with unixODBC so i had to recompile it in

./configure --with-unixODBC --with-mysql --with-apxs2=./blah/blah/blah
make
make install

after all that it worked.

see the Readme and changelog

http://www.itsngroup.com/software/asterisk/downloads/

The older version (1.0) only had MySQL support.

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Re: [Asterisk-Users] Addendum to Problem with Queues question


Crystal Stream, Incorporated wrote:

Here is the full transaction

-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing Macro(Local/[EMAIL PROTECTED],2,
sipline|3044) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/3044|20|t) in new stack
-- Called 3044

-- SIP/3044-ea92 is ringing

-- Agent/1001 is ringing

-- SIP/3044-ea92 answered Local/[EMAIL PROTECTED],2

-- Local/[EMAIL PROTECTED],1 answered, waiting for
'#' to acknowledge

-- Started music on hold, class 'default', on
Local/[EMAIL PROTECTED],2

-- Unable to find extension '' in context
'crystal-sip'

This must be the problem.  Not sure where that is coming from.


-- Playing 'pbx-invalid' (language 'en')

Sep 21 10:30:30 WARNING[52987]: file.c:550
ast_readaudio_callback: Failed to write frame
-- Stopped music on hold on
Local/[EMAIL PROTECTED],2

Sep 21 10:30:30 WARNING[52987]: res_features.c:450
ast_bridge_call: Bridge failed on channels
Local/[EMAIL PROTECTED],2 and SIP/3044-ea92
  == Spawn extension (macro-sipline, s, 1) exited
non-zero


It sounds like you may need to post the relevant extensions.


Kevin
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RE: [Asterisk-Users] sipuras 841 bad sound

2005-09-21 Thread Anton Krall




I think I got it but just to be sure, where do I find that 
setting on sipura 841?

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Jason 
  WilliamsSent: Miércoles, 21 de Septiembre de 2005 10:22 
  a.m.To: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [Asterisk-Users] sipuras 841 bad 
  sound
  
  Make sure you have turned off VAD as asterisk does not support Silence 
  supperssion.
  
  
  Jason
  On 9/21/05, Juan Jose 
  Comellas [EMAIL PROTECTED] 
  wrote: 
  Have 
you tried upgrading the firmware? I had several problems with 
theoutbound volume of these phones until I upgraded them. On 
Tuesday 20 September 2005 20:46, Anton Krall wrote: Hi 
Guys! I have a problems with some sipuras 841 and asterisk 
1.0.9. Im using 841 with asterisk 1.0.9 with a digium card 
(single e1 span) with  steve's unicall. Everything 
compiled fine and in fact I can make and receive calls but I have a 
problem with bad sound when the sipuras call the outside E1's lines. 
I can listen to the caller without problems but they heard me with a choppy 
 sound as if you were losing frames or cutting off. Calls between 
internal sipuras sound good (eventhough the speaker and headset 
sound comes and goes, for example, when you start talking, seems as 
if the sipuras takes a  few seconds to catch up with you on volume 
so the remote user listen to you as if the first words and the last 
were at low volume and the conversation in the middle sound good, 
any had that problem?)  So, internal calls sound good 
between 841's but sound volume is weird at the start and end of a 
sentence. Calling the outside lines via E1's, I can listen to people 
without problems but they heard me as choppy or cut off. 
 Anybody had issues like this? Is it asterisk or the phones 
or what? Hope you can help Guys, Im really banging my head 
against the wall here. 
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Re: [Asterisk-Users] Problem with Queues

features.conf is devoid of #
the queue doesn't have h in it.
only have tT



--- Kevin Bockman [EMAIL PROTECTED] wrote:

 Crystal Stream, Incorporated wrote:
  I am getting this on the console once people call
 in 
  
 -- outgoing agentcall, to agent '1001', on
  'Local/[EMAIL PROTECTED],1'
  -- Called Agent/1001
  -- Executing Macro(Local/[EMAIL PROTECTED],2,
  sipline|3044) in new stack
  -- Executing Dial(Local/[EMAIL PROTECTED],2,
  SIP/3044|20|t) in new stack
  -- Called 3044
  
  -- SIP/3044-6a6e is ringing
  
  -- Agent/1001 is ringing
  
  -- SIP/3044-6a6e answered
 Local/[EMAIL PROTECTED],2
  -- Local/[EMAIL PROTECTED],1 answered, waiting
 for
  '#' to acknowledge
  
  -- Started music on hold, class 'default', on
  Local/[EMAIL PROTECTED],2
  
  -- Stopped music on hold on
  Local/[EMAIL PROTECTED],2
  
  
  Now, I answer, press # on my phone to acknowledge
 the
  call and it goes back again.Am I doing
 something
  wrong?
 
 You mean, it hangs up and calls you back again? 
 Sounds like you have 
 option h on Queue and have # set to hangup in
 features.conf.
 
 
 Kevin
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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

2005-09-21 Thread Matt Roth

All,

This message has generated a lot of responses, so I'm going to address
each of them here in an attempt to consolidate the thread.

Matt,

- I'm very interested in the specifics of your setup.
- How much space is on the RAM disk?
Currently it is 10 GB. We are upgrading it to 16 GB.

