date:20050117

Oh i forgot to mention
I have found a limitationcalls going through the queue system can NOT 
be parked properly.  More precisely with my stdexten macro and/or the agent 
logic stuff the calls can NOT be rang-back to the original extension.  They 
end up (in my example) in from-sip,s,1 which equates to default,s,1 but 
they have ALL the internal extensions and dial plan.

Why?  Heck if I know.  Somehow the C code loses track of who I'm dialling 
and in 1.0.1 chan_park can't find the origianl extension in the event of a 
timeout.  Yup you could code aroudn this in the dial plan logic by leaving 
some sort of hint, but I don't get why it's missing.

Also don't put a /n at the end of the Dial(Local...) stuff in the 
AgentCallBack macro, it will cause zombies, lots of them, and weird 
behaviour of 7940 and 7960 SIP phones.  Why?  Again, don't know.  I'm 
simply saying 'here there be dragons' and not going in there :)

It DOES work and VERY reliably in practice, just there are the above 
caveats.  Sorry I forgot them in the original message.
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[Asterisk-Users] Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore)

2005-01-17 Thread Ronald Wiplinger

I bought three plus two Grandstream BudgeTone 101 phones.
The shipping cost more than the phone itself from Pulver store.
The first shipping had one phone defect. Nothing on the display. (Can 
happen!)

The second shipment had one phone with a defect display, but it still 
worked.
The second phone's handset was defect too (microphone did not work).
Changing the handset from this one to the other one, "repaired" one of 
the three defect phone sets.

NOW the next question. What is with the warranty?
Jeff Pulver & his team is silent!
In case I do not get the info for the warranty replacements I will 
cancel the credit card for the purchase!

In the meantime I suggest to all of you:
1. Don't buy Grandstream!
2. () !
Ronald
very angry Pulver customer!!!
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[Asterisk-Users] Dial Plan Agents (2 of 2) extensions.com

Attached is the example extensions.conf

extensions.conf
Description: Binary data
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[Asterisk-Users] Dial Plan Agents (1 of 2) agent-dialplan.conf

Well because I had sooo may problems with chan_agent.c I wrote this. I'm 
releasing it under LGPL but if you use it or anything please let me know. 
It'd be interesting if anyone finds this more useful than just a pile of 
junk.

I've included a (working) example extensions file.  SIP phones are assumed 
to have the same identifier as their extension number, but it'd be trivial 
to do lookups against the database system using the AgentGetVar bits and 
ring a given SIP channel.

PRI's are meant to go to from-pri where the ID is interpreted as an 
extension.  Most ring inside lines, but a few specials will get you to an 
IVR or directly to a FAX machine attached to an analog Zap channel.

I havent' included the queue or voicemail confs since that's elementary. 
voicemail must be configured without passwords AND users have to be 
instructed to NOT change the password via VM unless you do more work to fix 
that menu.

PArt of the reason for writing this was another thing that makes me want to 
use a club, the fact that EVERY bloody module has it's own authentication 
bits, so you can't have any shared PINs or anything.  This maddens me to no 
end, because of the complexity to users mostly, and the headache to me. 
Hopefully dev's will fix this somehow.  I don't h ave the clout to do it, 
and probably not the time.

Something like PAM is NEEDED or something that alteast allows ALL the 
modules to SHARE some auth mechanisms and password changing mechs.  Must 
also be available via dial plans like my stuff does.

In any event...I'm releasing this version atleast under the LGPL.
The features demonstrated and used don't have any bugs that I know of, but 
heh, it's only been used and tested in one installation.  Hopefully the 
community will have some feedback for me.

--
Undocumented Features quote of the moment...
"It's not the one bullet with your name on it that you
have to worry about; it's the twenty thousand-odd rounds
labeled `occupant.'"
  --Murphy's Laws of Combat


agent-dialplan.conf
Description: Binary data
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[Asterisk-Users] Re: IAX2 doesn't respect bindaddr?

2005-01-17 Thread Tom Ivar Helbekkmo

I wrote:

> OK, I just [updated].  No change.  :-(

Here's what Asterisk says on startup:

Asterisk CVS-HEAD-01/18/05-07:51:47, Copyright (C) 1999-2004 Digium.
[...]
  == Parsing '/etc/asterisk/iax.conf': Found
  == Using TOS bits 0
  == Binding IAX2 to '193.71.27.8:4569'
  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
  == IAX Ready and Listening on 0.0.0.0 port 4569

-tih
-- 
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
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RE: [Asterisk-Users] Canadian Content: Telus and Shaw...

2005-01-17 Thread Colin Anderson

>I called Telus before Christmas requesting some sort of VOIP connection.

>We are going with babytel.  I'll advise how that works when it is up and
>running, hopefully next week. 

[plug] www.thinktel.ca 

I know the guys they are competent they will sell IAX. Peered thru GT in
Downtown Edmonton. 
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[Asterisk-Users] Auto Protocol (depending upon registration....

2005-01-17 Thread Gary

Hi folks,

I'm sure I had this in a previous life :-)

Basically the ability to dial with autoselection of either IAX2 or SIP
depending upon the registration of the endpoint.

Ok, I have probably missed it in the wiki as well.

hints ?

Gary
.


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[Asterisk-Users] Re: IAX2 doesn't respect bindaddr?

2005-01-17 Thread Tom Ivar Helbekkmo

"Brian West" <[EMAIL PROTECTED]> writes:

> update

OK, I just did.  No change.  :-(

-tih
-- 
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
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[Asterisk-Users] Is anybody using an IAXy?

2005-01-17 Thread Ronald Wiplinger

(second try !)
Can anybody check my settings below. I doubt most settings in the 
extensions.conf.
I get a dialtone if I pick up the phone!


I have provisioned with iaxy.conf:
;
; IAXY Provisioning description
;
dhcp
codec: ulaw
server: 61.220.xx.xx
user: aaabbb
pass: cccddd
register
iax.conf:
=
[623]   ; IAXy
type=friend
host=dynamic
accountcode=aaabbb
disallow=all
allow=ulaw
secret=cccddd
callerid="IAXy at ELMIT" <623>
trunk=no
extensions.conf:

[inhouse]

PHONE_623=IAX2/aaabbb:[EMAIL PROTECTED]/623; 3 IAXy adapter
exten => 623,1,Dial(${PHONE_623},60,Ttrm)
exten => 623,2,Voicemail,u623
exten => 623,103,Voicemail,b623
[default]
...
include => inhouse   

What do I miss?

bye
Ronald
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Re: [Asterisk-Users] Planning "hotel" phone system - Need input


--On Monday, January 17, 2005 22:20 -0800 [EMAIL PROTECTED] wrote:
<...>
The basic arrangement would be:
Telco <-> T1 <-> Asterisk <-> T1 <-> Channel Bank <-> POTS <-> regular
phone
Here are my questions thus far:
- Firstly, which channel bank would best suit me?  I only need FXS, but
I'll  need Caller ID, Visual Message waiting, and a decent ring
generator.  I haven't  found much info on which FXS channel banks output
FSK, or what loop length they  can tolerate.  ASDI would be nice too, but
not yet a requirement.
- Can someone point me in the direction of sample configs for DID
applications?
Check the asterisk wiki http://www.voip-info.org/  but basically incoming 
lines are tied to a context, with a T1 the phone company usually delivers a 
4 digit DID so you'll have an incoming context with a bunch of 4 digit DIDs 
*or* you can do something more complex and do a lookup to some sort of 
database, which for you might be ideal since you want things more dynamic. 
extension changes on the fly are definitely doable, reload reconfigures the 
extensions list without disturbing in-flight calls.

- What size server will I need?  Assume for now a pair of quad-T1 cards,
2 T1s  incoming, and 5 channel banks.  Shouldn't require much horsepower
since it's T1  <-> T1 switching
Without transcoding you'll be able to get by with almost any mid to high 
end P4 Xeon (not celeron) system.  Go for the 1MB cache as the Pentiums 
really suffer badly without the extra cache in my experience, and the price 
jump is not that much.  go to a 2.8ghz and 1mb or a 3.0 ghz and 1mb instead 
of a higher clock rate.

