date:20050516

A kernel recompile managed to give me an average of 99.98%, but I am not
certain that this is optimal.  It seems that I get some background hiss,
although the static is gone.

Anyone out there with higher than this?

Thanks,
Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Wiktor - ADCom Corp.
Sent: Monday, May 16, 2005 5:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Static on TDM Zaptel FXO

Hello,
Yes in the bios I set unused irq's.  Zttest is sporadic.

This is a non HT cpu, on a standard ide drive.  It is basically just a
p4-2.4 with a single ide, onboard lan, and that's about it.  All other
asterisk functions work well.

Next step I think I'll try it on another board and see what happens...
Or maybe recompile the kernel.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damian
Funnell
Sent: Monday, May 16, 2005 5:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Static on TDM Zaptel FXO

Hi Gregory,

Have you checked that the card is on its own IRQ?

There has been a bit of discussion about this type of thing recently on
the list, do a search on the archive to find the various threads.

Try running zttest and see what accuracy it is reporting - anything less
than 99.99% is supposedly bad.

Various 'fixes' (work-arounds) are to make sure the zaptel is on it's
own IRQ, disable Hyperthreading and remove RAID controllers or any other
high-performance devices that support DMA and other things that are
likely to compete with the zaptel.

Good luck.
D.

FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz

Gregory Wiktor - ADCom Corp. wrote:

>Hello All,
>I recently put in a zaptel 1fxo/1fxs card.  I am experiencing heavy 
>static.
>
>Even with the pots line disconnected, if I do a dial I still get
static.
>This way I know it's not the line, but rather something on the card.
>
>I tried alternate pci slots.
>
>This card has a power connector, does anyone know what the power 
>requirements are?  The unit is in a small case with a 2.4ghz p-4 and 
>512mb ram, on an intel board with 533fsb.  All other functions are
fine.
>
>I am using the latest CVS on Debian 2.6test
>
>Anyone experience this?
>
>Regards,
>Greg
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[Asterisk-Users] H323 to SIP

2005-05-16 Thread Jeromy Grimmett

Title: Message




Hi 
all,
 
All outbound calls 
work perfect from my SIP ATA 
186...
 
SIP ATA 186 (w/ 
private IP) > Asterisk > H323 (public IP) > TDM > works 
perfect
 
TDM > H323 
(public IP) > Asterisk > SIP ATA 186 (w/ private IP) > fast busy with 
this error message in the CLI of Asterisk:
 
May 17 05:29:20 
WARNING[27823]: pbx.c:1890 ast_pbx_run: Channel 'OH323/R156' sent into invalid 
extension 's' in context 'default', but no invalid handlerCall 
'ip$200.94.273.2:10172/156' cleared.
anyone with any 
ideas i would greatly appreciate it...
 
Thanks,
Jeromy
 



  
  

  


  

  
  

  


  
  Global reach, local 
  touch...

  

  


  Jeromy GrimmettCEO 
  Comuniquémonos, Inc. / SA1212 South Hampton DriveAlexandria, LA 
71301 

  [EMAIL PROTECTED]IM: MSN: 
[EMAIL PROTECTED]http://www.comuniquemonos.com 
  

  
  
tel: fax: 
  mobile: 
+593 (4) 287 3854(501) 
  646-0680+593 (9) 366 6521 
  
  
  

  


  Add me to your address book...
  Want a signature like 
  this?
 
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[Asterisk-Users] Queued calls

2005-05-16 Thread Jason Walker





 
I need to track the 
final leg of a queue'd call from a ZAP channel to 
SIP channel through a call queue.
 
What I currently 
have setup is a queue that internal SIP traffic can call into and then receive 
on the far end. Since the ${EXTEN} is pre-determined based on the calling party, 
I can do a DBget within the dial plan to extract the IP from their 
registration.
 
When external calls 
come into the * box, the call gets routed to the same queue but I can not figure 
out how to determine the SIP extension the call eventually ended up at. In the 
CLI, I see an 'agent_call' directive and the call is sent to the appropriate SIP 
extension. But how, within the dialplan, can I retrieve that SIP 
call?
 
Also, once I route a 
call into the queue, the dialplan no longer appears to have control of the call. 
I can not find the config on how to control the call in 
queue.
 
 
Thank 
you.
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.11 - Release Date: 5/16/2005
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Re: [Asterisk-Users] outlook express intregation

2005-05-16 Thread C F

Why would you use outlook express when you could use Outlook? I don't
think outlook express is worth the effort of click to dial.

On 5/16/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> All of the stuff I've googled for and read on wiki all relate to "Outlook".
> Has anyone been successful in getting "Outlook Express" to do click to dial?
> 
> -Matthew
> 
> --
> 
> Matthew Boehm, IT DirectorCypress Telecommunications
> [EMAIL PROTECTED]   3838 N. Sam Houston Parkway E #400
> T: 832-200-8640 x3044  Houston, TX 77032
> 
> My girlfriend was recently diagnosed with multiple personality disorder;
>  When she called yesterday, my CallerID box exploded.
> 
> 
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[Asterisk-Users] Dial plan - does not stop after first match

2005-05-16 Thread Michael Stahl




My dial plan seems 
to work great - in that when I call extensions 1234 it connects to 1234.  
Strangely, after the call terminates (the other side hangs up first), Asterisk 
continues in the same context and then matches to extensions _. which causes an 
invalid extension error!
 
Why does asterisk 
not leave the context (called internalmenu) after the remote hangup?  
Instead, it continues to the InternalInvalid context (included later in the 
InternalMenu context).   I'm confused!
 
Here is a snippet of 
the relevant context and macro.  Thanks,
Mike
 
;; 
Macros;
 
[macro-stdexten];; Standard extension macro:;   
${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as 
well;   ${ARG2} - Device(s) to ring;exten => 
s,1,Playback(transfer,skip)  ; "Please hold while..." but skip if 
channel is not up;exten => s,2,SendText('Calling extension 
'${ARG1}) ; Tell the user what extension being calledexten => 
s,2,SetVar(LastStatus=CallDone) ; Ensure script knows that a Dial was 
completedexten => s,3,Dial(${ARG2},${RINGTIME},r) ; Ring the 
interface, 20 seconds maximumexten => 
s,4,NoOp(${DIALSTATUS})  ; Show status after hangupexten => 
s,5,Goto(s-${DIALSTATUS},1)  ; Jump based on status 
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
 
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to 
voicemail w/ unavail announce
 
exten => s-BUSY,1,Voicemail(b${ARG1})  ; If busy, send to 
voicemail w/ busy announce
 
exten => s-ANSWER,1,NoOp   ; If call answered, then do 
nothing after hangup
 
exten => _s-.,1,Goto(s-NOANSWER,1)  ; Treat anything else as 
no answer
 
exten => a,1,VoicemailMain(${ARG1})  ; If they press *, send 
the user into VoicemailMain
 
 
 
 
;; 
Extensions available to all 
callers;[default]
 
; Local extensions begin with 2XXXexten => 
${EXT_MIKE},1,Macro(stdexten,${EXTEN},${CON_MIKE})  ; Mike Stahl 
extensionexten => 
${EXT_SPA1},1,Macro(stdexten,${EXTEN},${CON_SPA1})  ; SPA line 1 
extensionexten => 
${EXT_SPA2},1,Macro(stdexten,${EXTEN},${CON_SPA2})  ; SPA line 2 
extensionexten => 
${EXT_DIANA},1,Macro(fakeexten,${EXTEN},${CON_DIANA})  ; Diana Stahl 
extensionexten => 
${EXT_KLAUS},1,Macro(stdexten,${EXTEN},${CON_KLAUS})  ; Klaus Stahl 
extensionexten => 
${EXT_BOB},1,Macro(fakeexten,${EXTEN},${CON_BOB})  ; Bob Stahl 
extensionexten => 
${EXT_BILL},1,Macro(fakeexten,${EXTEN},${CON_BILL})  ; Bill Stahl 
extension
 
; Direct into voicemail systemexten => 
${EXT_VOICEMAIL},1,VoicemailMainexten => ${EXT_VOICEMAIL},2,Hangup
 
; Dial by nameexten => 1,1,Directory(default)   ; 
Send user to the directory
 
 
 
;; Menu 
available to internal callers - full 
access;[menuinternal]
 
include => fwd ; Allow outbound access on 
FWD using 8 prefixinclude => trunkfullaccess   ; Allow 
full access to external line(s)include => 
default; Allow access to all extensions
 
; Administration menuexten => 
${EXT_ADMIN},1,Goto(menuadmin,s,1)
 
; After a message left, offer instruction what to do.; This menu is 
never reached INITIALLY if a specific extension is enteredexten => 
s,1,GotoIf($["${LastStatus}" != "CallDone"] & $["${LastStatus}" != 
""]?4) exten => 
s,2,SetVar(LastStatus=Try1)exten => s,3,Goto(11)exten => 
s,4,GotoIf($["${LastStatus}" != 
"Try1"]?10) exten => 
s,5,SetVar(LastStatus=UserIdle)exten => s,6,Goto(11)exten => 
s,7,GotoIf($["${LastStatus}" != 
"InvalidExtension"]?13) 
exten => s,8,SetVar(LastStatus=UserInvalid)exten => 
s,9,Goto(11)exten => s,10,SetVar(LastStatus=Error)exten => 
s,11,Background(ocg/enterextension) ; PLAY enter the extension you wish to 
dial 
 