- What kind of RAM drive is it?
The current RAM is 12 GB DDR2 400 MHz (12 x 1GB) Single Ranked DIMMs.
The details for each 1 GB DIMM can be seen here:

http://www.samsung.com/Products/Semiconductor/DRAM/DDR2SDRAM/DDR2SDRAMmodule/RegisteredDIMM/M393T2950CZ3/M393T2950CZ3.htm

The upgrade will involve adding 2 GB DIMMs to the system, but I don't
have the details on these yet.

The RAM disk is setup by adding the following kernel command line option
to grub.conf:

ramdisk_size=10485760

We are running Fedora Core 3 with the most up to date 2.6 SMP kernel.

By default the RAM disk's block size is 1024 bytes, so we are formatting
it as an ext2 file system with a block size of 1024 bytes using the
following command:

mke2fs -b 1024 -m 0 /dev/ram0

The block size can easily be changed from the kernel's view (using the
kernel command line option ramdisk_blocksize=) or from mke2fs's view
(using the -b argument), so please let me know if I can make an
easy optimization here.

Finally, the RAM disk is mounted using the command:

mount /dev/ram0 /digrec

A good RAMDISK howto exists at:

http://www.vanemery.com/Linux/Ramdisk/ramdisk.html

- What format are you recording to?
- What codec are the SIP calls being placed over?
We are recording to the PCM format and using the G711 uLaw codec. High
voice quality is essential to our application (we are a call center) so
we partnered with MCI to configure our network for the required
bandwidth and chose the highest quality, zero compression codec. We
noload all other codecs in order to avoid transcoding on the switch, so
we must record to PCM. Later (on a separate server) the recordings are
mixed to GSM which provides a 5 to 1 compression ratio with very little
artifacts.

- freak out if a 10ms delay occurs.
Are you saying that these messages indicate a gap in a digital
recording? If so, what is the duration of the gap? If it's comparable
to a CD skip, I think we can deal with it until a buffer or another
solution is implemented.

- Good luck and please keep us updated on your progress
Thank you. I'll be keeping the list informed of our progress.

Zoa,

- I suppose you are the person from the digium forum
That was actually my boss's boss. We thank you all the way up and down
the line for your suggestion.

- The reason i recommended you to use a ramdisk is because i think the
- problem with recording to disk is saving 20ms of stream 1, then 20 ms of
- stream 2, then 20ms of stream 3 etc etc meaning you write everytime
- very small things. (with a lot of seeking).
Agreed. This is why we hope that decoupling the copy (memory to disk)
from Asterisk itself and, more importantly, Asterisk's real-time
handling of the call being recorded will be sufficient.

For the record, when recording 512 simultaneous calls to the local disk
we saw a peek of about 13,000 blocks written per second.

- Our best test results were with:
-
- - buffering the recordings to a ramdisk, then
We're doing that, as per your suggestion.

- - on low load (at night) copy the files over the network (easy to shape
- the pipe, so that you dont overload anything), This way, the memory
- buffer will take care of the 'fragmentation' and not your harddisk.
If you'll note the format of the recordings and that we'll be recording
up to 200,000 minutes of calls a day, with a little quick math you'll
realize that it would take 80 to 100 GBs of memory for us to buffer a
full day's recordings. Combined with the fact that a server failure
late in the day would cause us to lose them all, this is not a desirable
solution.

Instead, we plan to write an application to call from the MONITOR_EXEC
hook that will be executed at the end of each call. This application
will be niced down to the lowest priority, and simply copy the leg files
from memory to disk. Under normal conditions (ie. our NFS server is up)
this will actually be a copy to a remote disk using an asynchronous NFS
transfer. All actual disk I/O will occur on our digital recording
server and any handling of the digital recordings will occur only after
the call they are bound to is completed.

Do you have any suggestions regarding this

Re: [Asterisk-Users] Macro exists if an application returned -1

2005-09-21 Thread Armin Schindler

On Wed, 21 Sep 2005, Thorsten Lockert wrote:
 On Sep 21, 2005, at 8:27 , Armin Schindler wrote:
  Is there any solution for using ChanIsAvail() in a macro?
 
 Yes.  Fix app_chanisavail.c such that it says if (ast_goto_if_exists(...
 instead of if (!ast_goto_if_exists(  Somone bungled the conversion to
 use ast_goto_if_exists -- this may or may not be incorrect in other places
 as well.

Ah yes, thanks !

Is this already known to developers or why is not fixed in CVS yet?

Armin

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Re: [Asterisk-Users] presence settings and Eyebeam

2005-09-21 Thread Kevin Hanson


Olle E. Johansson wrote:


Vahan Yerkanian wrote:
 


What is the proper way of adding hints to multiple extensions?


In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence

exten = 1234,hint,SIP/1234 works,

exten = _1,hint,SIP/${EXTEN} doesn't. Not sure if I can even use
${EXTEN} here...

Any hints?
   


File a bug report if it does not work. I think it would be a good idea
if it works, even though I usually don't recommend using the extension
as the peer name. ;-)

/O

 

Can you elaborate on why you don't recommend using the extension as the 
peer name?


Cheers,
Kevin

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RE: [Asterisk-Users] Complete NPA-NXX list for USA/Canada npanxx,

2005-09-21 Thread Damon Estep

 
  On a related note, I wanted our phones to display city, st for the
  caller-ID name in the event that none was provided.
 