You might want to consider one of the other options rather than the Digium 
hardware...someone had a link to some hardware that had onboard DSPs and 
was very similar to the Digium hardware otherwise.  This would certainly 
make a solution much more scaleable.  That said I think you'll be fine with 
2xQuad PRIs in the same box, and if you're worried though it'd be simple to 
split the Quad PRIs to two different boxes, then have them handoff to 
eachother only when necessary.

I'm posting probably later tonight a few bits of framework to 
asterisk-users here that I used to make a dialplan based 'agent' system 
since chan_agent was horribly and uselessly broken for me.  I'm still 
running 1.0/1.0.1 so I dont' know what's been fixed in 1.0.3 or 1.0.4 -- 
nor will i find out anytime quite soon, everything's in production now and 
i can't afford to experiment for lack of time at the moment.

- Can I get my NMS AG-T1/E1 card working with Asterisk for tinkering
purposes?
Uhm, probably not, but someone else can better answer this.
- What kind of uptime am I going to expect on Asterisk?  Am I going to
have to  reboot the server every 2 weeks?  Can I hope for carrier class
service?
I'm having problems with 1.0 and 1.0.1 on the analog TDM ports after about 
a week to two weeks, they stop ringing.  However I'm pretty sure there are 
patches in that fix this as of 1.0.2.  I haven't had time to investigate it 
nor upgrade but scheduling a late night reboot after checking for live 
channels has me fixed for the time being.  The T1 card has been flawless 
since we put it in.

- Is it possible to change extensions on the fly?  For example, this week
555-  rings in unit #10.  Next week theres a different tenant, so i
want to make  555- ring in unit #10, and send 555- to voicemail.
Some kind of GUI to  accomplish this would also be nice.
See my above noteI am not aware of anything exists out of the box, but 
there are certainly those whose services could be contracted to produce 
this.

Thanks in advance for your responses
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[Asterisk-Users] Planning "hotel" phone system - Need input

2005-01-17 Thread tech

Ok, I'm working on an implementation of Asterisk to service approximately 50 
fractional (read: timeshare) residences.  Basically what I'm starting with is a 
hotel phone system, but with additional functionality that Asterisk can 
provide.  From the end user's perspective, I want the exact same functionality 
as the Telco provides (Caller ID, visual message waiting, etc) with a few minor 
modifications to the dialing plan.  Most calls will be transparent, but I want 
the ability to call the concierge, etc.  On the Management side, we'll probably 
have to integrate the configuration and call records into our property 
manangement software.

My base requirements are 2 lines per unit minimum (voice/fax) with standard TDM 
POTS distribution due to the geographic dispersity.  On the Telco side I'm 
thinking T1 DID w/ outgoing, as we'd like to have independant numbers for each 
unit, but not have to pay for 100 trunks.

The basic arrangement would be:
Telco <-> T1 <-> Asterisk <-> T1 <-> Channel Bank <-> POTS <-> regular phone

Here are my questions thus far:

- Firstly, which channel bank would best suit me?  I only need FXS, but I'll 
need Caller ID, Visual Message waiting, and a decent ring generator.  I haven't 
found much info on which FXS channel banks output FSK, or what loop length they 
can tolerate.  ASDI would be nice too, but not yet a requirement.

- Can someone point me in the direction of sample configs for DID applications?

- What size server will I need?  Assume for now a pair of quad-T1 cards, 2 T1s 
incoming, and 5 channel banks.  Shouldn't require much horsepower since it's T1 
<-> T1 switching

- Can I get my NMS AG-T1/E1 card working with Asterisk for tinkering purposes?

- What kind of uptime am I going to expect on Asterisk?  Am I going to have to 
reboot the server every 2 weeks?  Can I hope for carrier class service?

- Is it possible to change extensions on the fly?  For example, this week 555-
 rings in unit #10.  Next week theres a different tenant, so i want to make 
555- ring in unit #10, and send 555- to voicemail.  Some kind of GUI to 
accomplish this would also be nice.

Thanks in advance for your responses

~Rick

---
Rick Jonker
Systems Administrator
The Falls Resort Community
www.thefalls.ca
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[Asterisk-Users] 2nd try Mediatrix 1204

2005-01-17 Thread Gonzalo Gasca Meza

Hi everybody,
I have setup a Mediatrix 1204, the calls worked fine, both incoming and outgoing.
The problem here is the delay.
When I do a call to the PSTN or receive a call from the PSTN exists a delay of 4 seconds after answer or sending the call.
For OUTGOING
My Dialplan for the Mediatrix box is the following, here at Mexico we use 8 digits for local calls.
([1-9]xxx|01xx||060|0xx)
I have verified that inmediatly after I dial from my IP phone, the in-use light turns on in Mediatrix but the call is not pass until the 4 seconds timer expires.
I have tried disabling the Dial plan but it didnt help
Form Mediatrix documentation
The Timer is set to 4 seconds. It can be used to indicate that if users have not dialed a digit for 4 seconds, it is likely that they have finished dialing and the gateway can make the call. A Dial Map for this could be:
[2-9]xxT
FOR INCOMING
The same 4 seconds delay after the call is sent to Asterisk.
The problem here, is that despite we answer or not the call, once the call is sent to Mediatrix, the calling party hear 2 ring-back tones generated by Mediatrix, then the ringback for Asterisk
Once the call is passed to Asterisk and starts ringing, if we call from a cell phone,home or office the call is marked as answered and the call timer starts no matter if is answered or not.
Any ideas?
I have tried sending the # at the end with no success.
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[Asterisk-Users] Sound quality - commercial vs. Asterisk




So far in my playing with Asterisk I've messed with 
soft phones (x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters 
(Grandstream 286, Digium IAXy).
 
I've also got a Vonage line, using a Linksys 
ATA.
 
None of the devices I've connected to my Asterisk 
server have been able to maintain the same consistent sound quality over a long 
distance as the Vonage line.    Don't get me wrong, the 
Grandstreams are actually not too bad, but there is still some breakups that can 
be annoying.
 
Meanwhile the Vonage ATA maintains an almost 
flawless connection, all the time.
 
I'm assuming (perhaps wrongly?) that the Linksys 
ATA that Vonage uses is still using SIP with some standardized codec.  If 
that assumption is correct, then how the heck to they manage to get the 
consistent connection quality?  Is it just a matter of the right setting 
tweaks within Asterisk and/or the SIP devices?
 
I don't think it's a question of Asterisk hardware, 
since if I connect via local network to the Asterisk server with a SIP device 
the quality is pretty consistent.   It's generally when remotely 
connecting that I have the inconsistent sound quality.  This would lead me 
to believe that it's a matter of tweaking something to deal with latency or 
packet dropping issues (?).
 
What has Vonage got figured out that I still need 
to?  Any comments would be appreciated...
 
regards,
 
Paul
 
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[Asterisk-Users] VoIP Routes and Terminations

2005-01-17 Thread lonnie

Hello All,

I have been trying to do a lot of reading and researching while still
being very new to the VoIP arena so I will appreciata any and all help
that you may wish to provide.

In my research I have come actross the idea of VoIP routes and
terminations but am not exactly clear on them. Could some one please
comment more?

There seems to be a internet traffic with various VoIP services talking
about how people can start their own business and set up route and
terminations.

My goal is to set up an online VoIP service like this one that I have just
found called Skype but not as large of course.

I do plan to pick up the Asterisk development kit to get a feel for
things, but what else do one need to get the basic business up and
running?

I wish to find a Howto on the basic equipment, software, some initial
expenses even for a small operation, Trabas billing, Asterisk engine, and
I would guess that I would have to get some connections with existing
services so that I could offer national and international rates as well.

Thanks for all of the help so far and I will keep reading and trying to
put together my game plan.

Regards,
Lonnie


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Re: [Asterisk-Users] I Don't Want Asterisk in the Media Path

2005-01-17 Thread Dhennys Pestana

Thank you all for replies and comments. I finally figured out what was
happening.