; "*" Repeat the previous menuexten => *,1,Goto(s,1)
 
; "#" hangup exten => 
#,1,Goto(internalendcall,s,1)   ; End the call
 
; If any other extension then say invalid (used only for initial connection 
to allow connection; instead of 403 error)include => 
internalinvalid
 
; If they take too long, check if completed something useful (LastStatus) 
go back to menu, else timeoutexten => t,1,GotoIf($["${LastStatus}" = 
"UserIdle"] | $["${LastStatus}" = "UserInvalid"] | $["${LastStatus}" = 
"Error"]?internalendcall,s,1) 
exten => t,2,Wait(RepeatMenuWaitTime)  ; Wait for seconds 
before playing menu after  last command/eventexten => 
t,3,Goto(s,1); Go back to the main menu and repeat 
prompt
 
; If they enter an invalid extension.  Should only get here from 
_.,1exten => i,1,SetVar(LastStatus=InvalidExtension)exten => 
i,2,Background(pbx-invalid)   ; "That's not valid, try again" 
exten => i,3,Wait(1) exten => i,4,Goto(s,1) 
 
; timeout overallexten => T,1,Goto(internalendcall,s,1)
 
[internalinvali

Re: [Asterisk-Users] Outgoing spool file ignored

2005-05-16 Thread Jean-Denis Girard

Andres Paglayan a écrit :
File::copy does copy, it re-writes the file,
you need to move it.
so when the the pointer is placed the file is already there.
Well from File::Copy man page, about the move() function:
"If possible, move() will simply rename the file."
I thought it was the case on Linux, but I'll probably change to 
system(mv ...) just to be sure.

Thanks for all the replies.
Jean-Denis
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FW: [Asterisk-Users] Static on TDM Zaptel FXO

-Original Message-
From: Gregory Wiktor - ADCom Corp. 
Sent: Monday, May 16, 2005 5:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Static on TDM Zaptel FXO

Hello,
Yes in the bios I set unused irq's.  Zttest is sporadic.

This is a non HT cpu, on a standard ide drive.  It is basically just a
p4-2.4 with a single ide, onboard lan, and that's about it.  All other
asterisk functions work well.

Next step I think I'll try it on another board and see what happens...
Or maybe recompile the kernel.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damian
Funnell
Sent: Monday, May 16, 2005 5:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Static on TDM Zaptel FXO

Hi Gregory,

Have you checked that the card is on its own IRQ?

There has been a bit of discussion about this type of thing recently on
the list, do a search on the archive to find the various threads.

Try running zttest and see what accuracy it is reporting - anything less
than 99.99% is supposedly bad.

Various 'fixes' (work-arounds) are to make sure the zaptel is on it's
own IRQ, disable Hyperthreading and remove RAID controllers or any other
high-performance devices that support DMA and other things that are
likely to compete with the zaptel.

Good luck.
D.

FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz

Gregory Wiktor - ADCom Corp. wrote:

>Hello All,
>I recently put in a zaptel 1fxo/1fxs card.  I am experiencing heavy 
>static.
>
>Even with the pots line disconnected, if I do a dial I still get
static.
>This way I know it's not the line, but rather something on the card.
>
>I tried alternate pci slots.
>
>This card has a power connector, does anyone know what the power 
>requirements are?  The unit is in a small case with a 2.4ghz p-4 and 
>512mb ram, on an intel board with 533fsb.  All other functions are
fine.
>
>I am using the latest CVS on Debian 2.6test
>
>Anyone experience this?
>
>Regards,
>Greg
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>
>
>  
>
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[Asterisk-Users] spandsp in 64 bit Linux on AMD64

2005-05-16 Thread Juan Jose Comellas

Is there any stable version of spandsp that works on a 64 bit Linux on an 
AMD64 machine. When compiling version 0.0.1k I get the following error:

gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c testcpuid.c -MT testcpuid.lo -MD 
-MP -MF .deps/testcpuid.TPlo  -fPIC -DPIC -o .libs/testcpuid.lo
/tmp/ccXxGHg6.s: Assembler messages:
/tmp/ccXxGHg6.s:8: Error: suffix or operands invalid for `pushf'
/tmp/ccXxGHg6.s:9: Error: suffix or operands invalid for `pushf'
/tmp/ccXxGHg6.s:10: Error: suffix or operands invalid for `pop'
/tmp/ccXxGHg6.s:13: Error: suffix or operands invalid for `push'
/tmp/ccXxGHg6.s:14: Error: suffix or operands invalid for `popf'
/tmp/ccXxGHg6.s:15: Error: suffix or operands invalid for `pushf'
/tmp/ccXxGHg6.s:16: Error: suffix or operands invalid for `pop'
/tmp/ccXxGHg6.s:17: Error: suffix or operands invalid for `popf'
make[2]: *** [testcpuid.lo] Error 1



-- 
Juan Jose Comellas
([EMAIL PROTECTED])

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RE: [Asterisk-Users] Redirect two channels to each other?

2005-05-16 Thread mattf

This may be somewhat of a cross-post with -dev, but I have Manager Bridge
Action working under CVS_HEAD and releases 1.0.6-7.
http://bugs.digium.com/view.php?id=4297

MATT---

-Original Message-
From: Josiah Bryan [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 27, 2005 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Redirect two channels to each other?

On Wednesday 27 April 2005 11:38 am, Alexander Lopez wrote:
> As ny 10 year old step-daugher says "I don't get it.."
>
> Can't you just do a redirect if you specify the channels, * doesn't care
> if they are bridged together or not.  You may end up with zombie
> channels if the other leg does not drop, but you could do a soft hangup
> and take care of that..
>
>
> Or am I missing something

I dunno...maybe _im_ missing something..

IIRC (http://www.voip-info.org/wiki-Asterisk+Manager+API+Action+Redirect), 
Manager action 'Redirect' only takes Channel, ExtraChannel, Exten, Context, 
and Priority as parameters.  

Example (transferring existing 2 party call to a meetme room): 

Action: Redirect 
Channel: Zap/73-1 
ExtraChannel: SIP/199testphone-1f3c 
Exten: 8600029 
Context: default 
Priority: 1 

The problem is that you cant redirect to an existing _channel_.  I dont know

of any channel 'hack' like there is for Local extensions (e.g. to make an 
extension look like a channel, use Local/[EMAIL PROTECTED], etc - is there the 
inverse of that? Make a channel look like an extension?) 

It almost sounds like there needs to me a new manager action:

Action: Bridge
ChannelA: SIP/199testfone-1f3c
ChannelB: Zap/6-1

It sounds like the intrinsic functionality for 'bridging' is already there
in 
Asterisk (duh!), it just needs to be encapsulated in a manager action.

Any takers? Maybe a bounty is needed...?

-josiah

-- 
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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Re: [Asterisk-Users] voicemail.conf from DB

2005-05-16 Thread Matthew Boehm

Preston Garrison wrote:

> the database.  But that is mostly due to the fact that realtime
> queries the database 4 times everytime it needs a sip record.  Also


?? I have not observed this 4 query behavior. I only see 1 query.

-Matthew
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RE: [Asterisk-Users] voicemail.conf from DB

2005-05-16 Thread Senad J

[EMAIL PROTECTED] wrote:
> Actually using voicemail from db is alot smarter then using SIP from
> the database.


Depens how you administer your configuration files.

>But that is mostly due to the fact that realtime
> queries the database 4 times everytime it needs a sip record.  Also
> mysql doesn't keep everything in memory at all.  It only keeps stuff
> it needs to in memory, that it finds itself accessing alot.  The
> algorithms are pretty damn good in mysql about what to keep in memory.

I was not talking about MySQL on a single host...
I was refering to MYSQL Cluster.

Regards,

Senad 

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Re: [Asterisk-Users] Pass variable to Authenticate?

2005-05-16 Thread Moises Silva

Hi Dana. 

Not sure how to do it with DBget, but currently im using AGI for all
the authentication process. I Think is a very good solution. Its a lot
easier to make a program in a high level language like PHP and tell
asterisk with a simple SetVar(authenticated=1) if the user is allowed
to continue.
Basically the flow of events is:

- the user dial some number
- asterisk prompts for the agent login and saves it in a channel variable
- asterisk prompts for the agent password and saves it in other channel var
- asterisk launch an AGI application
   - the agi script retrieves the login and password saved by Asterisk
   - the agi script check the database to find a match entry
   - the agi script set an Asterisk variable to tell if the user
its authorized or not
- asterisk reads the variable that agi should have set
- if the variable is 1 continue, otherwise ask the password again or
hangup, or whatever...

If you need more help just ask. I tough many people here knows other
options, or may be a better implementation.