 Interesting code.  What sort of memory does * take up when you load up
 all those CLID values?
 
 Nathan
 
I am a little late to this thread, but the answer is WAY TO MUCH.

With 150,000 pattern match extensions * takes a very long time to
reload, during which time calls do not proceed.

If you use Realtime MySQL it pulls in ALL patter match extensions in the
context on every call (150,000 rows per query).

There are two ways to fix this;

The one we did, use the application command realtime() to pull the
record from a database based on napnxx and then use gotoif to route to
the lowest cost provider in that records (realtime must be used on a
unique index so ONLY 1 row is retruned). We are testing upgrading this
to mysql 5 where a view could be used to eliminate the gotoif. With the
gotoif and 2 carriers per npa nxx it is fast. Realtime() can only do
simple queries as of right now, so views would be a huge plus.

The other option I know others are using is to get the route via an agi
script.

Bottom line, YOU CAN NOT load all 150,000 NPA/NXX pattern matches in
asterisk via text file or realtime and expect acceptable performance,
YOU MUST use a database query solution to get only info you need to *

BTW - for $5/mo you can have access to a reasonably well maintained
complete listing of NPA/NXX/OCN/CLLI/ADDRESS, ETC.

https://www.telcodata.us/secure/account/signup (no affiliation)

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RE: [Asterisk-Users] Asterisk in Spanish

2005-09-21 Thread Sergio Serrano

 

Try in www.asterisk-es.org

-Mensaje original-
De: Sebastian Milioto [mailto:[EMAIL PROTECTED] 
Enviado el: lunes, 19 de septiembre de 2005 15:08
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] Asterisk in Spanish

Hi all,

I've been installing [EMAIL PROTECTED] and (of course) all the answering
machine (I don't sure that's the right word in english, preatendedora in
spanish) speech is in enlgish languaje.
Is there anyway to download all those .gsm files speaked in spanish?
Or may be another site which contain this kind of stuff (.wav, .gsm files
for answering machines in spanish)?


Thank you very much,

Regards,

Sebastian Milioto
Telecommunications Engineer
IM: [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
Mobile: 549 3571 543658
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Re: [Asterisk-Users] VM low volume - testers needed

2005-09-21 Thread Patrick

On Mon, 2005-09-19 at 10:38 -0600, Rich Adamson wrote:
 For those that have experienced low VM recording volumes when using
 a Digium TDM04b (or similar analog pstn card), a work around has been
 committed to cvs-head.

Does this mean that tracking down the cause of the low volume issue was
not successful or was the workaround an easier solution?

Regards,
Patrick
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Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?

2005-09-21 Thread Tony Hoyle


Matthew Boehm wrote:

I've been trying to diagnose why my server has a constant idle time of 90%
even when nothing is running.

After finally discovering what hi means in 'top' (it means hardware
interrupts) I find that this percentage always averages around 7-10%.

How can I find out what is causing this constant load of interrupts?


cat /proc/interrupts

One of your devices will have a *very* large number of interrupts I expect.

Tony
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Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?


 I've been trying to diagnose why my server has a constant idle time of 90%
 even when nothing is running.
 
 After finally discovering what hi means in 'top' (it means hardware
 interrupts) I find that this percentage always averages around 7-10%.
 
 How can I find out what is causing this constant load of interrupts?
 
 I have a Dell 1850 3.0Ghz with on board RAID and 2GB RAM.
 
 Anyone else experiencing this?

Do a ps ax |grep mpg and see if you have more then two mpg's running.
If you do, stop asterisk, kill all remaining mpg's, and restart asterisk.


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[Asterisk-Users] Pinging ...

2005-09-21 Thread Alan Bunch

Ok, if I missed something in the wiki please point me there with the 
correct search terms.


Asterisk 1.0.7  (AAH really)

4 co lines from Bellsouth into a Diguim T400P.

Polycom 501 x 4 on the desktops.

My problem is on calls to or from the CO I hear a pinging (thing sonar 
ping in a submarine) every 12 seconds.  You can set your watch to it.  
COuld this be a call recording in progress tone.  I have not made any 
effort to turn on call recording.  If it is I'll go figure out how to 
turn it off.  Could this be echo cancelation retraining  Am I just 
u8nder dosing on my morning meds ?


Any guesses from anyone here. I just want the pinging to stop.

One more thing, this machine is remotely managed and is in a distant 
city.  If I really have to go I can.


Thank in advance
Alan
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RE: [Asterisk-Users] SIP audio port usage

2005-09-21 Thread Sherwood McGowan

It depends on the ATA, and our router, etc... Typically in the range between
1 and 2 

--Original Message-
-From: [EMAIL PROTECTED] 
-[mailto:[EMAIL PROTECTED] On Behalf Of 
-Adrien Laurent
-Sent: Monday, September 19, 2005 12:23 PM
-To: asterisk-users@lists.digium.com
-Subject: [Asterisk-Users] SIP audio port usage
-
-Hi,
-
-I know that SIP is using port 5060 for session initiation, 
-but which port does it use for audio ? is it dynamically assigned ?
-
-Thanks,
-
-Adrien
-
---
-Adrien Laurent - CIO
-www.modulis.ca
-514-284-2020 ext 202
-[EMAIL PROTECTED]
-
-
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[Asterisk-Users] hints and the sNOM 360

2005-09-21 Thread Paul Hewlett

Hi 

   I am trying to get a SNOM 360 to monitor other extensions i.e. when someone 
makes a call to/from another extension, one of the LED's on the SNOM 360 will 
change state. I am using 1.0.9/bristuff-8l.