I was performing these tests using Xten Pro. There was an option enabled under
Codec Order on Xten preferences.

For some reason when "Use Remote Preferred Coded as Local Preferred Codec"
option is enabled, Asterisk will ALWAYS be kept in the middle of RTP traffic
regardless of parameters set up on Dial() string or codecs enabled for each call
leg.

Thanks again,

-Dhennys



- Original Message - 
From: "Eric Wieling aka ManxPower" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, January 14, 2005 16:56
Subject: Re: [Asterisk-Users] I Don't Want Asterisk in the Media Path


| Dhennys Pestana wrote:
| > I'm trying to find a way to connect two (or more) extensions directly
without
| > being kept in the middle during the conversation but it won't happen.
|
| Asterisk will always stay in the SIP signaling path.  It can get out of
| the RTP path (only way to really see this is using something like
| tcpdump since sip show channels shows the signaling not the RTP path).
| Asterisk CANNOT get out of the RTP path if you are using the "t" or "T"
| option to dial (maybe other options too) or if the codec for the two
| legs of the call are different.
|


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Re: [Asterisk-Users] internal dial tone on password from outside

> > exten => s,1,Authenticate(X)
> > exten => s,2,DISA,no-password|local
> > 
> > Can someone explain to me what passcode is used for?
> > 
> > If I enter "no-password" I can make a call but if I enter any number
> > instead of word "passcode" it will not let me IN.
> > Is passcode a second level password; the asterisk is not prompting me to
> > enter anything.
> > 
> 
> The passcode is so that you don't need the Authenticate line before it.

If I comment-out the Authenticate line and enter a number in place of
"no-password"; asterisk is not prompting me to enter that number.  Even
if I enter that number it will not let me IN.

How do you use it?
Does it need to be used in connection with a file?

-- 
#Joseph
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Re: [Asterisk-Users] Wait(n) -v- Background(silence/n) ?

2005-01-17 Thread Steven Critchfield

On Tue, 2005-01-18 at 10:44 +1100, Howard Lowndes wrote:
> Will Wait(n) still listen for DTMF input from the caller after there has
> been a Background(some-message) prompt, or do I need to use
> Background(silence/n) to still listen for DTMF?

You don't need anything but a proper gap. You need to program the
extensions like you do with a event loop. 

exten => s,1,Wait,0
exten => s,2,Answer
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => s,5,BackGround,demo-congrats

; This is a blank area that just waits to get DTMF for up to 10 
; seconds due to the ResponseTimeout

exten => t,1,Goto(somewhere-due-to-timeout)


-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] On Hold music

Computer Onsite Support wrote:
You right the sound card is NOT necessary like you just said. I mean when I
dial extension 216 I should be able to hear the on-hold music but is not
working on a new machine I want to put in production but it works on a PIII
500Mh that I want to retire.
And you have the m option in the dial command?
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] internal dial tone on password from outside

Joseph wrote:
On Mon, 2005-01-17 at 21:43 -0500, Brian Dingman wrote:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA

Thank you!
DISA (Direct Inward System Access) - that is what I need.
DISA,passcode|context 

exten => s,1,Authenticate(X)
exten => s,2,DISA,no-password|local
Can someone explain to me what passcode is used for?
If I enter "no-password" I can make a call but if I enter any number
instead of word "passcode" it will not let me IN.
Is passcode a second level password; the asterisk is not prompting me to
enter anything.
The passcode is so that you don't need the Authenticate line before it.
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] On Hold music

Computer Onsite Support wrote:
Please be more specific regarding symbolic link of mpg321 so I can
troubleshoot it myself. The strength thing is that I tried this in three
other different computers and can't get it to work using same installation
guide was able to get it to run on a PIII 500 which I want to get rid of it
now.
Just run mpg123
the result should be:
[EMAIL PROTECTED] ~]# mpg123
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp.
Uses code from various people. See 'README' for more!
THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] transfers with zap channel




Ah, suddenly everything becomes clear.   
I've never looked in features.conf before.  I now understand that 700 is 
supposed to intitiate the call park, and it's taking precidence over the 
extension I was trying to dial of 7007.  I've changed the call parking 
extension and now I can do regular attended and unattended transfers without 
having to park the call... 
(note to anyone else changing features.conf, you 
have to 'restart' asterisk, a 'reload' won't do).
 
thanks a bunch for the help, guys...
 
Paul

  - Original Message - 
  From: 
  Lyle 
  Giese 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Monday, January 17, 2005 8:20 
  PM
  Subject: Re: [Asterisk-Users] transfers 
  with zap channel
  
  Have you looked at features.conf?
   
  Lyle
  
- Original Message - 
From: 
Paul 
Fielding 
To: Asterisk Users Mailing List - 
Non-Commercial Discussion 
Sent: Monday, January 17, 2005 8:53 
PM
Subject: Re: [Asterisk-Users] transfers 
with zap channel

The outside line isn't actually being dropped - 
the outside line hanging up is me hanging up the outside line after finding 
that my transfer failed.
 
I must be not understanding how the flash-hook 
works then.  My understanding was that when I flash-hook and get a 
second dialtone I should be able to dial the extention I want to reach (7007 
is another extension, via SIP).  Normally, if I pick up the analog 
phone and dial 7007 it rings the extention fine.   Apparently, 
though, when you get that second dialtone, it has different 
rules?   I haven't been able to find documentation on this, can it 
be found anywhere?  For example, why does dialing 700 park the 
call?  I haven't found anything on this... *shrug*...
 
Paul
 

  - Original Message - 
  From: 
  Lyle 
  Giese 
  To: Asterisk Users Mailing List 
  - Non-Commercial Discussion 
  Sent: Monday, January 17, 2005 7:22 
  PM
  Subject: Re: [Asterisk-Users] 
  transfers with zap channel
  
  How long between getting parked is the 
  orginal call dropping?  
   
  Depending on your dialplan, yes dialing 700x 
  will almost immediately send the call to call parking. (IMHO, poor 
  extension planning can also cause this.)
   
  I don't use the t or T 
  options.  IMHO, you just lose the ability to use the # 
  key and confused the heck out of my users.  Took it out and use the 
  flash method only in my dial plan.  Dial 700, park the call.  
  Dial the other extension, tell them to pick up 701.  Or use meetme 
  for conference calling?
   
  I know I need to play with three way calling 
  here also.
   
  Lyle
   
  
- Original Message - 
From: 
Paul 
Fielding 
To: Asterisk Users Mailing 
List - Non-Commercial Discussion 
Sent: Monday, January 17, 2005 6:12 
PM
Subject: [Asterisk-Users] transfers 
with zap channel

Ok, I've seen discussion before on doing 
transfers (attended and unattended), there seems to be much confusion 
over it.
 
As things sit, I've been trying 
(unsuccessfully) to do transfers with a zap channel (analog phone 
attached to TDM400).  I have no idea what I'm missing.  My 
current understanding is that I need to have transfer=yes in 
zapata.conf, do a flash hook, dial the 2nd number, flash hook again and 
we're linked (attended).   Then if I hang up the call will be 
transfered.
 
However, when I try to do this things don't 
work.   Here's what I do:
 
- connection is made between Zap/3 (analog 
phone) and Zap/1 (outside line).
- flash hook to get dialtone (I do get 
dialtone)
- attempt to transfer to extension 7007 - I 
dial 7007
- after dialing the 2nd zero, and before 
dialing the 2nd seven, I hear Asterisk announce (seven - zero - one) and 
then Zap/3 is hung up (I get a busy signal).  Zap/1 gets 
parked.
 