Best Regards

- moy

On 5/16/05, Dana Olson <[EMAIL PROTECTED]> wrote:
> I'm trying to figure out a way to make my own agent login, because I
> don't like how the default works.
> 
> I have the login and logout working fine using the dynamic add and
> remove commands, but I need to be able to create a list of users and
> passwords.
> 
> I thought of a way to do it using a list of passwords, but the agent
> would only ever be prompted for their password. I won't want that.
> 
> Basically, what I want to do is this:
> 
> Read in the username, ex. 4567
> Read in the password, ex. 1234
> Pass 12344567 to the Authenticate command, which checks in the file
> for that password.
> 
> I can't figure out a way to do it, and it may not even be possible.
> 
> Is there another way to do it? Can I somehow utilize the DBget and
> DBput commands? I couldn't find much info on it in the wiki, and my
> impression is that I have to use only the blacklist or cidname
> databases.
> 
> Any ideas would be helpful.
> --
> Dana
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-- 
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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RE: [Asterisk-Users] Static on TDM Zaptel FXO

Hello,
Yes in the bios I set unused irq's.  Zttest is sporadic.

This is a non HT cpu, on a standard ide drive.  It is basically just a
p4-2.4 with a single ide, onboard lan, and that's about it.  All other
asterisk functions work well.

Next step I think I'll try it on another board and see what happens...
Or maybe recompile the kernel.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damian
Funnell
Sent: Monday, May 16, 2005 5:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Static on TDM Zaptel FXO

Hi Gregory,

Have you checked that the card is on its own IRQ?

There has been a bit of discussion about this type of thing recently on
the list, do a search on the archive to find the various threads.

Try running zttest and see what accuracy it is reporting - anything less
than 99.99% is supposedly bad.

Various 'fixes' (work-arounds) are to make sure the zaptel is on it's
own IRQ, disable Hyperthreading and remove RAID controllers or any other
high-performance devices that support DMA and other things that are
likely to compete with the zaptel.

Good luck.
D.

FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz

Gregory Wiktor - ADCom Corp. wrote:

>Hello All,
>I recently put in a zaptel 1fxo/1fxs card.  I am experiencing heavy 
>static.
>
>Even with the pots line disconnected, if I do a dial I still get
static.
>This way I know it's not the line, but rather something on the card.
>
>I tried alternate pci slots.
>
>This card has a power connector, does anyone know what the power 
>requirements are?  The unit is in a small case with a 2.4ghz p-4 and 
>512mb ram, on an intel board with 533fsb.  All other functions are
fine.
>
>I am using the latest CVS on Debian 2.6test
>
>Anyone experience this?
>
>Regards,
>Greg
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Re: [Asterisk-Users] Broadvoice Toll-Free IVR issues

You should try each possible setting and see if the primary and tollfree 
numbers work with them. Then call BV, wait on hold for 15 minutes and 
get dropped to a fast busy signal.

[EMAIL PROTECTED] wrote:
I'm using dtmf=inband and dtmfmode=inband. Should I try rfc2833? From what
I could tell, Broadvoice "might" support it, but others have said that they
don't.
Original Message:
-
From: Paul [EMAIL PROTECTED]
Date: Mon, 16 May 2005 15:28:48 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Broadvoice Toll-Free IVR issues
[EMAIL PROTECTED] wrote:
 

Perhaps someone else has had this issue. I currently have BroadVoice as my
SIP provider with two phone numbers. The first number is my primary local
number; the second is an 877 toll-free number.
The issue is with the IVR abilities. When calling the primary number you
can press 200 at the main menu and receive my extension. When calling the
877 number, 200 will result in “that option is invalid”… Heck, just after
pressing “2” it will give that message. If you press 1 it takes you to the
support queue. The IVR is set up to give you the sales queue after pressing
2, but it doesn’t do that either.
Anyone have any ideas?

   

I remember running into something else that was different for the 
tollfree number with BV. IIRC - the supported codecs differed. It makes 
me think that the tollfree calls come from a different gateway. They 
only allow one SIP device(whether hardware or software) per account 
which rules out the most obvious workaround for the problem. Looks to me 
like dtmf settings are different on the two gateways/routers they are using.

 

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RE: [Asterisk-Users] voicemail.conf from DB


and running mysql with master slave config, can also allow load balancing
on the queries, but the problem is I just cant get it too work :-)

Iqbal

On 5/16/2005, "Preston Garrison" <[EMAIL PROTECTED]> wrote:

>Actually using voicemail from db is alot smarter then using SIP from
>the database.  But that is mostly due to the fact that realtime queries
>the database 4 times everytime it needs a sip record.  Also mysql
>doesn't keep everything in memory at all.  It only keeps stuff it needs
>to in memory, that it finds itself accessing alot.  The algorithms are
>pretty damn good in mysql about what to keep in memory.
>
>Preston Garrison
>direct: 877-748-4142
>fax: 310-774-3901
>cell: 623-748-4140
>
>-Original Message-
>From: Senad J <[EMAIL PROTECTED]>
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>
>Sent: Mon, 16 May 2005 20:33:32 +0100
>Subject: RE: [Asterisk-Users] voicemail.conf from DB
>
>In my opinion.. using voicemail from DB is not good
>for "scaling up" scenarios... in particular if you need to
>provide HA of five nine... 99.999%
>
>The cluster database (well in particular MYSQL puts everything in
>RAM)... so
>as the user base grows.. so does RAM requirement...
>
>Ta
>Senad
>
>[EMAIL PROTECTED] wrote:
>> Hi
>>
>> I have been playing with trying to get voicemail.conf from DB, I am
>> using cvs-head, but when I start asterisk, it dies a horrible death,
>> because it cant load any voicemil setting.
>>
>> I looked at my mysql logs to see what query was being sent, and I get
>>
>> SELECT category, var_name, var_val, cat_metric FROM voicemail_users
>> WHERE filename='voicemail.conf' and commented=0 ORDER BY filename,
>> cat_metric desc, var_metric asc, category, var_name, var_val, id;
>>
>> which is in no way related to the table I created following the
>> example on the wiki.
>>
>> Is there another table format , I have
>>
>>
>+-+---+--+-+-++
>>> Field   | Type  | Null | Key | Default | Extra  |
>>
>+-+---+--+-+-++
>>> uniqueid| int(11)   |  | PRI | NULL| auto_increment |
>>> customer_id | int(11)   |  | | 0   ||
>>> context | varchar(50)   |  | | ||
>>> mailbox | int(5)|  | MUL | 0   ||
>>> password| varchar(4)|  | | 0   ||
>>> fullname| varchar(50)   |  | | ||
>>> email   | varchar(50)   |  | | ||
>>> pager   | varchar(50)   |  | | ||
>>> stamp   | timestamp(14) | YES  | | NULL||
>>
>+-+---+--+-+-++
>>
>>
>> tks
>> Iqbal
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[Asterisk-Users] Home Usage

2005-05-16 Thread Nathan Pralle

Hi all.
I'm curious to hear about other people's HOME usage of Asterisk.  Do you 
have a really neat setup for home use?  Fun stuff with VM and/or 
forwarding and custom scripts?

I'm working on my own solution; curious as to what others have/use.
Nathan
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[Asterisk-Users] Re: SpanDSP TXFax and multipage faxes problems (aditional info)

2005-05-16 Thread Nenad Radosavljevic

Lee,
thanks for the explanations.
Since I finaly figured out where to find spandsp debug information (it is 
printed only on the console where the asterisk is run, not in Asterisk 
remote console nor in Asterisk log files), I have gathered some debug logs 
and emailed them to Steve, asking him for his oppinion on problem (logs are 
long so I won't bother list with them).

Anyway, thanks again.
Regards,
Nenad

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Re: [Asterisk-Users] xbox asterisk?

2005-05-16 Thread C F

I didn't read the whole article, but with the known ways of getting
the xBox to boot into your favorite *nix kernel (just google a bit),
it just sounds like an easy way to get your favorite game to pause
when the phone rings.

On 5/16/05, Dean Collins <[EMAIL PROTECTED]> wrote:
>  
>  
> 
> http://www.pbs.org/cringely/pulpit/pulpit20050512.html 
> 
>   
> 
> interesting comment this week about the Xbox – any intelligent thoughts
> here? 
> 
>   
> 
> I know the price point puts it above most users Asterisk outlay (I run mine
> on a $100 P3 -800) 
> 
>   
> 
> But interesting to see what happens if people start running video
> conferencing etc on their home asterisk servers, and lets face it where else
> can you buy this kind of subsidized processing power from at that price. 
> 
>   
> 
>   
> 
> Cheers, 
> 
> Dean 
> 
>   
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RE: [Asterisk-Users] Forwarding To Cell Phones with Asterrisk PBX

2005-05-16 Thread Colin Anderson




Yeah, I have it in 
my dialplan and use it heavily. Just make another Dial() command to the 
cellphone the next priority in the dialplan underneath the Dial() statement for 
your extension. For example:
 
Extension is: 
12345
SIP extension is: 
SIP/12345
Cell number is: 
555-1212
 
in 
Extensions.conf:
 
[myphonecontext]
 
exten => 
12345,1,Dial(SIP/12345,40) 'Dial extension 12345 for 40 seconds. If no one picks 
up then...
exten => 
12345,2,Dial(ZAP/g0/5551212,25) 'Forward the call out to the user's cell. Once 
they pick up, a native bridge of ZAP channels occur and Asterisk is out 'of the 
media stream
exten => 
12345,3,(anything else that happens later, like go to voicemail, 
etc)
 
It's important to 
time how long it takes for the remote user's cellphone to pick up for voicemail. 
If the user's voicemail on the cell kicks in after, say 4 rings, time your 
second Dial() command to be just short of that, otherwise the remote caller will 
get the cell phone's voicemail, which is probably not the desired behavior. In 
my case, I set it for 25 seconds, as our cells' voicemail kicks in after 30 
seconds. If there's no call pickup on the cell, call processing continues to the 
next priority, which is voicemail or IVR depending on what number they called. 