   I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running 
the latest firmware. (3.60i for 190, 4.0 for 360). I have read all the 
relevant articles on the wiki on 'hints' and also on the 'devstate' app.

   I set the first function key on the 360 to extension 2001 - this transforms 
itself into sip:[EMAIL PROTECTED];user=phone when I save it. The function key 
type is set to 'Destination' as recommended by a number of articles on the 
Wiki. 

aside
This seems to contradict the 360 manual which states that the function key 
type should be set to 'Line'. /aside

In the dialplan I put

[myhints]
exten = 2001,hint,sip/2001
exten = 2001,1,macro(stdexten,sip/2001)
exten = 2001,2,hangup

In sip.conf I have

[2001]
type=friend
username=2001
subscribecontext=myhints
host=dynamic
mailbox=2001
callerid=ext 2001
incominglimit=1

[2002]
type=friend
username=2002
subscribecontext=myhints
host=dynamic
mailbox=2001
callerid=SNOM360 2002

I restart asterisk from scratch and then reboot the 360. The * console
shows one entry when typing the command 'sip show subscriptions' which looks 
correct. Inspection of the sip trace log on the 360's web page reveals that 
the registration  succeeds and that the subscription of the 2001 from the 360 
also gets a 200 OK reply. However when I dial into extension 2001 nothing 
happens to the led's on the 360. Inspection of the 'sip trace log' on the 
360's web page reveals that it does not receive any NOTIFY from asterisk.

I am at my wits end - anybody got any ideas ?

Paul HE
~
-- 
Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za
Tel: +27 21 852 8812  Cel: +27 84 420 9282  Fax: +27 86 672 0563
-- 
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Re: [Asterisk-Users] VM low volume - testers needed

2005-09-21 Thread Andrew Kohlsmith

On Monday 19 September 2005 12:38, Rich Adamson wrote:
 The g(6) adds a 6 db gain for zap calls that end up recording a Voicemail
 message.
...

 * 'g(#)' the specified amount of gain will be requested during message
  recording (units are whole-number decibels (dB))

How in the hell does that make any sense?  are your normal incoming calls 
quiet too or just voicemail?  

 In my case, the asterisk box is located about 7db from the central
 office. As noted in bug 2023 (and 2022), calls from an outside pstn
 line coming into asterisk incure a 7db pstn loss (which can't be adjusted
 for with rxgain and txgain as changing those values to something
 reasonable generates echo).  Retrieving that VM message from an outside
 location creates another 7db loss (now -14db down in total), making it
 very difficult (if not impossible) to hear the message. (And, yes I've
 gone through all the recommendations with wav vs gsm files, etc.)

I am not sure I understand why the txgain/rxgain isn't fixing it without 
adding unacceptable echo...  this all seems very odd...  I mean for a test 
you should be able to dial an echo() application and have extremely quiet 
echoed audio... is this the case?

 The work around only kicks in if the call comes from a zap channel
 and ends up in voicemail, adding a 6db gain to that recorded message.
 No other channel types are impacted by this new parameter.

This is a HELL of a band-aid.

-A.
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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

2005-09-21 Thread Matt Hess

In light of the I/O bottleneck problem I'd have to ask why asterisk
can't just buffer incoming audio and then flush a complete audio file to
disk.. I'm assuming that recordings vary in length.. the problem with
this idea is what happens if 50 recordings all complete at the same
time.. a dump like that might not be very pretty (a fast drive plus a
little scheduler or limiter so that only x number of files get written
to disk at a time would probably help out there) but I can imagine that
a single file being written is much more efficient and more
disk-friendly.. perhaps I don't know what the heck I'm talking about but
at least in my mind the theory sounds better than the current
'stream-to-file' method employed by asterisk.

Matt Roth wrote:

All,

This message has generated a lot of responses, so I'm going to address
each of them here in an attempt to consolidate the thread.

Matt,

- I'm very interested in the specifics of your setup.
- How much space is on the RAM disk?
Currently it is 10 GB. We are upgrading it to 16 GB.

- What kind of RAM drive is it?
The current RAM is 12 GB DDR2 400 MHz (12 x 1GB) Single Ranked DIMMs.
The details for each 1 GB DIMM can be seen here:

http://www.samsung.com/Products/Semiconductor/DRAM/DDR2SDRAM/DDR2SDRAMmodule/RegisteredDIMM/M393T2950CZ3/M393T2950CZ3.htm

The upgrade will involve adding 2 GB DIMMs to the system, but I don't
have the details on these yet.

The RAM disk is setup by adding the following kernel command line option
to grub.conf:

ramdisk_size=10485760

We are running Fedora Core 3 with the most up to date 2.6 SMP kernel.