Here's what the log shows:
 
    -- Zap/1-1 answered 
Zap/3-1    -- Attempting native bridge of Zap/3-1 and 
Zap/1-1    -- Started three way call on channel 
1    -- Started music on hold, class 'default', on 
Zap/3-1    -- Attempting native bridge of Zap/3-1 and 
Zap/1-1    -- Starting simple switch on 
'Zap/1-2'    -- Started music on hold, class 
'default', on Zap/3-1  == Parked Zap/3-1 on 701. Will timeout 
back to dostuff,7001,1 in 45 seconds    -- Added 
extension '701' priority 1 to parkedcalls    -- 
Playing 'digits/7' (language 'en')    -- Hungup 
'Zap/1-1'  == Spawn extens

[Asterisk-Users] Granstream Phone "Login incorrect message"

Can anybody send me a web screen shot of their Granstream Phone
configuration so  I can figure out why I'm NOT able to check voice mail
throw my granstream Phone.
I Appreciate in advance.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matt
Riddell
Sent: Monday, January 17, 2005 10:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] On Hold music


Computer Onsite Support wrote:
> Thanks you but that didn't work. Any other solutions?

Make sure you are using the correct version of mpg123...

I think from memory the correct version is 0.59r

Also, I think Redhat has simply made a symlink to mpg321 (which is *not*
the same).

--
Cheers,

Matt Riddell
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Re: [Asterisk-Users] internal dial tone on password from outside

On Mon, 2005-01-17 at 21:43 -0500, Brian Dingman wrote:
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA

Thank you!
DISA (Direct Inward System Access) - that is what I need.

DISA,passcode|context 

exten => s,1,Authenticate(X)
exten => s,2,DISA,no-password|local

Can someone explain to me what passcode is used for?

If I enter "no-password" I can make a call but if I enter any number
instead of word "passcode" it will not let me IN.
Is passcode a second level password; the asterisk is not prompting me to
enter anything.

-- 
#Joseph
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RE: [Asterisk-Users] On Hold music

You right the sound card is NOT necessary like you just said. I mean when I
dial extension 216 I should be able to hear the on-hold music but is not
working on a new machine I want to put in production but it works on a PIII
500Mh that I want to retire.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jerry
Rasmussen
Sent: Monday, January 17, 2005 11:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] On Hold music


This may sound kind of crazy and I maybe missing something.  But are you
placing the call on hold so you can hear the hold music.  This may not
be the case but you may have to place the call on hold to here the
music.

Also you mentioned sound, you do not need a sound card in the asterisk
box to use this hold music feature.

Hope this helps.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Computer
Onsite Support
Sent: Monday, January 17, 2005 10:51 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] On Hold music

Please be more specific regarding symbolic link of mpg321 so I can
troubleshoot it myself. The strength thing is that I tried this in three
other different computers and can't get it to work using same
installation guide was able to get it to run on a PIII 500 which I want
to get rid of it now.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matt
Riddell
Sent: Monday, January 17, 2005 10:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] On Hold music


Computer Onsite Support wrote:
> Thanks you but that didn't work. Any other solutions?

Make sure you are using the correct version of mpg123...

I think from memory the correct version is 0.59r

Also, I think Redhat has simply made a symlink to mpg321 (which is *not*
the same).

--
Cheers,

Matt Riddell
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RE: [Asterisk-Users] On Hold music

2005-01-17 Thread Jerry Rasmussen

This may sound kind of crazy and I maybe missing something.  But are you
placing the call on hold so you can hear the hold music.  This may not
be the case but you may have to place the call on hold to here the
music.

Also you mentioned sound, you do not need a sound card in the asterisk
box to use this hold music feature.  

Hope this helps. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Computer
Onsite Support
Sent: Monday, January 17, 2005 10:51 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] On Hold music

Please be more specific regarding symbolic link of mpg321 so I can
troubleshoot it myself. The strength thing is that I tried this in three
other different computers and can't get it to work using same
installation guide was able to get it to run on a PIII 500 which I want
to get rid of it now.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matt
Riddell
Sent: Monday, January 17, 2005 10:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] On Hold music


Computer Onsite Support wrote:
> Thanks you but that didn't work. Any other solutions?

Make sure you are using the correct version of mpg123...

I think from memory the correct version is 0.59r

Also, I think Redhat has simply made a symlink to mpg321 (which is *not*
the same).

--
Cheers,

Matt Riddell
___

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RE: [Asterisk-Users] On Hold music

Please be more specific regarding symbolic link of mpg321 so I can
troubleshoot it myself. The strength thing is that I tried this in three
other different computers and can't get it to work using same installation
guide was able to get it to run on a PIII 500 which I want to get rid of it
now.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matt
Riddell
Sent: Monday, January 17, 2005 10:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] On Hold music


Computer Onsite Support wrote:
> Thanks you but that didn't work. Any other solutions?

Make sure you are using the correct version of mpg123...

I think from memory the correct version is 0.59r

Also, I think Redhat has simply made a symlink to mpg321 (which is *not*
the same).

--
Cheers,

Matt Riddell
___

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Re: [Asterisk-Users] On Hold music

Computer Onsite Support wrote:
Thanks you but that didn't work. Any other solutions?
Make sure you are using the correct version of mpg123...
I think from memory the correct version is 0.59r
Also, I think Redhat has simply made a symlink to mpg321 (which is *not* 
the same).

--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] transfers with zap channel

2005-01-17 Thread Lyle Giese




Have you looked at features.conf?
 
Lyle

  - Original Message - 
  From: 
  Paul 
  Fielding 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Monday, January 17, 2005 8:53 
  PM
  Subject: Re: [Asterisk-Users] transfers 
  with zap channel
  
  The outside line isn't actually being dropped - 
  the outside line hanging up is me hanging up the outside line after finding 
  that my transfer failed.
   
  I must be not understanding how the flash-hook 
  works then.  My understanding was that when I flash-hook and get a second 
  dialtone I should be able to dial the extention I want to reach (7007 is 
  another extension, via SIP).  Normally, if I pick up the analog phone and 
  dial 7007 it rings the extention fine.   Apparently, though, when 
  you get that second dialtone, it has different rules?   I haven't 
  been able to find documentation on this, can it be found anywhere?  For 
  example, why does dialing 700 park the call?  I haven't found anything on 
  this... *shrug*...
   
  Paul
   
  
- Original Message - 
From: 
Lyle 
Giese 
To: Asterisk Users Mailing List - 
Non-Commercial Discussion 
Sent: Monday, January 17, 2005 7:22 
PM
Subject: Re: [Asterisk-Users] transfers 
with zap channel

How long between getting parked is the orginal 
call dropping?  
 
Depending on your dialplan, yes dialing 700x 
will almost immediately send the call to call parking. (IMHO, poor extension 
planning can also cause this.)
 
I don't use the t or T 
options.  IMHO, you just lose the ability to use the # 
key and confused the heck out of my users.  Took it out and use the 
flash method only in my dial plan.  Dial 700, park the call.  Dial 
the other extension, tell them to pick up 701.  Or use meetme for 
conference calling?
 
I know I need to play with three way calling 
here also.
 
Lyle
 

  - Original Message - 
  From: 
  Paul 
  Fielding 
  To: Asterisk Users Mailing List 
  - Non-Commercial Discussion 
  Sent: Monday, January 17, 2005 6:12 
  PM
  Subject: [Asterisk-Users] transfers 
  with zap channel
  
  Ok, I've seen discussion before on doing 
  transfers (attended and unattended), there seems to be much confusion over 
  it.
   
  As things sit, I've been trying 
  (unsuccessfully) to do transfers with a zap channel (analog phone attached 
  to TDM400).  I have no idea what I'm missing.  My current 
  understanding is that I need to have transfer=yes in zapata.conf, do a 
  flash hook, dial the 2nd number, flash hook again and we're linked 
  (attended).   Then if I hang up the call will be 
  transfered.
   
  However, when I try to do this things don't 
  work.   Here's what I do:
   
  - connection is made between Zap/3 (analog 
  phone) and Zap/1 (outside line).
  - flash hook to get dialtone (I do get 
  dialtone)
  - attempt to transfer to extension 7007 - I 
  dial 7007
  - after dialing the 2nd zero, and before 
  dialing the 2nd seven, I hear Asterisk announce (seven - zero - one) and 
  then Zap/3 is hung up (I get a busy signal).  Zap/1 gets 
  parked.
   