 
Also note that once 
the native bridging happens, you are using two lines, 1 inbound to Asterisk, and 
1 outbound from Asterisk to the cell phone. Line capacity becomes an issue 
unless you have lots of channels, like a PRI, or if your useage is light, like 
no more than 1/2 of your total Zap channels could be inbound and forwarded to 
your remote user's cells at any one point in time. 
 
hth

  -Original Message-From: Theo Chao 
  [mailto:[EMAIL PROTECTED]Sent: Monday, May 16, 2005 2:39 
  PMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] Forwarding To Cell Phones with Asterrisk 
  PBX
  
  Hello,
   
  My company has a 800 number that we use for 
  customer service.  However, instead of having our customer service reps 
  at the office, we route these calls to their cell phones using a service 
  provided by gosolo.com.  However our current system isn't ideal because 
  it will call each number in order causing longer and longer waits when we've 
  got people on the line.   Our goal is to set up a system that allows 
  us to design the way calls into an 800 number are routed out to our cell 
  phones.
   
  I've been reading about Asterisk and VoIP to see 
  if we can better this process by having all the reps called at once, and 
  whichever person answers first handles the call.  It looks like this may 
  be possible with the dial command (http://www.voip-info.org/wiki-Asterisk+cmd+DIal)   
  However, would we be able to forward incoming calls to cellular phones?  
  The wiki reference for the dial command talks about using channels and the 
  list of possible channels doesn't seem to have an option for cell 
  phones.  
   
  Additionally, I've looked at http://www.voip-info.org/tiki-index.php?page=Asterisk%20Connecting%20to%20the%20Cellular%20Network 
  to see how to connect Asterisk to a cellular network.  However it looks 
  to me like this is a method to take a call placed to a cell phone and route it 
  so that other home phones could be used instead.  Is there a method to go 
  the other way around and take a call placed to a land line and route it to the 
  cell phone?
   
  Thanks,Theo
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Re: [Asterisk-Users] Forwarding To Cell Phones with Asterrisk PBX

Theo Chao wrote:
Hello,

My company has a 800 number that we use for customer service.
However, instead of having our customer service reps at the office, we
route these calls to their cell phones using a service provided by
gosolo.com. However our current system isn't ideal because it will
call each number in order causing longer and longer waits when we've
got people on the line. Our goal is to set up a system that allows
us to design the way calls into an 800 number are routed out to our
cell phones.

I've been reading about Asterisk and VoIP to see if we can better this
process by having all the reps called at once, and whichever person
answers first handles the call. It looks like this may be possible
with the dial command
(http://www.voip-info.org/wiki-Asterisk+cmd+DIal) However, would we
be able to forward incoming calls to cellular phones? The wiki
reference for the dial command talks about using channels and the list
of possible channels doesn't seem to have an option for cell phones.

Additionally, I've looked at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Connecting%20to%20the%20Cellular%20Network
to see how to connect Asterisk to a cellular network. However it
looks to me like this is a method to take a call placed to a cell
phone and route it so that other home phones could be used instead.
Is there a method to go the other way around and take a call placed to
a land line and route it to the cell phone?

Thanks,
Theo
Have the system call multiple numbers and start playing an identifying
message on answer. The first one to send back a dtmf 1 gets the call.
That prevents the * system from sending callers into things like cell
voicemail or network error messages.

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RE: [Asterisk-Users] POE hub

2005-05-16 Thread Max W Blackmer Jr

>  Original Message 
> Subject: [Asterisk-Users] POE hub
> From: "Chris Mason" <[EMAIL PROTECTED]>
> Date: Sun, May 15, 2005 9:19 pm
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
>
> I need to connect up to sixteen phones per building, I can use a cheap hub,
> but POE would be useful. Is there a cheap POE hub available? Everything I
> have seen has been expensive.

Well for inexpensive you could look at D-Link They have 2 switches that
provide POE.

DES-1316 Web-Smart 16-port Switch Web-Smart Switch with (8) Ports
10/100Mbps and (8) PoE 802.3af 10/100 ports
List price is  $499.99 USD. I can get them for $420 USD.

DES-1526 Web-Smart 24-port PoE 10/100 + 2 Combo Gigabit Copper/SFP ports
Switch.
List price is $999. I can get them for $850.

Cheers,

Max W. Blackmer, Jr.


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[Asterisk-Users] voicemail on server, path format


Hi

can the path format be changed on the server, eg can lookup be

../context/1/2/3/123.gsm

kind of like what you would have in order to store email across a
filesystem, it makes lookups and scalability much much better/faster

Iqbal
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Re: [Asterisk-Users] Static on TDM Zaptel FXO

2005-05-16 Thread Damian Funnell

Hi Gregory,
Have you checked that the card is on its own IRQ?
There has been a bit of discussion about this type of thing recently on 
the list, do a search on the archive to find the various threads.

Try running zttest and see what accuracy it is reporting - anything less 
than 99.99% is supposedly bad.

Various 'fixes' (work-arounds) are to make sure the zaptel is on it's 
own IRQ, disable Hyperthreading and remove RAID controllers or any other 
high-performance devices that support DMA and other things that are 
likely to compete with the zaptel.

Good luck.
D.
FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz

Gregory Wiktor - ADCom Corp. wrote:
Hello All,
I recently put in a zaptel 1fxo/1fxs card.  I am experiencing heavy
static.  

Even with the pots line disconnected, if I do a dial I still get static.
This way I know it's not the line, but rather something on the card.
I tried alternate pci slots.
This card has a power connector, does anyone know what the power
requirements are?  The unit is in a small case with a 2.4ghz p-4 and
512mb ram, on an intel board with 533fsb.  All other functions are fine.
I am using the latest CVS on Debian 2.6test
Anyone experience this?
Regards,
Greg
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Re: [Asterisk-Users] ShoreTel 210 MGCP phone drops calls with MGCP RSIP

2005-05-16 Thread Ben Dugdale

Not sure how much of this you want to see.  The drop is at the end.

MGCP Debugging Enabled
MGCP read:
NTFY 4255 [EMAIL PROTECTED] MGCP 1.0
X: 1C8A5876
O: l/hd

from 192.168.1.137:2427
Verb: 'NTFY', Identifier: '4255', Endpoint:
'[EMAIL PROTECTED]', Version: 'MGCP 1.0'
3 headers, 0 lines
Handling request 'NTFY' on [EMAIL PROTECTED]
Transmitting:
200 4255 OK

 to 192.168.1.137:2427
-- Creating connection for [EMAIL PROTECTED] in
cxmode: sendrecv callid: 5af2b0987ba3281b
We're at 192.168.0.5 port 5004
Answering with capability 4
Posting Request:
CRCX 209 [EMAIL PROTECTED] MGCP 1.0
C: 5af2b0987ba3281b
L: p:20, a:PCMU
M: sendrecv
X: 7ba3281b

v=0
o=root 28578 28578 IN IP4 192.168.0.5
s=session
c=IN IP4 192.168.0.5
t=0 0
m=audio 5004 RTP/AVP 0
a=rtpmap:0 PCMU/8000
 to 192.168.1.137:2427
-- MGCP Asked to indicate tone: L/dl on
[EMAIL PROTECTED] in cxmode: sendrecv
Posting Request:
RQNT 210 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
S: L/dl
 to 192.168.1.137:2427
MGCP read:
200 209 OK
I: 3886

v=0
o=Broadcom 14470 209 IN IP4 192.168.1.137
s=MGCP call
c=IN IP4 192.168.1.137
b=AS:64
t=0 0
m=audio 5004 RTP/AVP 0
a=sendrecv
a=ptime:20

from 192.168.1.137:2427
Verb: '200', Identifier: '209', Endpoint: 'OK', Version: '(null)'
2 headers, 9 lines
Capabilities: us - 4, them - 4, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
MGCP read:
NTFY 4256 [EMAIL PROTECTED] MGCP 1.0
X: 7BA3281B
O: aud(2)

from 192.168.1.137:2427
Verb: 'NTFY', Identifier: '4256', Endpoint:
'[EMAIL PROTECTED]', Version: 'MGCP 1.0'
3 headers, 0 lines
Handling request 'NTFY' on [EMAIL PROTECTED]
Transmitting:
200 4256 OK