By default the RAM disk's block size is 1024 bytes, so we are formatting
it as an ext2 file system with a block size of 1024 bytes using the
following command:

mke2fs -b 1024 -m 0 /dev/ram0

Finally, the RAM disk is mounted using the command:

mount /dev/ram0 /digrec

A good RAMDISK howto exists at:

http://www.vanemery.com/Linux/Ramdisk/ramdisk.html

- Good luck and please keep us updated on your progress
Thank you. I'll be keeping the list informed of our progress.

Zoa,

- I suppose you are the person from the digium forum
That was actually my boss's boss. We thank you all the way up and down
the line for your suggestion.

For the record, when recording 512 simultaneous calls to the local disk
we saw a peek of about 13,000 blocks written per second.

- Our best test results were with:
-
- - buffering the recordings to a ramdisk, then
We're doing that, as per your suggestion.

[Asterisk-Users] Weird Over Lapping Asterisk Calls via SIP Phones

2005-09-21 Thread asterisk

I am trying to create an IVR system that uses both POTS and IP phones
and I have a few problems that I encountered with the IP SIP phones
(Grandstream Budge Tone 102).

1. When a user hits the hook fast enough, the user can create multiple
IVR connections that gives the appearance of an echo that is phased a
few seconds apart. The way to reproduce this is by hitting the hook
fast and furious. The telltale sign of multiple connections is when you
get an echo from the IVR and or static caused by the multiple
prompting. What could be causing this and how do I solve this. Here is
the SIP config.

2. Poor audio quality over VPN broadband and direct Full T1. The audio
recordings (in uncompress WAV format) that comes out at the IVR end is
either muddled, static, or contains gaps. This occurs on both broadband
and T1. What can be done to improve the audio recordings so they are
audible.

3. How do I set up QOS on a broadband VPN when I do not control the
bandwidth of the Internet?






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[Asterisk-Users] How can i call to a cellphone here in Mexico?

2005-09-21 Thread Claudio Canseco

Hi,
I've been trying to dial out to a cellphone, but all my calls get redirected to 066 (the emergency number at my city, like 911)
does anyone know how to fix this, any ideas,?
does anyone from mexico has done this?

Any comment will be highly appreciated,

Regards,
Claudio
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RE: [Asterisk-Users] iax2 trunking wackyness

2005-09-21 Thread Adam Robins

I have two Asterisk boxes that I thought were trunked, but based on not
seeing the (T) in iax2 show peers, now I'm not sure.

Server 192.168.xxx.1 extensions.conf has:
Exten = _2XXX,1,Dial(IAX2/interoffice:[EMAIL PROTECTED]/${EXTEN})

Server 192.168.xxx.1 iax.conf has:
[general]
trunk=yes
[interoffice]
type=friend
host=dynamic
context=extensions
secret=password
disallow=all
allow=g729

Server 192.168.xxx.2 extensions.conf has:
Exten = _3XXX,1,Dial(IAX2/interoffice:[EMAIL PROTECTED]/${EXTEN})

Server 192.168.xxx.2 iax.conf has:
[general]
trunk=yes
[interoffice]
type=friend
host=dynamic
context=extensions
secret=password
disallow=all
allow=g729

Should I plug in the actual IP addresses instead of host=dynamic?  Also,
I do not currently have register statements.
In iax.conf for these.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Wednesday, September 21, 2005 10:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iax2 trunking wackyness

Andrew Kohlsmith wrote:
 On Wednesday 21 September 2005 07:27, Clive wrote:
 
My setup is:
telco-asterisk(voip)-asterisk{ITSP}telco
 
 
 Are both your asterisk boxes peered to each other?  IIRC trunking ONLY

 works between peers.

If you do iax2 show peers in the console, it should show a (T) for
trunked connections.

--
Cheers,

Matt Riddell
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Re: [Asterisk-Users] add 0 (zero) to incoming callerID - how?

2005-09-21 Thread Michiel van Baak

On 02:57, Thu 22 Sep 05, Matt Riddell wrote:
 Tomasz Chmielewski wrote:
  How can I manipulate the incoming callerID number (and add 0 before it)?
 
 exten = s,1,Answer()
 exten = s,2,SetCIDNum(0${CIDNUM})
 exten = s,3,...

And when using CVS head this will become:
exten = s,1,Set(CALLERID(number)=0${CALLERIDNUM})
exten = s,2,.

Just to be complete ;)
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Why is it drug addicts and computer afficionados are both called users?

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Re: [Asterisk-Users] Pinging ...

2005-09-21 Thread Steve Totaro

Its probably an IRQ sharing problem.

- Original Message - 
From: Alan Bunch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, September 19, 2005 10:09 AM
Subject: [Asterisk-Users] Pinging ...

 Ok, if I missed something in the wiki please point me there with the
 correct search terms.

 Asterisk 1.0.7  (AAH really)

 4 co lines from Bellsouth into a Diguim T400P.

 Polycom 501 x 4 on the desktops.

 My problem is on calls to or from the CO I hear a pinging (thing sonar
 ping in a submarine) every 12 seconds.  You can set your watch to it.
 COuld this be a call recording in progress tone.  I have not made any
 effort to turn on call recording.  If it is I'll go figure out how to
 turn it off.  Could this be echo cancelation retraining  Am I just
 u8nder dosing on my morning meds ?

 Any guesses from anyone here. I just want the pinging to stop.