  Here's what the log shows:
   
      -- Zap/1-1 answered 
  Zap/3-1    -- Attempting native bridge of Zap/3-1 and 
  Zap/1-1    -- Started three way call on channel 
  1    -- Started music on hold, class 'default', on 
  Zap/3-1    -- Attempting native bridge of Zap/3-1 and 
  Zap/1-1    -- Starting simple switch on 
  'Zap/1-2'    -- Started music on hold, class 'default', 
  on Zap/3-1  == Parked Zap/3-1 on 701. Will timeout back to 
  dostuff,7001,1 in 45 seconds    -- Added extension 
  '701' priority 1 to parkedcalls    -- Playing 
  'digits/7' (language 'en')    -- Hungup 
  'Zap/1-1'  == Spawn extension (dostuff, 7001, 1) exited non-zero 
  on 'Parked/Zap/3-1'    -- Stopped music 
  on hold on Parked/Zap/3-1    -- Playing 
  'digits/0' (language 'en')    -- Playing 'digits/1' 
  (language 'en')    -- Parking call to 
  'Zap/1-2'    -- Hungup 'Zap/1-2'    
  -- Stopped music on hold on Zap/3-1  == Zap/3-1 got tired of 
  being parked    -- Hungup 'Zap/3-1'
   
  I'm not sure what I'm missing.  
  Apparently something to do with parked calls, so I must be 
  misunderstanding how do to the call transfer.
   
  I've also tried enabling Asterisk transfer on 
  the channel (exten => 7010,1,Dial(${CORDLESS},20,tT)).
  My understanding of this method is that this 
  allows one to hit the pound (#) to start a transfer.  Yet pound does 
  nothing.  Is it fair to assume that the tT only works on SIP 
  channels, or am I missing something else.
   
  Any help is much appreciated

RE: [Asterisk-Users] On Hold music

Thanks you but that didn't work. Any other solutions?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Joseph
Sent: Monday, January 17, 2005 9:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] On Hold music


On Mon, 2005-01-17 at 20:53 -0500, Computer Onsite Support wrote:
> Can anyone of you help me out with this issue. My Asterisk is working
> fine except my music-on-hold will NOT work even though I just retry
> three different other machines with different board and sound.

You don't need any sound card for musing on hold to work.
Loot at this webpage it will guide you:
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf

uncomment in musiconhold.conf one of the line:
[classes]
;default => quietmp3:/var/lib/asterisk/mohmp3
loud => mp3:/var/lib/asterisk/mohmp3
;random => quietmp3:/var/lib/asterisk/mohmp3,-z

specify in you extension.conf what you want to play, example:
[office-open]
exten => s,1,Wait(2)
exten => s,2,Answer()
exten => s,3,SetMusicOnHold(loud)

-- 
#Joseph
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Re: [Asterisk-Users] transfers with zap channel




The outside line isn't actually being dropped - the 
outside line hanging up is me hanging up the outside line after finding that my 
transfer failed.
 
I must be not understanding how the flash-hook 
works then.  My understanding was that when I flash-hook and get a second 
dialtone I should be able to dial the extention I want to reach (7007 is another 
extension, via SIP).  Normally, if I pick up the analog phone and dial 7007 
it rings the extention fine.   Apparently, though, when you get that 
second dialtone, it has different rules?   I haven't been able to find 
documentation on this, can it be found anywhere?  For example, why does 
dialing 700 park the call?  I haven't found anything on this... 
*shrug*...
 
Paul
 

  - Original Message - 
  From: 
  Lyle 
  Giese 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Monday, January 17, 2005 7:22 
  PM
  Subject: Re: [Asterisk-Users] transfers 
  with zap channel
  
  How long between getting parked is the orginal 
  call dropping?  
   
  Depending on your dialplan, yes dialing 700x will 
  almost immediately send the call to call parking. (IMHO, poor extension 
  planning can also cause this.)
   
  I don't use the t or T 
  options.  IMHO, you just lose the ability to use the # key 
  and confused the heck out of my users.  Took it out and use the flash 
  method only in my dial plan.  Dial 700, park the call.  Dial the 
  other extension, tell them to pick up 701.  Or use meetme for conference 
  calling?
   
  I know I need to play with three way calling here 
  also.
   
  Lyle
   
  
- Original Message - 
From: 
Paul 
Fielding 
To: Asterisk Users Mailing List - 
Non-Commercial Discussion 
Sent: Monday, January 17, 2005 6:12 
PM
Subject: [Asterisk-Users] transfers 
with zap channel

Ok, I've seen discussion before on doing 
transfers (attended and unattended), there seems to be much confusion over 
it.
 
As things sit, I've been trying 
(unsuccessfully) to do transfers with a zap channel (analog phone attached 
to TDM400).  I have no idea what I'm missing.  My current 
understanding is that I need to have transfer=yes in zapata.conf, do a flash 
hook, dial the 2nd number, flash hook again and we're linked 
(attended).   Then if I hang up the call will be 
transfered.
 
However, when I try to do this things don't 
work.   Here's what I do:
 
- connection is made between Zap/3 (analog 
phone) and Zap/1 (outside line).
- flash hook to get dialtone (I do get 
dialtone)
- attempt to transfer to extension 7007 - I 
dial 7007
- after dialing the 2nd zero, and before 
dialing the 2nd seven, I hear Asterisk announce (seven - zero - one) and 
then Zap/3 is hung up (I get a busy signal).  Zap/1 gets 
parked.
 
Here's what the log shows:
 
    -- Zap/1-1 answered 
Zap/3-1    -- Attempting native bridge of Zap/3-1 and 
Zap/1-1    -- Started three way call on channel 
1    -- Started music on hold, class 'default', on 
Zap/3-1    -- Attempting native bridge of Zap/3-1 and 
Zap/1-1    -- Starting simple switch on 
'Zap/1-2'    -- Started music on hold, class 'default', 
on Zap/3-1  == Parked Zap/3-1 on 701. Will timeout back to 
dostuff,7001,1 in 45 seconds    -- Added extension '701' 
priority 1 to parkedcalls    -- Playing 'digits/7' 
(language 'en')    -- Hungup 'Zap/1-1'  == Spawn 
extension (dostuff, 7001, 1) exited non-zero on 
'Parked/Zap/3-1'    -- Stopped music on 
hold on Parked/Zap/3-1    -- Playing 
'digits/0' (language 'en')    -- Playing 'digits/1' 
(language 'en')    -- Parking call to 
'Zap/1-2'    -- Hungup 'Zap/1-2'    -- 
Stopped music on hold on Zap/3-1  == Zap/3-1 got tired of being 
parked    -- Hungup 'Zap/3-1'
 
I'm not sure what I'm missing.  Apparently 
something to do with parked calls, so I must be misunderstanding how do to 
the call transfer.
 
I've also tried enabling Asterisk transfer on 
the channel (exten => 7010,1,Dial(${CORDLESS},20,tT)).
My understanding of this method is that this 
allows one to hit the pound (#) to start a transfer.  Yet pound does 
nothing.  Is it fair to assume that the tT only works on SIP channels, 
or am I missing something else.
 
Any help is much appreciated
 
Paul




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Re: [Asterisk-Users] Wait(n) -v- Background(silence/n) ?

2005-01-17 Thread Howard Lowndes

On Tue, 2005-01-18 at 13:18, Eric Wieling wrote:
> Howard Lowndes wrote:
> 
> > Will Wait(n) still listen for DTMF input from the caller after there has
> > been a Background(some-message) prompt, or do I need to use
> > Background(silence/n) to still listen for DTMF?
> > 
> 
> The WaitExten and Read applications won't work for you?

Duh!

Ta!

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-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
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Get rid of the Australian states."