 to 192.168.1.137:2427
-- MGCP Asked to indicate tone: L/dl on
[EMAIL PROTECTED] in cxmode: sendrecv
Queueing Request:
RQNT 211 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
S: L/dl
 to 192.168.1.137:2427
May 16 13:47:07 NOTICE[28578]: chan_mgcp.c:3242 handle_request: Received
unknown event 'aud(2)' from [EMAIL PROTECTED]
MGCP read:
200 210 OK

from 192.168.1.137:2427
Verb: '200', Identifier: '210', Endpoint: 'OK', Version: '(null)'
1 headers, 0 lines
Posting Queued Request:
RQNT 211 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
S: L/dl
 to 192.168.1.137:2427
MGCP read:
200 211 OK

from 192.168.1.137:2427
Verb: '200', Identifier: '211', Endpoint: 'OK', Version: '(null)'
1 headers, 0 lines
-- MGCP asked to indicate -1 'UNKNOWN' condition on channel
MGCP/[EMAIL PROTECTED]
-- MGCP Asked to indicate tone:  on
[EMAIL PROTECTED] in cxmode: sendrecv
Posting Request:
RQNT 212 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
 to 192.168.1.137:2427
-- MGCP asked to indicate -1 'UNKNOWN' condition on channel
MGCP/[EMAIL PROTECTED]
-- MGCP Asked to indicate tone:  on
[EMAIL PROTECTED] in cxmode: sendrecv
Queueing Request:
RQNT 213 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
 to 192.168.1.137:2427
MGCP read:
200 212 OK

from 192.168.1.137:2427
Verb: '200', Identifier: '212', Endpoint: 'OK', Version: '(null)'
1 headers, 0 lines
Posting Queued Request:
RQNT 213 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
 to 192.168.1.137:2427
MGCP read:
200 213 OK

from 192.168.1.137:2427
Verb: '200', Identifier: '213', Endpoint: 'OK', Version: '(null)'
1 headers, 0 lines
-- MGCP asked to indicate -1 'UNKNOWN' condition on channel
MGCP/[EMAIL PROTECTED]
-- MGCP Asked to indicate tone:  on
[EMAIL PROTECTED] in cxmode: sendrecv
Posting Request:
RQNT 214 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
 to 192.168.1.137:2427
-- MGCP asked to indicate -1 'UNKNOWN' condition on channel
MGCP/[EMAIL PROTECTED]
-- MGCP Asked to indicate tone:  on
[EMAIL PROTECTED] in cxmode: sendrecv
Queueing Request:
RQNT 215 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
 to 192.168.1.137:2427
MGCP read:
200 214 OK

from 192.168.1.137:2427
Verb: '200', Identifier: '214', Endpoint: 'OK', Version: '(null)'
1 headers, 0 lines
Posting Queued Request:
RQNT 215 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
 to 192.168.1.137:2427
MGCP read:
200 215 OK

from 192.168.1.137:2427
Verb: '200', Identifier: '215', Endpoint: 'OK', Version: '(null)'
1 headers, 0 lines
-- MGCP asked to indicate -1 'UNKNOWN' condition on channel
MGCP/[EMAIL PROTECTED]
-- MGCP Asked to indicate tone:  on
[EMAIL PROTECTED] in cxmode: sendrecv
Posting Request:
RQNT 216 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
 to 192.168.1.137:2427
-- MGCP asked to indicate -1 'UNKNOWN' condition on channel
MGCP/[EMAIL PROTECTED]
-- MGCP Asked to indicate tone:  on
[EMAIL PROTECTED] in cxmode: sendrecv
Queueing Request:
RQNT 217 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
 to 192.168.1.137:2427
MGCP read:
200 216 OK

from 192.168.1.137:2427
Verb: '200', Identifier: '216', Endpoint: 'OK', Version: '(null)'
1 headers, 0 lines
Posting Queued Request:
RQNT 217 [EMAIL PROTECTED] MGCP 1.0
X: 1c8a5876
R: L/hu(N),L/hf(N)
 to 192.168.1

Re: [Asterisk-Users] zttest

2005-05-16 Thread Rich Adamson

> Rich, did you check IRQ's?  Our zttest results didn't improve markedly 
> when we did either change (IRQ's or H/T), but the problem went away 
> regardless.
> 
> Other than that Digium have recommended throwing out our SCSI320 RAID 
> hardware and replacing it with IDE (i.e. not SATA) kit, although 
> thankfully we haven't had to make this retarded change (yet).
> 
> I would also recommend trying to disable H/T in the BIOS (rather than 
> via software) as I wonder if H/T still runs on your box (but is not 
> accessed by the OS), so may still be causing you grief.

Yes, been through the majority of items including irq's, swapped
motherboards, ide only, nothing else running, command line only (no
frame buffers), upgrade from rhv9 to fedora 3, mucked with pci
latency, etc.  Had not tried the hyperthread thing until now, but 
that had no effect either. Even upgraded the TDM card to the latest 
rev h.

Still think its either a bad TDM card design, or, TDM driver issue
as to why we can't approach zttest results of 100%.

I'm hoping that Steve Underwood's utility to measure the TDM results
helps to identify the root cause.



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RE: [Asterisk-Users] voicemail.conf from DB

Hi

tks for the feedback, the admintool i cant use, because users create/add
themselves to the system themselves, could be 100 or 1000+ users. Hence
I could get my script which create user/pass details in myqsql to call
the voicemail script to create the physical path on the server, but
appending 1000+ lines to voicemail.conf doesnt seem like a good idea,
and then reloading each time.

I looked at the realtime feature in asterisk and it looked good, I want
to store the voicemail on the server itself, (wish i could change the
dir struture though, rather than have one ../context/voicemail etc I
would want it split like a mbox mail directory structure for large
scalabilty---but thats another story-new thread), and the mapping ,
user/pass details for each user pull from the DB.

Iqbal

On 5/16/2005, "Senad J" <[EMAIL PROTECTED]> wrote:

>[EMAIL PROTECTED] wrote:
>> but the other choice is to keep editing the voicemail.conf file,
>> everytime I add a new user, which again is not really scalable.
>
>Using an administration interface of some kind will solve this issue.
>>
>> I dont wish to store the voicemail in the DB, just the conf file
>> itself, mysql will easily in a clusered scenario support 100K+
>> entries.
>>
>
>Fine...
>> What setup would you recommend
>
>Do do what?
>
>Senad
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[Asterisk-Users] Static on TDM Zaptel FXO

Hello All,
I recently put in a zaptel 1fxo/1fxs card.  I am experiencing heavy
static.  

Even with the pots line disconnected, if I do a dial I still get static.
This way I know it's not the line, but rather something on the card.

I tried alternate pci slots.

This card has a power connector, does anyone know what the power
requirements are?  The unit is in a small case with a 2.4ghz p-4 and
512mb ram, on an intel board with 533fsb.  All other functions are fine.

I am using the latest CVS on Debian 2.6test

Anyone experience this?

Regards,
Greg
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[Asterisk-Users] Forwarding To Cell Phones with Asterrisk PBX

2005-05-16 Thread Theo Chao





Hello,
 
My company has a 800 number that we use for 
customer service.  However, instead of having our customer service reps at 
the office, we route these calls to their cell phones using a service provided 
by gosolo.com.  However our current system isn't ideal because it will call 
each number in order causing longer and longer waits when we've got people on 
the line.   Our goal is to set up a system that allows us to design 
the way calls into an 800 number are routed out to our cell phones.
 
I've been reading about Asterisk and VoIP to see if 
we can better this process by having all the reps called at once, and whichever 
person answers first handles the call.  It looks like this may be possible 
with the dial command (http://www.voip-info.org/wiki-Asterisk+cmd+DIal)   
However, would we be able to forward incoming calls to cellular phones?  
The wiki reference for the dial command talks about using channels and the list 
of possible channels doesn't seem to have an option for cell phones.  

 
Additionally, I've looked at http://www.voip-info.org/tiki-index.php?page=Asterisk%20Connecting%20to%20the%20Cellular%20Network 
to see how to connect Asterisk to a cellular network.  However it looks to 
me like this is a method to take a call placed to a cell phone and route it so 
that other home phones could be used instead.  Is there a method to go the 
other way around and take a call placed to a land line and route it to the cell 
phone?
 
Thanks,Theo
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RE: [Asterisk-Users] voicemail.conf from DB

2005-05-16 Thread Senad J

[EMAIL PROTECTED] wrote:
> but the other choice is to keep editing the voicemail.conf file,
> everytime I add a new user, which again is not really scalable.

Using an administration interface of some kind will solve this issue.
> 
> I dont wish to store the voicemail in the DB, just the conf file
> itself, mysql will easily in a clusered scenario support 100K+
> entries. 
> 

Fine...
> What setup would you recommend

Do do what?

Senad
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Re: [Asterisk-Users] Broadvoice Toll-Free IVR issues

2005-05-16 Thread [EMAIL PROTECTED]

I'm using dtmf=inband and dtmfmode=inband. Should I try rfc2833? From what
I could tell, Broadvoice "might" support it, but others have said that they
don't.