 One more thing, this machine is remotely managed and is in a distant
 city.  If I really have to go I can.

 Thank in advance
 Alan
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Re: [Asterisk-Users] VM low volume - testers needed

 On Mon, 2005-09-19 at 10:38 -0600, Rich Adamson wrote:
  For those that have experienced low VM recording volumes when using
  a Digium TDM04b (or similar analog pstn card), a work around has been
  committed to cvs-head.
 
 Does this mean that tracking down the cause of the low volume issue was
 not successful or was the workaround an easier solution?

It means this is a short-term work around while there is an ongong
effort to identify the root cause (along with other possible TDM issues).


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Re: [Asterisk-Users] iax2 trunking wackyness

2005-09-21 Thread Andrew Kohlsmith

On Wednesday 21 September 2005 13:52, Adam Robins wrote:
 Should I plug in the actual IP addresses instead of host=dynamic?  Also,
 I do not currently have register statements.
 In iax.conf for these.

register = each to the other.

-A.
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[Asterisk-Users] oh323 driver and RFC2833

2005-09-21 Thread Fernando Herrera




Hello,

I have installed oh323 channel driver. Outgoing calls to 
H.323 world do not include RFC2833 in the outgoing TerminalCapabilitiesSet 
despite that userInputMode=RFC2833 has already been set. 

Does anyone know how to make RFC 2833 DTMF relay work over 
oh323 channel?

Kind regards, 

Fernando 
Herrera


  
  
  De: Fernando Herrera 
  [mailto:[EMAIL PROTECTED] Enviado el: Miércoles, 21 de 
  Septiembre de 2005 12:51Para: 
  'asterisk-users@lists.digium.com'Asunto: [Asterisk-Users] Help with 
  asterisk-oh323 driver
  
  
  

DV,

Have you solved 
this? I am facing the same problem. I am running Asterisk 1.0.9 and outgoing 
TCS does not show the 
receiveRTPAudioTelephonyEventCapability.

Kind regards, 


Fernando 
Herrera





  Hi all,

Sorry if this has been answered previously, but I have not had any
luck trying to find it.

I am trying to connect my Asterisk server (1.0 stable, Fedora Core 2,
kernel 2.6.8-1.521) to connect to a gateway that can only support
H323. I have installed the asterisk-oh323 channel driver (version
0.6.3b) using Open H323 1.13.5 (patched as per asterisk-oh323's
instructions) and PWLIB 1.6.6. This is all working fine for very basic
call setup and tear down, from any of my SCCP, SIP, H323 or POTS
(X100P card) phones.

NB: The gateway only handles signalling, so all media will flow
between the endpoints and the gateway will handle signalling to the
receiving gateway, as such (excuse the dodgy diagram :) ):

-[Gateway]---
|  |
(H323)(H323 or MGCP/ISUP)
|  |
   V V
[Asterisk]---(RTP)--[Terminating gateway]
   |
(Signalling + RTP)
   |
(Zaptel/SIP/H323/SCCP phones)


There are some requirements for me to connect to this switch:

1. I must support H245 tunneling and faststart (working fine)
2. I must dynamically negotiate the codecs (i.e. send multiple codecs
as part of the faststart and the softswitch will decide which of the
codecs to use based on the terminating gateway's capabilities). The
codec picked will be passed back in the return faststart from the
gateway.
3. It must support RFC2833 for OOB DTMF.

The problems I am facing are that my faststart in my setup messages
only ever has one codec, regardless of what I have set in the [codecs]
section of oh323.conf, and even if I specify userInputMode=RFC2833 in
oh323.conf my TCS does not include the capability
receiveRTPAudioTelephonyEventCapability hence RFC2833 is never
neogitated. I'm sure this is just a minor tweak of the source code,
but not being an expert in C I am having problems figuring out what
needs to be done and where.

Any help on this matter would be appreciated.

Cheers
DV


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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

2005-09-21 Thread [EMAIL PROTECTED]

I would think memory would be the limiting factor. A 3-4 minute wav
file is what, 30Meg or so? And there is one for each end of the call,
so that's 60Meg. Now let's say it's a 15 minute call and then are 10 of
them at once. That's 30Meg x 5 (5 times the length of my estimate) x 2
(each leg) x 10 simultaneous callsequals 3 Gig of RAM. Once you run
out of RAM, then what does it do? It would have to try and dump it all
to disk at once and you are back where you started. I think the average
* implementation doesn't have nearly enough free RAM to do this.

The RAMdisk solution seems to be pretty elegant in it's simplicity.

Matt Hess wrote:
In light of the I/O bottleneck problem I'd have to ask why asterisk
can't just buffer incoming audio and then flush a complete audio file to
disk.. I'm assuming that recordings vary in length.. the problem with
this idea is what happens if 50 recordings all complete at the same
time.. a dump like that might not be very pretty (a fast drive plus a
little scheduler or limiter so that only x number of files get written
to disk at a time would probably help out there) but I can imagine that
a single file being written is much more efficient and more
disk-friendly.. perhaps I don't know what the heck I'm talking about but
at least in my mind the theory sounds better than the current
'stream-to-file' method employed by asterisk.

Matt Roth wrote:

All,

This message has generated a lot of responses, so I'm going to address
each of them here in an attempt to consolidate the thread.