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Re: [Asterisk-Users] internal dial tone on password from outside

When I experimented with DISA, I found it to be very unreliable - sometimes 
it would ignore my key presses and just keep giving dialtone, sometimes it 
would work.  I couldn't find a rhyme or reason to it.   I ended up just 
giving up and going with the silence

Paul
- Original Message - 
From: "Brian Dingman" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Monday, January 17, 2005 7:43 PM
Subject: Re: [Asterisk-Users] internal dial tone on password from outside


http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA
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Re: [Asterisk-Users] On Hold music

On Mon, 2005-01-17 at 20:53 -0500, Computer Onsite Support wrote:
> Can anyone of you help me out with this issue. My Asterisk is working
> fine except my music-on-hold will NOT work even though I just retry
> three different other machines with different board and sound.

You don't need any sound card for musing on hold to work.
Loot at this webpage it will guide you:
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf

uncomment in musiconhold.conf one of the line:
[classes]
;default => quietmp3:/var/lib/asterisk/mohmp3
loud => mp3:/var/lib/asterisk/mohmp3
;random => quietmp3:/var/lib/asterisk/mohmp3,-z

specify in you extension.conf what you want to play, example:
[office-open]
exten => s,1,Wait(2)
exten => s,2,Answer()
exten => s,3,SetMusicOnHold(loud)

-- 
#Joseph
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Re: [Asterisk-Users] callers who don't press any keys

2005-01-17 Thread John Millican

> Warren Burstein wrote:
> > I've noticed that some callers listen to our main menu and don't
> > press any keys.  

>
 Remember Rotary Phones?  They are still in use in some homes/areas
John M
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Re: [Asterisk-Users] internal dial tone on password from outside

2005-01-17 Thread Brian Dingman

http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA
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RE: [Asterisk-Users] On Hold music

I'm confuse because I was able to install on a shit machine and everything
works fine even though It was not necessary installing such "timer" what you
just mentioned. NOW that I just installed it on a better machine "Celeron
850Mhz 394M of Ram instead of PIII 500Mhz 128M and this thing will no play
music on hold. Is there anything I've forgot during installation and
configuration. I'm running Red hat 8.0 as recommended with X100P Card.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mike Sander
Sent: Monday, January 17, 2005 9:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] On Hold music


Do you have Zaptel cards installed? You need to have a timer installed
(whatever that means). If you dont have a zaptel card, then use ztdummy to
fake one.

You need to download and compile the zaptel drivers (from asterisk website).
Edit the makefile and find the line:

TZOBJS=zonedata.lo tonezone.lo
LIBTONEZONE=libtonezone.so.1.0
MODULES=zaptel tor2 torisa wcusb wcfxo wcfxs \
ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp # ztdummy
#MODULES+=wcfxsusb

Then remove the # before ztdummy

Type Make All
Type Make Install

Add a line to load the module ztdummy on boot using the /etc/rc.d files
The command is modprobe ztdummy

More information at:
http://www.voip-info.org/tiki-print.php?page=Asterisk+timer+ztdummy

Mike Sander
Operations Manager

Suite 4 / 38-48 Waterloo St
Surry Hills N.S.W 2010
Phone:(02) 8307 8877
Fax:(02)93182254
Mobile:0401 010 289
Email: [EMAIL PROTECTED]
Website: www.corporatebankinginternational.com

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Computer
Onsite Support
Sent: Tuesday, 18 January 2005 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] On Hold music

Can anyone of you help me out with this issue. My Asterisk is working fine
except my music-on-hold will NOT work even though I just retry three
different other machines with different board and sound.

[Manny Teixeira] 
 al Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Manjit Riat
Sent: Monday, January 17, 2005 8:42 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP URL for incoming
I want to set up my asterisk to receive SIP calls using the URL
[EMAIL PROTECTED] . I have my own DNS server but would like know what entry
goes into it as I have never set up SRV records before. (if it matter it is
a BIND dns server).

thanx

--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005


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[Asterisk-Users] internal dial tone on password from outside

Is it possible to get an internal dial tone when I call to my asterisk
and enter password?

I would like to call my line enter extension - password - and get
internal dial tone.
once I'm in I would like to dial based on what context permits, mostly
long distance calls VOIP.

I can not preset the extension to certain number as I don't know what
number I will be dialing.

-- 
#Joseph
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Re: [Asterisk-Users] transfers with zap channel

2005-01-17 Thread Lyle Giese




How long between getting parked is the orginal call 
dropping?  
 
Depending on your dialplan, yes dialing 700x will 
almost immediately send the call to call parking. (IMHO, poor extension planning 
can also cause this.)
 
I don't use the t or T options.  
IMHO, you just lose the ability to use the # key and confused the heck out of my 
users.  Took it out and use the flash method only in my dial plan.  
Dial 700, park the call.  Dial the other extension, tell them to pick up 
701.  Or use meetme for conference calling?
 
I know I need to play with three way calling here 
also.
 
Lyle
 

  - Original Message - 
  From: 
  Paul 
  Fielding 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Monday, January 17, 2005 6:12 
  PM
  Subject: [Asterisk-Users] transfers with 
  zap channel
  
  Ok, I've seen discussion before on doing 
  transfers (attended and unattended), there seems to be much confusion over 
  it.
   
  As things sit, I've been trying (unsuccessfully) 
  to do transfers with a zap channel (analog phone attached to TDM400).  I 
  have no idea what I'm missing.  My current understanding is that I need 
  to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd number, 
  flash hook again and we're linked (attended).   Then if I hang up 
  the call will be transfered.
   
  However, when I try to do this things don't 
  work.   Here's what I do:
   
  - connection is made between Zap/3 (analog phone) 
  and Zap/1 (outside line).
  - flash hook to get dialtone (I do get 
  dialtone)
  - attempt to transfer to extension 7007 - I dial 
  7007
  - after dialing the 2nd zero, and before dialing 
  the 2nd seven, I hear Asterisk announce (seven - zero - one) and then Zap/3 is 
  hung up (I get a busy signal).  Zap/1 gets parked.
   
  Here's what the log shows:
   
      -- Zap/1-1 answered 
  Zap/3-1    -- Attempting native bridge of Zap/3-1 and 
  Zap/1-1    -- Started three way call on channel 
  1    -- Started music on hold, class 'default', on 
  Zap/3-1    -- Attempting native bridge of Zap/3-1 and 
  Zap/1-1    -- Starting simple switch on 
  'Zap/1-2'    -- Started music on hold, class 'default', on 
  Zap/3-1  == Parked Zap/3-1 on 701. Will timeout back to 
  dostuff,7001,1 in 45 seconds    -- Added extension '701' 
  priority 1 to parkedcalls    -- Playing 'digits/7' 
  (language 'en')    -- Hungup 'Zap/1-1'  == Spawn 
  extension (dostuff, 7001, 1) exited non-zero on 
  'Parked/Zap/3-1'    -- Stopped music on hold 
  on Parked/Zap/3-1    -- Playing 'digits/0' 
  (language 'en')    -- Playing 'digits/1' (language 
  'en')    -- Parking call to 'Zap/1-2'    
  -- Hungup 'Zap/1-2'    -- Stopped music on hold on 
  Zap/3-1  == Zap/3-1 got tired of being parked    
  -- Hungup 'Zap/3-1'
   
  I'm not sure what I'm missing.  Apparently 
  something to do with parked calls, so I must be misunderstanding how do to the 
  call transfer.
   
  I've also tried enabling Asterisk transfer on the 
  channel (exten => 7010,1,Dial(${CORDLESS},20,tT)).
  My understanding of this method is that this 
  allows one to hit the pound (#) to start a transfer.  Yet pound does 
  nothing.  Is it fair to assume that the tT only works on SIP channels, or 
  am I missing something else.
   
  Any help is much appreciated
   
  Paul
  
  
  

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Re: [Asterisk-Users] Wait(n) -v- Background(silence/n) ?