Original Message:
-
From: Paul [EMAIL PROTECTED]
Date: Mon, 16 May 2005 15:28:48 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Broadvoice Toll-Free IVR issues


[EMAIL PROTECTED] wrote:

>Perhaps someone else has had this issue. I currently have BroadVoice as my
>SIP provider with two phone numbers. The first number is my primary local
>number; the second is an 877 toll-free number.
>
>The issue is with the IVR abilities. When calling the primary number you
>can press 200 at the main menu and receive my extension. When calling the
>877 number, 200 will result in that option is invalid Heck, just after
>pressing 2 it will give that message. If you press 1 it takes you to the
>support queue. The IVR is set up to give you the sales queue after pressing
>2, but it doesnt do that either.
>
>Anyone have any ideas?
>
>  
>
I remember running into something else that was different for the 
tollfree number with BV. IIRC - the supported codecs differed. It makes 
me think that the tollfree calls come from a different gateway. They 
only allow one SIP device(whether hardware or software) per account 
which rules out the most obvious workaround for the problem. Looks to me 
like dtmf settings are different on the two gateways/routers they are using.


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mail2web - Check your email from the web at
http://mail2web.com/ .


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Re: [Asterisk-Users] NAT and sip issues

2005-05-16 Thread Richard Malcolm-Smith


G.Marshall wrote:
The rtp audio is going phone to phone, not via asterisk.  This is one of
the reasons I am trying to set up SER with Asterisk.
I thought that canreinvite=no was supposed to force the audio to go via asterisk?


smime.p7s
Description: S/MIME Cryptographic Signature
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RE: [Asterisk-Users] voicemail.conf from DB

2005-05-16 Thread Preston Garrison

Actually using voicemail from db is alot smarter then using SIP from 
the database.  But that is mostly due to the fact that realtime queries 
the database 4 times everytime it needs a sip record.  Also mysql 
doesn't keep everything in memory at all.  It only keeps stuff it needs 
to in memory, that it finds itself accessing alot.  The algorithms are 
pretty damn good in mysql about what to keep in memory.

Preston Garrison
direct: 877-748-4142
fax: 310-774-3901
cell: 623-748-4140
-Original Message-
From: Senad J <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Mon, 16 May 2005 20:33:32 +0100
Subject: RE: [Asterisk-Users] voicemail.conf from DB

In my opinion.. using voicemail from DB is not good
for "scaling up" scenarios... in particular if you need to
provide HA of five nine... 99.999%
The cluster database (well in particular MYSQL puts everything in 
RAM)... so
as the user base grows.. so does RAM requirement...

Ta
Senad
[EMAIL PROTECTED] wrote:
Hi
I have been playing with trying to get voicemail.conf from DB, I am
using cvs-head, but when I start asterisk, it dies a horrible death,
because it cant load any voicemil setting.
I looked at my mysql logs to see what query was being sent, and I get
SELECT category, var_name, var_val, cat_metric FROM voicemail_users
WHERE filename='voicemail.conf' and commented=0 ORDER BY filename,
cat_metric desc, var_metric asc, category, var_name, var_val, id;
which is in no way related to the table I created following the
example on the wiki.
Is there another table format , I have

+-+---+--+-+-++
Field   | Type  | Null | Key | Default | Extra  |

+-+---+--+-+-++
uniqueid| int(11)   |  | PRI | NULL| auto_increment |
customer_id | int(11)   |  | | 0   ||
context | varchar(50)   |  | | ||
mailbox | int(5)|  | MUL | 0   ||
password| varchar(4)|  | | 0   ||
fullname| varchar(50)   |  | | ||
email   | varchar(50)   |  | | ||
pager   | varchar(50)   |  | | ||
stamp   | timestamp(14) | YES  | | NULL||

+-+---+--+-+-++

tks
Iqbal
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RE: [Asterisk-Users] Asterisk@home backup/restore question

2005-05-16 Thread Irakli Natsvlishvili

As far as I understand, it is in .gz file. Based on what schedule backup is
performed? Is it changeable? How to set more then one backup destination?
For example, over the net? How do I selectively restore files? For example,
I do want to restore only sip.conf and leave everything? Do I have to do it
manually?


I.N. 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Manny A. Wise
Sent: Sunday, May 15, 2005 1:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] backup/restore question

@home do that for you everyday...;)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irakli
Natsvlishvili
Sent: Sunday, May 15, 2005 2:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] [EMAIL PROTECTED] backup/restore question

Hello,

How do I routinely backup all necessary configuration files on [EMAIL 
PROTECTED] Is
there any procedure/tool/script for it? And if I need to move * with
existing configuration on a new hardware, what is the best way to do it? 

Thanks
I.N. 

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Re: [Asterisk-Users] zttest

2005-05-16 Thread Damian Funnell

Rich, did you check IRQ's?  Our zttest results didn't improve markedly 
when we did either change (IRQ's or H/T), but the problem went away 
regardless.

Other than that Digium have recommended throwing out our SCSI320 RAID 
hardware and replacing it with IDE (i.e. not SATA) kit, although 
thankfully we haven't had to make this retarded change (yet).

I would also recommend trying to disable H/T in the BIOS (rather than 
via software) as I wonder if H/T still runs on your box (but is not 
accessed by the OS), so may still be causing you grief.

D.
FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz

Rich Adamson wrote:
On May 16, 2005, at 14:37, Rich Adamson wrote:
   

Hi Waldo, it really depends on who you ask - Digium say that anything
less than 99.99% is going to result in problems, but ours  
regularly runs
at around 99.98% and we don't have any problems.

One of our boxes was running at around 99.96% and we had major issues
with the voice quality packing up from time to time.  We disabled  
hyper
threading and put the TDM400P on its own IRQ and the results came  
back
up over 99.98% (haven't had any problems since).

   

How do you disable hyper threading (what's the command and where is it
placed)?
 

If this is a Linux box, look at the kernel boot arguments in [lilo| 
grub].conf and append "noht", that disables it. My grub.conf on one  
of my boxes looks like this:

title CentOS (2.4.21-27.0.4.ELsmp)
root (hd0,0)
kernel /vmlinuz-2.4.21-27.0.4.ELsmp ro root=LABEL=/ noht
initrd /initrd-2.4.21-27.0.4.ELsmp.img
   

Thanks, I added the noht, rebooted, and still get zttest results that
consistently at 99.987793%, both before and after the change. Guess 
hyper threading has nothing to do with it on this particular system.

This is a new motherboard and it doesn't indicate anything in the bios
relative to hyper threads either.
Rich
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[Asterisk-Users] Transfer of Calls Between Legacy PBX and Asterisk

2005-05-16 Thread Joan Bautista

Hi, We have a scenario where we receive calls from 2 different places:
1- Avaya IP Office
2- CIC Interactive Intelligence PBX 
and the calls are transfer automatically to an Asterisk Box.

The problem we are experiencing is that more that half of those calls
come with Echo and Jitter. For outbounds calls everything works
perfect.
I'm on RH9 Stable v1.4 Version, using an quadspad TMD400P digium card
and have attached 2 pri lines.
Calls transfers to my Asterisk Box are received directly to my T1 lines.
My server is configure accordingly Digiums standards, however I found
this issue annoying and don't have a clue was going on.
We someone has experienced the same problem I maybe would guide me in
solving this problem I would be more than glad.
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RE: [Asterisk-Users] outlook express intregation

2005-05-16 Thread Florian Overkamp

Hi, 

> -Original Message-
> All of the stuff I've googled for and read on wiki all relate 
> to "Outlook".
> Has anyone been successful in getting "Outlook Express" to do 
> click to dial?

I don't think Outlook Express has any support for that kind of thing at all.
No TAPI hooks in there at least as far as I can tell. 

Florian


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Re: [Asterisk-Users] voicemail.conf from DB

but the other choice is to keep editing the voicemail.conf file, 
everytime I add a new user, which again is not really scalable.

I dont wish to store the voicemail in the DB, just the conf file itself, 
mysql will easily in a clusered scenario support 100K+ entries.

What setup would you recommend
iqbal
Senad J wrote:
In my opinion.. using voicemail from DB is not good
for "scaling up" scenarios... in particular if you need to
provide HA of five nine... 99.999%
The cluster database (well in particular MYSQL puts everything in RAM)... so
as the user base grows.. so does RAM requirement...
Ta
Senad
[EMAIL PROTECTED] wrote:
 

Hi
I have been playing with trying to get voicemail.conf from DB, I am
using cvs-head, but when I start asterisk, it dies a horrible death,
because it cant load any voicemil setting.
I looked at my mysql logs to see what query was being sent, and I get
SELECT category, var_name, var_val, cat_metric FROM voicemail_users
WHERE filename='voicemail.conf' and commented=0 ORDER BY filename,
cat_metric desc, var_metric asc, category, var_name, var_val, id;
which is in no way related to the table I created following the
example on the wiki.
Is there another table format , I have
+-+---+--+-+-++
   

Field   | Type  | Null | Key | Default | Extra  |
 

+-+---+--+-+-++
   

uniqueid| int(11)   |  | PRI | NULL| auto_increment |
customer_id | int(11)   |  | | 0   ||
context | varchar(50)   |  | | ||
mailbox | int(5)|  | MUL | 0   ||
password| varchar(4)|  | | 0   ||
fullname| varchar(50)   |  | | ||
email   | varchar(50)   |  | | ||
pager   | varchar(50)   |  | | ||
stamp   | timestamp(14) | YES  | | NULL||
 

+-+---+--+-+-++
tks
Iqbal
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.
 