Matt,

- I'm very interested in the specifics of your setup.
- How much space is on the RAM disk?
Currently it is 10 GB. We are upgrading it to 16 GB.

- What kind of RAM drive is it?
The current RAM is 12 GB DDR2 400 MHz (12 x 1GB) Single Ranked DIMMs.
The details for each 1 GB DIMM can be seen here:

http://www.samsung.com/Products/Semiconductor/DRAM/DDR2SDRAM/DDR2SDRAMmodule/RegisteredDIMM/M393T2950CZ3/M393T2950CZ3.htm

The upgrade will involve adding 2 GB DIMMs to the system, but I don't
have the details on these yet.

The RAM disk is setup by adding the following kernel command line
option to grub.conf:

ramdisk_size=10485760

We are running Fedora Core 3 with the most up to date 2.6 SMP kernel.

By default the RAM disk's block size is 1024 bytes, so we are
formatting it as an ext2 file system with a block size of 1024 bytes
using the following command:

mke2fs -b 1024 -m 0 /dev/ram0

The block size can easily be changed from the kernel's view (using the
kernel command line option ramdisk_blocksize=) or from mke2fs's
view (using the -b argument), so please let me know if I can make
an easy optimization here.

Finally, the RAM disk is mounted using the command:

mount /dev/ram0 /digrec

A good RAMDISK howto exists at:

http://www.vanemery.com/Linux/Ramdisk/ramdisk.html

- What format are you recording to?
- What codec are the SIP calls being placed over?
We are recording to the PCM format and using the G711 uLaw codec.
High voice quality is essential to our application (we are a call
center) so we partnered with MCI to configure our network for the
required bandwidth and chose the highest quality, zero compression
codec. We noload all other codecs in order to avoid transcoding on
the switch, so we must record to PCM. Later (on a separate server) the
recordings are mixed to GSM which provides a 5 to 1 compression ratio
with very little artifacts.

- Good luck and please keep us updated on your progress
Thank you. I'll be keeping the list informed of our progress.

Zoa,

- I suppose you are the person from the digium forum
That was actually my boss's boss. We thank you all the way up and
down the line for your suggestion.

- The reason i recommended you to use a ramdisk is because i think the
- problem with recording to disk is saving 20ms of stream 1, then 20
ms of

- stream 2, then 20ms of stream 3 etc etc meaning you write everytime
- very small things. (with a lot of seeking).
Agreed. This is why we hope that decoupling the copy (memory to disk)
from Asterisk itself and, more importantly, Asterisk's real-time
handling of the call being recorded will be sufficient.

For

Re: [Asterisk-Users] VM low volume - testers needed

 On Monday 19 September 2005 12:38, Rich Adamson wrote:
  The g(6) adds a 6 db gain for zap calls that end up recording a Voicemail
  message.
 ...
 
  * 'g(#)' the specified amount of gain will be requested during message
   recording (units are whole-number decibels (dB))
 
 How in the hell does that make any sense?  are your normal incoming calls 
 quiet too or just voicemail?  

Yes, see bug 2022 and 2023 for details, as well as 
  http://www.routers.com/asteriskprob/asterisk-config.htm
for a very detailed analysis of the problem.

I believe one of the more serious issues amounts to: if asterisk is
located a fair distance from the central office (-7db in my case), setting
the rxgain and/or txgain to any level that would be considered reasonable
for that loss (eg, rxgain=5, txgain=5), hugh amounts of echo result that
cannot be addressed through zapata.conf echo entris, and changing
compile options to agressive, etc, does not help. Its my believe 
(from working with several TDM users), the further one is from the CO,
the bigger the problem. (Or, short pstn cable lengths less then about
4 or 5db can almost always be addressed via parameters.)

The above workaround is very usable (assuming it works) when someone
calls in via the pstn and leaves a voicemail (which is already at
least 7db down plus their own pstn loss), and then I call in via the
pstn to retrive the voicemail (now 14db down PLUS the original callers
pstn loss), the audio is so faint its difficult to impossible to 
listen to. 

  In my case, the asterisk box is located about 7db from the central
  office. As noted in bug 2023 (and 2022), calls from an outside pstn
  line coming into asterisk incure a 7db pstn loss (which can't be adjusted
  for with rxgain and txgain as changing those values to something
  reasonable generates echo).  Retrieving that VM message from an outside
  location creates another 7db loss (now -14db down in total), making it
  very difficult (if not impossible) to hear the message. (And, yes I've
  gone through all the recommendations with wav vs gsm files, etc.)
 
 I am not sure I understand why the txgain/rxgain isn't fixing it without 
 adding unacceptable echo...  this all seems very odd...  I mean for a test 
 you should be able to dial an echo() application and have extremely quiet 
 echoed audio... is this the case?

As an ex-telco transmission engineer, believe me I've done my homework
and some very solid testing with expensive well-calibrated test equipment.
As I've mentioned to Kevin, its almost like the TigerJet pci controller
on the TDM card is reversing bits six and seven (or something very odd
like that). Digium apparently now has a pci engineering type looking
at the issues, which I'm told is using a pci logic analyzer, etc.
 
  The work around only kicks in if the call comes from a zap channel
  and ends up in voicemail, adding a 6db gain to that recorded message.
  No other channel types are impacted by this new parameter.
 