2005-01-17 Thread Eric Wieling

Howard Lowndes wrote:
Will Wait(n) still listen for DTMF input from the caller after there has
been a Background(some-message) prompt, or do I need to use
Background(silence/n) to still listen for DTMF?
The WaitExten and Read applications won't work for you?
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Re: [Asterisk-Users] iaxtel - -- Format for call is ADPCM

2005-01-17 Thread Eric Wieling

There was a bug with codecs for a very long time with Asterisk.  In 
[general] remove the bandwidth= line (all it does is allow specific 
codecs) and disallow=all and allow= for eac codec you want.

Joseph wrote:
When I try to call iaxtel it goes to codec ADPCM even though I have
define in iax.conf gsm
Call accepted by 69.73.19.178 (format ADPCM)
-- Format for call is ADPCM
My settings:
[general]
port=4569
register => :[EMAIL PROTECTED]
bandwidth=high
jitterbuffer=no
tos=lowdelay
[voipjet]
type=peer
host= xxx.xxx.xxx.xx
secret= xxx
auth=md5
notransfer=yes
context=incoming
disallow=all ; Prevent all codecs...
allow = ulaw ; ...except G.711 ulaw
[iaxtel]
type=friend
host=iaxtel.com
secret=
auth=rsa
context=incoming
inkeys=iaxtel
disallow=all
allow=gsm
Why is it switching me to Codec: ADPCM?
PS. It seems to me iaxtel has a problem with connection today, can
anybody confirm it?
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RE: [Asterisk-Users] On Hold music

2005-01-17 Thread Mike Sander

Do you have Zaptel cards installed? You need to have a timer installed
(whatever that means). If you dont have a zaptel card, then use ztdummy to
fake one.

You need to download and compile the zaptel drivers (from asterisk website).
Edit the makefile and find the line:

TZOBJS=zonedata.lo tonezone.lo
LIBTONEZONE=libtonezone.so.1.0
MODULES=zaptel tor2 torisa wcusb wcfxo wcfxs \
ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp # ztdummy
#MODULES+=wcfxsusb

Then remove the # before ztdummy

Type Make All
Type Make Install

Add a line to load the module ztdummy on boot using the /etc/rc.d files
The command is modprobe ztdummy

More information at:
http://www.voip-info.org/tiki-print.php?page=Asterisk+timer+ztdummy

Mike Sander
Operations Manager

Suite 4 / 38-48 Waterloo St
Surry Hills N.S.W 2010
Phone:(02) 8307 8877
Fax:(02)93182254
Mobile:0401 010 289
Email: [EMAIL PROTECTED]
Website: www.corporatebankinginternational.com

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Computer
Onsite Support
Sent: Tuesday, 18 January 2005 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] On Hold music

Can anyone of you help me out with this issue. My Asterisk is working fine
except my music-on-hold will NOT work even though I just retry three
different other machines with different board and sound.

[Manny Teixeira] 
 al Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Manjit Riat
Sent: Monday, January 17, 2005 8:42 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP URL for incoming
I want to set up my asterisk to receive SIP calls using the URL
[EMAIL PROTECTED] . I have my own DNS server but would like know what entry
goes into it as I have never set up SRV records before. (if it matter it is
a BIND dns server).

thanx

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005
 

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[Asterisk-Users] On Hold music




Can 
anyone of you help me out with this issue. My Asterisk is working fine except my 
music-on-hold will NOT work even though I just retry three different other 
machines with different board and sound.
[Manny Teixeira] 
 al 
Message-From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]On Behalf Of Manjit 
RiatSent: Monday, January 17, 2005 8:42 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP URL for 
incoming

  
  I want to set up my asterisk to 
  receive SIP calls using the URL [EMAIL PROTECTED] . I have my own DNS server 
  but would like know what entry goes into it as I have never set up SRV records 
  before. (if it matter it is a BIND dns server).
   
  thanx
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[Asterisk-Users] TDM13B - FXO ports not seeing incoming calls

2005-01-17 Thread Adam Goryachev

I seem to have tripped over a problem that just seems plain weird to
me...

I have the TDM card with 1 FXS and 3 FXO interfaces. 
The FXS connects to a modem and EFTPOS terminal, which is working fine.
The 3 FXO ports are connecting to another PBX as internal extensions.

Usually, everything works fine, however, yesterday incoming calls
stopped working. When an extension on the PBX calls one of the three
extensions, it just rings out, with absolutely no output on the asterisk
CLI.

However, when an internal asterisk extension makes an outbound call to
one of the other PBX extensions, it uses the FXO port, and works
perfectly.
-- Executing Dial("SIP/polycom_l1-c386", "Zap/g10/5586") in new
stack
-- Called g10/5586
-- Zap/126-1 answered SIP/polycom_l1-c386
-- Hungup 'Zap/126-1'

zapata.conf
---
; Analog incoming channel
context => remote
immediate => yes
usecallerid=no
callerid=no
group => 10
signalling => fxs_ks
channel => 126
channel => 127
channel => 128

zaptel.conf
---
fxoks=125
fxsks=126
fxsks=127
fxsks=128

Tonight I'll probably just stop asterisk, unload/reload the modules, and
restart asterisk. Usually this is enough to fix things...

If anyone can suggest some debugging to help find the problem, I'd
appreciate it.

Regards,
Adam
-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 8304 [EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

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[Asterisk-Users] SIP URL for incoming

2005-01-17 Thread Manjit Riat









I want to set up my asterisk to receive SIP calls using the
URL [EMAIL PROTECTED] . I have my own DNS
server but would like know what entry goes into it as I have never set up SRV records
before. (if it matter it is a BIND dns
server).

 

thanx






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Re: [Asterisk-Users] callers who don't press any keys

2005-01-17 Thread el Flynn

Warren Burstein wrote:
I've noticed that some callers listen to our main menu and don't press 
any keys.  I have it set up to restart the menu a few times and 
eventually hang up.  I'm wondering if these are wrong numbers (in that 
case, why don't they hang up) or they really want to speak to someone 
here but don't understand the menu (what's so hard about "for the 
operator, press zero"?).  They couldn't still have rotary phones (or 
phones set to pulse dial), could they?

I've been thinking of changing the menu so that if they don't press any 
keys, the eventually get the operator.

Does anyone have any experience with this?
For me most of the times its due to the TDM400P not being able to recognize the 
caller had hung up (although sometimes it does, which is puzzling).

What I normally do is have a global "attempt" counter, and increment it 
everytime the caller ends up in the "t" extension. Once it reaches a specified 
maxattempt value, decided via a GotoIf statement in the "t" extension, you just 
hang up the caller.

Flynn
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RE: [Asterisk-Users] pattern matching problem

> > How do I solve the problem with between patterns:
> > _1800
> > _1NXX
> > 
> > I would like all numbers 1800, 1877 etc to go through iaxtel 
> > but all other numbers 1xxx via voipjet
> > 
> In your default context (i.e. the one specified in sip.conf/iax.conf) 
> include the iaxtel context before the outgoing-voipjet context.  The 
> system should stop at the first match.
> 
> Good luck,
> 
> Robert Jackson

Thank you!
I was under impression that the order in extension.conf is important but
actually it is iax/sip.conf file.

-- 
#Joseph
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[Asterisk-Users] here's my IAX callthrough app and some questions about problems I have.

2005-01-17 Thread Jess Coburn

Hello all,

What my app does is accepts a call in on a Dial-In Number (DID) via
IAX, and then prompts the caller for the top secret password (123) and
then authenticates the user and prompts them to dial in the number
they'd like to call. Once they press pound after dialing in the number
it will read it back to them, if they press pound it will attempt to
connect via the second IAX provider, if they press star it will allow
them enter in the number over again.

Now here's the problems and questions:

1. DTMF detection seems flawed, sometimes it's dead on but alot of
times it will see a single keypress as multiple keypresses. So I may
press 561 but it will see 51 and all three keypresses are about
the same length. Is this unique to my case or do you others see this
too. I suspect it's due to either background noise or maybe
packetloss? Any ideas on how to clean this up?

2. The only way I can get the app to fire off is if I put the
extension mapping in as _NXXNXX,1,CMD I'd like to use s,1,CMD but
I don't know what I'm missing here or doing wrong.