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RE: [Asterisk-Users] voicemail.conf from DB

2005-05-16 Thread Senad J

In my opinion.. using voicemail from DB is not good
for "scaling up" scenarios... in particular if you need to
provide HA of five nine... 99.999%

The cluster database (well in particular MYSQL puts everything in RAM)... so
as the user base grows.. so does RAM requirement...

Ta
Senad

[EMAIL PROTECTED] wrote:
> Hi
>
> I have been playing with trying to get voicemail.conf from DB, I am
> using cvs-head, but when I start asterisk, it dies a horrible death,
> because it cant load any voicemil setting.
>
> I looked at my mysql logs to see what query was being sent, and I get
>
> SELECT category, var_name, var_val, cat_metric FROM voicemail_users
> WHERE filename='voicemail.conf' and commented=0 ORDER BY filename,
> cat_metric desc, var_metric asc, category, var_name, var_val, id;
>
> which is in no way related to the table I created following the
> example on the wiki.
>
> Is there another table format , I have
>
> +-+---+--+-+-++
>> Field   | Type  | Null | Key | Default | Extra  |
> +-+---+--+-+-++
>> uniqueid| int(11)   |  | PRI | NULL| auto_increment |
>> customer_id | int(11)   |  | | 0   ||
>> context | varchar(50)   |  | | ||
>> mailbox | int(5)|  | MUL | 0   ||
>> password| varchar(4)|  | | 0   ||
>> fullname| varchar(50)   |  | | ||
>> email   | varchar(50)   |  | | ||
>> pager   | varchar(50)   |  | | ||
>> stamp   | timestamp(14) | YES  | | NULL||
> +-+---+--+-+-++
>
>
> tks
> Iqbal
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[Asterisk-Users] Using PAP2 with g723

2005-05-16 Thread M. Ehsanul Karim

Hello,
  Anyone using Linksys PAP2 NA with g723 with asterisk ? Please let me know if you could make it work.as noramlly the voice quality gets horrible with tversion of g723 for pap2.
 
Thanks.
 
Ehsan___
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Re: [Asterisk-Users] Pass variable to Authenticate?

2005-05-16 Thread Justin B Newman

Dana Olson wrote:
I thought of a way to do it using a list of passwords, but the agent
would only ever be prompted for their password. I won't want that.
Basically, what I want to do is this:
Read in the username, ex. 4567
Read in the password, ex. 1234
Pass 12344567 to the Authenticate command, which checks in the file
for that password.
I can't figure out a way to do it, and it may not even be possible.
Is there another way to do it? Can I somehow utilize the DBget and
DBput commands? I couldn't find much info on it in the wiki, and my
impression is that I have to use only the blacklist or cidname
databases.
Dana:
- Read in the username
- Do a DBGET for the password associated with that username
- Authenticate(password)
Yours,
-jbn
Justin B Newman
BinFone Telecom
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[Asterisk-Users] Asterisk and Avaya 4602 SIP phone

2005-05-16 Thread Yao, Yuanbin









Hi,

 

I have been trying to connect Avaya 4602 SIP phone
to Asterisk, but the phone keeps rebooting after I downloaded the SIP software
(Avaya phone release 050205).  I would like to know if anyone succeeded to
hook up Avaya SIP phone to Asterisk. I appreciate your help!

 

Regards,

 

yyao

 

 






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Re: [Asterisk-Users] zttest

2005-05-16 Thread Jens Vagelpohl

On May 16, 2005, at 19:04, Damian Funnell wrote:
...Jens makes a liar out of me, although I read that the 'noht'  
switch stops the OS from using H/T but doesn't disable it  
completely.  I make no warranties regarding the accuracy of this  
information, though.
OK, let me rephrase it: After using "noht" "top" showed the physical  
number of CPUs again, not double that. That's the one thing I can  
confirm! ;)

jens
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[Asterisk-Users] Outbound Faxes with spandsp

2005-05-16 Thread Matthew

This is what I'm trying to set up:
Incoming:
PRI-->Cisco VG200-->Cisco Call Manager 3.34 -->(over h.323)--> Asterisk Box
Outgoing:
Asterisk Box -->(over h.323)--> Cisco Call Manager 3.34 --> Cisco VG200 --> PRI

The incoming part seems to work great.

The outgoing part does not work quite so well. The pages that go
through end up being of random lengths. From what I've been reading
the most likely cause is frame slips and timing issues. Given the set
up how would I go about correcting the timing. I haven't been able to
find much on how to fix it. I realize that faxing over IP can be
problematic, but why would incoming work, but not outgoing?
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[Asterisk-Users] Re: IAX jitter

2005-05-16 Thread Tony Mountifield

In article <[EMAIL PROTECTED]>,
Steven Langley <[EMAIL PROTECTED]> wrote:
> 
> I have a question regarding IAX jitter. I have 3 users on a LAN dialing into
> a Meetme conference on an Asterisk box which is also hosted on the LAN. I
> have set jitterbuffer = no and tos = lowdelay. Now, for 2 of the users the
> audio is fine, but for the 3rd user there is intermittent break up in the
> audio when they are receiving. I have had a look at "iax2 show channels" and
> for the first 2 users (those with no audio problems), the Jitter is low
> (0006ms), and the Lag is relatively high (00070ms). For the 3rd user (the
> one with audio breaking up), the Jitter is relatively high (0627ms) and the
> Lag is relatively low compared to the others (00012ms).
> 
> All of the users are on the same LAN, so can't quite understand the
> differences.

Are you using ztdummy on kernel 2.6?

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] Error running Make config on Debian Sarge

2005-05-16 Thread Andres Paglayan

follow this link
ignore the German and see the commands
http://www.vonloesch.de/node/17 
for the last part be sure that you modprobe the right driver for your
particular device.
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[Asterisk-Users] mysql debug

2005-05-16 Thread Jim Sturtevant









I’m cdr_mysql for about a month and it appeared to
be working well.  My server has very little volume 10-20 calls an
hour.  Today there appears to be a gap of about 3 hours where no CDR
records were logged to the MySQL database.  I have done a couple of
reloads during that period which appears to have restarted it as it is now
logging CDR records again without needing to shutdown/restart asterisk or the
server.

 

I reviewed the asterisk logs and don’t see any
mention of problems writing to the database.  Which log should I be
looking at?

 

Is there a way to have * log to both MySQL and a .csv file
(as backup), or should I bite the bullet and move to Radius for capturing
billing records?

 

Thanks

 






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Re: [Asterisk-Users] zttest

2005-05-16 Thread Rich Adamson

> On May 16, 2005, at 14:37, Rich Adamson wrote:
> >> Hi Waldo, it really depends on who you ask - Digium say that anything
> >> less than 99.99% is going to result in problems, but ours  
> >> regularly runs
> >> at around 99.98% and we don't have any problems.
> >>
> >> One of our boxes was running at around 99.96% and we had major issues
> >> with the voice quality packing up from time to time.  We disabled  
> >> hyper
> >> threading and put the TDM400P on its own IRQ and the results came  
> >> back
> >> up over 99.98% (haven't had any problems since).
> >>
> >
> > How do you disable hyper threading (what's the command and where is it
> > placed)?
> 
> If this is a Linux box, look at the kernel boot arguments in [lilo| 
> grub].conf and append "noht", that disables it. My grub.conf on one  
> of my boxes looks like this:
> 
> title CentOS (2.4.21-27.0.4.ELsmp)
>  root (hd0,0)
>  kernel /vmlinuz-2.4.21-27.0.4.ELsmp ro root=LABEL=/ noht
>  initrd /initrd-2.4.21-27.0.4.ELsmp.img

Thanks, I added the noht, rebooted, and still get zttest results that
consistently at 99.987793%, both before and after the change. Guess 
hyper threading has nothing to do with it on this particular system.

This is a new motherboard and it doesn't indicate anything in the bios
relative to hyper threads either.

Rich


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[Asterisk-Users] Pass variable to Authenticate?

2005-05-16 Thread Dana Olson

I'm trying to figure out a way to make my own agent login, because I
don't like how the default works.

I have the login and logout working fine using the dynamic add and
remove commands, but I need to be able to create a list of users and
passwords.

I thought of a way to do it using a list of passwords, but the agent
would only ever be prompted for their password. I won't want that.

Basically, what I want to do is this:

Read in the username, ex. 4567
Read in the password, ex. 1234
Pass 12344567 to the Authenticate command, which checks in the file
for that password.

I can't figure out a way to do it, and it may not even be possible.

Is there another way to do it? Can I somehow utilize the DBget and
DBput commands? I couldn't find much info on it in the wiki, and my
impression is that I have to use only the blacklist or cidname
databases.