 This is a HELL of a band-aid.

If you actually follow the logic that was originally stated in 2023,
this gain setting is highly useful for those systems that are 
further away from the CO (as mentioned above). For those closer to
the CO, it has zero value.

Rich


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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

The problem is that then it won't work on systems with little memory. 50
streams would eat memory like crazy.

Zoa

Matt Hess wrote:

In light of the I/O bottleneck problem I'd have to ask why asterisk
can't just buffer incoming audio and then flush a complete audio file
to disk.. I'm assuming that recordings vary in length.. the problem
with this idea is what happens if 50 recordings all complete at the
same time.. a dump like that might not be very pretty (a fast drive
plus a little scheduler or limiter so that only x number of files get
written to disk at a time would probably help out there) but I can
imagine that a single file being written is much more efficient and
more disk-friendly.. perhaps I don't know what the heck I'm talking
about but at least in my mind the theory sounds better than the
current 'stream-to-file' method employed by asterisk.

Matt Roth wrote:

All,

This message has generated a lot of responses, so I'm going to
address each of them here in an attempt to consolidate the thread.

Matt,

- I'm very interested in the specifics of your setup.
- How much space is on the RAM disk?
Currently it is 10 GB. We are upgrading it to 16 GB.

- What kind of RAM drive is it?
The current RAM is 12 GB DDR2 400 MHz (12 x 1GB) Single Ranked DIMMs.
The details for each 1 GB DIMM can be seen here:

http://www.samsung.com/Products/Semiconductor/DRAM/DDR2SDRAM/DDR2SDRAMmodule/RegisteredDIMM/M393T2950CZ3/M393T2950CZ3.htm

The upgrade will involve adding 2 GB DIMMs to the system, but I don't
have the details on these yet.

The RAM disk is setup by adding the following kernel command line
option to grub.conf:

ramdisk_size=10485760

We are running Fedora Core 3 with the most up to date 2.6 SMP kernel.

By default the RAM disk's block size is 1024 bytes, so we are
formatting it as an ext2 file system with a block size of 1024 bytes
using the following command:

mke2fs -b 1024 -m 0 /dev/ram0

The block size can easily be changed from the kernel's view (using
the kernel command line option ramdisk_blocksize=) or from
mke2fs's view (using the -b argument), so please let me know if
I can make an easy optimization here.

Finally, the RAM disk is mounted using the command:

mount /dev/ram0 /digrec

A good RAMDISK howto exists at:

http://www.vanemery.com/Linux/Ramdisk/ramdisk.html

- What format are you recording to?
- What codec are the SIP calls being placed over?
We are recording to the PCM format and using the G711 uLaw codec.
High voice quality is essential to our application (we are a call
center) so we partnered with MCI to configure our network for the
required bandwidth and chose the highest quality, zero compression
codec. We noload all other codecs in order to avoid transcoding on
the switch, so we must record to PCM. Later (on a separate server)
the recordings are mixed to GSM which provides a 5 to 1 compression
ratio with very little artifacts.

- We've run into the Avoided deadlock recording issues several
times when trying to do
- more than 50 concurrent recordings. Changing the ast_channel_lock
loop from 10 to 20 has
- helped somewhat reduce the warnings and reduce audio gaps on the
recordings, but what is
- really needed for more robust recording is a configurable recording
buffer that wouldn't
- freak out if a 10ms delay occurs.
Are you saying that these messages indicate a gap in a digital
recording? If so, what is the duration of the gap? If it's
comparable to a CD skip, I think we can deal with it until a buffer
or another solution is implemented.

- Good luck and please keep us updated on your progress
Thank you. I'll be keeping the list informed of our progress.

Zoa,

- I suppose you are the person from the digium forum
That was actually my boss's boss. We thank you all the way up and
down the line for your suggestion.

- The reason i recommended you to use a ramdisk is because i think the
- problem with recording to disk is saving 20ms of stream 1, then 20
ms of
- stream 2, then 20ms of stream 3 etc etc meaning you write
everytime
- very small things. (with a lot of seeking).
Agreed. This is why we hope that decoupling the copy (memory to
disk) from Asterisk itself and, more importantly, Asterisk's
real-time handling of the call being recorded will be sufficient.

For the record, when recording 512 simultaneous calls to the local
disk we saw a peek of about 13,000 blocks written per second.

- Our best test results were with:
-
- - buffering the recordings to a ramdisk, then
We're doing that, as per your suggestion.

[Asterisk-Users] Get SIP to work over very limited network access

2005-09-21 Thread Benjamin Lawetz

I've got a friend who's spending 6 months on the other side of the world. So
before he left I configured him a softphone on his laptop to connect to my
asterisk so he can call home free of charge.

Unfortunately, he just found out he has horrible internet connection.
Bandwith and latency is ok, the problem is the stop almost all connections.
He has to connect to a proxy server for his web access. The only thing that
seems to go through straight away is ssh or ftp. Thought about doing a ssh
tunnel, but that doesn't work over udp. He has linux and windows on his
machine, and is full configurable. The server (and my net connection) is
under my control and fully configurable.

So does anyone have any idea how to get him to connect to my asterisk server
through a TCP connection ?

thanks


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[Asterisk-Users] re: Problems with Queues