Any ideas would be helpful.
--
Dana
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Re: [Asterisk-Users] ShoreTel 210 MGCP phone drops calls with MGCP RSIP

2005-05-16 Thread Duane Cox

RSIP = Restart in Progress.  This message is sent from and endpoint to the 
call-agent when the line is being disabled/reenabled,
and at startup.

I would think that * should tear down any active calls that is going on when a 
RSIP comes in, so from what I am reading this is
correct.  The bigger question is, why is your phone sending RSIP during a call? 
 Maybe your phone isn't 100% setup yet, but enough
to get calls up.

See if you can capture an "mgcp debug" from the * CLI.

Duane


- Original Message - 
From: "Ben Dugdale" <[EMAIL PROTECTED]>
To: 
Sent: Monday, May 16, 2005 12:39 PM
Subject: [Asterisk-Users] ShoreTel 210 MGCP phone drops calls with MGCP RSIP


> I've got a ShoreTel 210 MGCP phone drops calls.  My packet
> capture indicates that the phone may be trying to renew its registration
> with *, but reports Restart Method of Disconnected (frame 2), then *
> seems to take that as a sign that it has lost the connection and closes
> things down.  The phone, meanwhile, seems to think it can continue the
> conversation until a few ICMP "port unreachable" packets set it straight.
>
> I am able to duplicate this on a fairly regular basis and it appears
> that the call is always dropped similarly.  Everything is groovy until
> the first MGCP RSIP during a call.  The phone seems to send these MGCP
> RSIPs (perhaps on a schedule, looks like every 120 seconds) even when
> there is no call taking place.
>
> Is my diagnosis correct?  What can I do about it?
>
> ...Normal RTP, then one final RTP packet and the following...
>
>   1   0.00  192.168.0.5 5004 192.168.1.137 5004 UDP Source port:
> 5004  Destination port: 5004
>   2   0.007970 192.168.1.137 2427 192.168.0.5  2727 MGCP RSIP 14831
> [EMAIL PROTECTED] MGCP 1.0
>   3   0.008096  192.168.0.5 2727 192.168.1.137 2427 MGCP 200 14831 OK
>   4   0.008150  192.168.0.5 2727 192.168.1.137 2427 MGCP RQNT 7
> [EMAIL PROTECTED] MGCP 1.0
>   5   0.008186  192.168.0.5 2727 192.168.1.137 2427 MGCP AUEP 8
> [EMAIL PROTECTED] MGCP 1.0
>   6   0.008267  192.168.0.5 2727 192.168.1.137 2427 MGCP DLCX 9
> [EMAIL PROTECTED] MGCP 1.0
>   7   0.009226  192.168.0.5 4569 192.168.1.135 4569 IAX2 IAX, source
> call# 2, timestamp 61056ms HANGUP
>   8   0.012188 192.168.1.135 4569 192.168.0.5  4569 IAX2 IAX, source
> call# 10281, timestamp 61056ms ACK
>   9   0.015216 192.168.1.137 5004 192.168.0.5  5004 UDP Source port:
> 5004  Destination port: 5004
>  10   0.015265  192.168.0.5  192.168.1.137  ICMP Destination unreachable
> (Port unreachable)
>  11   0.029923 192.168.1.137 2427 192.168.0.5  2727 MGCP 200 7 OK
>  12   0.030090  192.168.0.5 2727 192.168.1.137 2427 MGCP RQNT 10
> [EMAIL PROTECTED] MGCP 1.0
>  13   0.035268 192.168.1.137 5004 192.168.0.5  5004 UDP Source port:
> 5004  Destination port: 5004
>  14   0.035306  192.168.0.5  192.168.1.137  ICMP Destination unreachable
> (Port unreachable)
>  15   0.050434 192.168.1.137 2427 192.168.0.5  2727 MGCP 200 8 OK
>  16   0.055251 192.168.1.137 5004 192.168.0.5  5004 UDP Source port:
> 5004  Destination port: 5004
>  17   0.055282  192.168.0.5  192.168.1.137  ICMP Destination unreachable
> (Port unreachable)
>  18   0.077638 192.168.1.137 2427 192.168.0.5  2727 MGCP 250 9
> Connection was deleted
>  19   0.100808 192.168.1.137 2427 192.168.0.5  2727 MGCP 200 10 OK
>
> rtp.conf:
>
> [general]
> rtpstart=5004
> rtpend=5005
>
> mgcp.conf:
>
> [general]
> port = 2727
> bindaddr = 0.0.0.0
>
> [192.168.1.137]
> host= 192.168.1.137
> context = home
> callerid= "ShoreTel" <4368>
> dtmfmode= inband
> accountcode = 1000
> amaflags= billing
> callwaiting = no
> callreturn  = no
> cancallforward  = no
> canreinvite = no
> transfer= no
> line => SHOR_001049007E83
>
> -- 
> Ben Dugdale <[EMAIL PROTECTED]>
>
>
>





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Re: [Asterisk-Users] Broadvoice Toll-Free IVR issues

[EMAIL PROTECTED] wrote:
Perhaps someone else has had this issue. I currently have BroadVoice as my
SIP provider with two phone numbers. The first number is my primary local
number; the second is an 877 toll-free number.
The issue is with the IVR abilities. When calling the primary number you
can press 200 at the main menu and receive my extension. When calling the
877 number, 200 will result in “that option is invalid”… Heck, just after
pressing “2” it will give that message. If you press 1 it takes you to the
support queue. The IVR is set up to give you the sales queue after pressing
2, but it doesn’t do that either.
Anyone have any ideas?
 

I remember running into something else that was different for the 
tollfree number with BV. IIRC - the supported codecs differed. It makes 
me think that the tollfree calls come from a different gateway. They 
only allow one SIP device(whether hardware or software) per account 
which rules out the most obvious workaround for the problem. Looks to me 
like dtmf settings are different on the two gateways/routers they are using.

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[Asterisk-Users] voicemail.conf from DB

Hi
I have been playing with trying to get voicemail.conf from DB, I am 
using cvs-head, but when I start asterisk, it dies a horrible death, 
because it cant load any voicemil setting.

I looked at my mysql logs to see what query was being sent, and I get
SELECT category, var_name, var_val, cat_metric FROM voicemail_users 
WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, 
cat_metric desc, var_metric asc, category, var_name, var_val, id;

which is in no way related to the table I created following the example 
on the wiki.

Is there another table format , I have
+-+---+--+-+-++
| Field   | Type  | Null | Key | Default | Extra  |
+-+---+--+-+-++
| uniqueid| int(11)   |  | PRI | NULL| auto_increment |
| customer_id | int(11)   |  | | 0   ||
| context | varchar(50)   |  | | ||
| mailbox | int(5)|  | MUL | 0   ||
| password| varchar(4)|  | | 0   ||
| fullname| varchar(50)   |  | | ||
| email   | varchar(50)   |  | | ||
| pager   | varchar(50)   |  | | ||
| stamp   | timestamp(14) | YES  | | NULL||
+-+---+--+-+-++
tks
Iqbal
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[Asterisk-Users] Asterisk Fax On Demand using SPANDSP?

2005-05-16 Thread mchapman2

Hi-

Does anyone have any dial plan examples pertaining to a Fax On Demand 
using * and Spandsp? 

Thanks,

Mike C.
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Re: [Asterisk-Users] Outgoing spool file ignored

2005-05-16 Thread Andres Paglayan

File::copy does copy, it re-writes the file,
you need to move it.
so when the the pointer is placed the file is already there.
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RE: [Asterisk-Users] IAX jitter

2005-05-16 Thread Marty Mastera




 

  
  
  Steven Langley wrote: 
  




Hi 
there

I have a question regarding IAX 
jitter. I have 3 users on a LAN dialing into a Meetme conference on an 
Asterisk box which is also hosted on the LAN. I have set jitterbuffer = no 
and tos = lowdelay. Now, for 2 of the users the audio is fine, but for the 
3rd user there is intermittent break up in the audio when they 
are receiving. I have had a look at “iax2 show channels” and for the first 2 
users (those with no audio problems), the Jitter is low (0006ms), and the 
Lag is relatively high (00070ms). For the 3rd user (the one with 
audio breaking up), the Jitter is relatively high (0627ms) and the Lag is 
relatively low compared to the others 
(00012ms).

All of the users are on the same 
LAN, so can’t quite understand the differences.

Any explanations / ideas would 
be much welcome.

Thanks

Steven
  I'm having the same issue; I'd love to know how to actually diagnose the 
  issue.I assume there is somewhere on the voip-info.org wiki about 
  this?JD 
I'm in the same boat, although I'm only using IAX2 for connectivity to my 
provider.  The phones themselves are SIP, but some users (really 
should be read all users, but at different times) experience bad audio - 
described as "cutting-out" from the remote party.  My user is 
heard correctly by the remote party but the remote party is bad to my 
user.  Doesn't seem to matter whether it's inbound or 
outbound.  
 
Marty 
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Re: [Asterisk-Users] Error running Make config on Debian Sarge