Below are a copies of my extensions.conf file and my iax.conf file. 

Regards,
Jess

extensions.conf
file-

[general]
static=yes
writeprotect=no

[globals]
${OUTGOING-NUM}=


[arbitrary-in]  ; <-- Should match the context you have 
   ; under [incoming] in iax.conf
exten => _NXXNXX,1,Answer
exten => _NXXNXX,2,Background(vm-password)
exten => _NXXNXX,3,Authenticate(123)
exten => _NXXNXX,4,Playback(beep)
exten => _NXXNXX,5,SetVar(NR=)
exten => _NXXNXX,6,Goto(testdtmf|s|1)

; 
; This context is used by the sample [arbitrary-name]
; context above to read back each digit you press.
; 
[testdtmf]
exten => s,1,SetVar(NR=)
exten => s,2,Background(pls-entr-num-uwish2-call)
exten => s,3,Background(and-prs-pound-whn-finished)
exten => s,4,Background(beep)
exten => s,5,WaitExten(10)
exten => _x,1,SetVar(NR=${NR}${EXTEN})
exten => _x,2,NoOp(${NR})
exten => _x,3,Goto(testdtmf|s|5)
exten => _#,1,Goto(verifynumber|s|1)
exten => i,1,Goto(testdtmf|s|1)
exten => t,1,Hangup

[verifynumber]
exten => s,1,Background(you-dialed)
exten => s,2,SayDigits(${NR})
exten => s,3,Background(if-correct-press)
exten => s,4,Background(pound)
exten => s,5,Background(otherwise-press)
exten => s,6,Background(star)
exten => _#,1,Background(pls-wait-connect-call)
exten => _#,2,Dial(IAX2/[EMAIL PROTECTED]/${NR},30)
exten => _#,3,Background(something-terribly-wrong);
exten => _#,4,Background(goodbye)
exten => _#,5,Hangup 
exten => _*,1,Goto(testdtmf|s|1)


iax.conf file 
--
; iax.conf   

[general]

${INCOMING-USR}=SECRET-USERNAME
${INCOMING-PWD}=SECRET-PWD
${LIVEVOIP-SVR}=217.160.244.186

bandwidth=high
disallow=lpc10  
jitterbuffer=yes
dropcount=2
maxjitterbuffer=500
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1

register => ${INCOMING-USR}:[EMAIL PROTECTED]
tos=lowdelay

[incoming]
; this is the incoming IAX provider
type=user
secret=ITS-SECRET
deny=0.0.0.0/0.0.0.0
permit=217.160.244.186/255.255.255.0
context=arbitrary-in


[outgoing] 
;this is the outgoing IAX provider
type=peer 
host= 216.118.117.46
secret= ITS-SECRET
auth=md5 
notransfer=yes 
context=default
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Re: [Asterisk-Users] callers who don't press any keys

2005-01-17 Thread Adam Goryachev

On Mon, 2005-01-17 at 19:32 -0500, Warren Burstein wrote:
> I've noticed that some callers listen to our main menu and don't press 
> any keys.  I have it set up to restart the menu a few times and 
> eventually hang up.  I'm wondering if these are wrong numbers (in that 
> case, why don't they hang up) or they really want to speak to someone 
> here but don't understand the menu (what's so hard about "for the 
> operator, press zero"?).  They couldn't still have rotary phones (or 
> phones set to pulse dial), could they?
> 
> I've been thinking of changing the menu so that if they don't press any 
> keys, the eventually get the operator.
> 
> Does anyone have any experience with this?

Not specifically, though typically, the usual handling of this is to
transfer to an operator if they do nothing You may want to modify
your current hangup behaviour.

You may not have disconnect notification, or detection working, and so
the end user may have hung up, but your end doesn't realise, so it
cycles through the menu before hanging up (if you change to transfer to
an operator, then the operator should tell you they are getting lots of
calls with "some tone??"

Some people don't actually bother listening to all the crap, and just
wait for someone to answer... The number of people that ignore the
announcement of our company name, etc, and still think they are talking
to a different company... it surprises me

Personally, it depends on how important the person contacting you is. If
it is a home phone, then don't worry, it is probably a tele-marketer or
something. If it is a problem, your mum/girlfriend/wife will tell you
sooner or later that you kept hanging up on her...
If it is a business, I'd strongly suggest changing to a default of 'send
to an operator' and let a human decide whether to hangup or not.

Just my 0.02c

Regards,
Adam
-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 8304 [EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

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[Asterisk-Users] Transferring calls on Asterisk with X-Lite

2005-01-17 Thread Mike Sander









I am having trouble transferring calls using asterisk. I
think it is my * installation, because this worked fine with the same system
when it was hosted at our VoIP providers.

 

I receive a call on my IAX Trunk, to my extensions. 

I speak to the incoming call and tell them I’ll just
transfer the call.

I click “Transfer”, dial extension and click “Transfer”
again.

Normally the call will disappear on my system and start
ringing on the new extension.

In this case, the call just hangs up.

 

 

Any Ideas? Do you have to setup transfers in the extensions
at all?

 

Does this have something to do with the “Reinvite”
status of the SIP phones?

 

With Thanks

 

Mike Sander
Operations Manager


Suite 4 / 38-48 Waterloo St
Surry Hills N.S.W 2010
Phone:(02) 8307 8877
Fax:(02)93182254
Mobile:0401 010
289
Email: [EMAIL PROTECTED]
Website: www.corporatebankinginternational.com

 








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[Asterisk-Users] Re: Media Path Optimization & NAT

2005-01-17 Thread Adam Sherman

Rich Adamson wrote:
Now, I would very much like to remove the "canreinvite=no" from the 
provider's definition on sip.conf, but doing so causes Asterisk to send 
a re-invite to the provider pointing to a private IP. I thought that 
correct localnet entries would solve this...

By changing to canreinvite=yes, are you expecting the asterisk box to
act as a router, passing rtp traffic from your sip provider through
the box to a sip phone with a private address (without passing
asterisk code in the middle of the rtp session)?
No, sorry. I'm looking for Asterisk to not issue the re-invites if the 
two devices can't see each other. Think of mobile users who are often 
behind the corporate firewall but also travel. I'm trying to avoid 
having the media path be "user->corporate lan->pstn provider". I want it 
to be "user->pstn provider".

If not to accomplish this, what are the localnet configuration entries for?

When a user is in the office, his phone registers with asterisk, and he
places calls through asterisk to the sip provider. But, when he's
out of the office, he takes his phone with him, and you are wanting
him to make use of the canreinvite=yes to allow his phone to connect
directly to the sip provider avoiding asterisk (from an rtp perspective). 
Is that right?
Yes. A softphone on a laptop makes for a more believable example though.
So, is this possible? And, if not, what do the "localnet" entries provide?
Thanks,
A.
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Re: [Asterisk-Users] callers who don't press any keys

Warren Burstein wrote:
I've noticed that some callers listen to our main menu and don't press 
any keys.  I have it set up to restart the menu a few times and 
eventually hang up.  I'm wondering if these are wrong numbers (in that 
case, why don't they hang up) or they really want to speak to someone 
here but don't understand the menu (what's so hard about "for the 
operator, press zero"?).  They couldn't still have rotary phones (or 
phones set to pulse dial), could they?

I've been thinking of changing the menu so that if they don't press any 
keys, the eventually get the operator.

Does anyone have any experience with this?
Yeah.  I get the same thing.  Not very often, but concerning none the less.
I guess the best bet would be to send all of these people to a 
particular extension.  I.E. if you have zap extensions, you could do a 
dial with r3 or r4 to change the ringing tone.  That way you would know 
that they came to you because they didn't enter a number, and you could 
interrogate them as to what/how/why etc...

--
Cheers,
Matt Riddell
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[Asterisk-Users] spandsp and app_txfax

2005-01-17 Thread Nir Simionovich









Hi
all,

 

  Ok,
I've been bashing my head for a few hours now on this, trying to figure out if
I've
done something wrong, but everything seems to me hunky-dory. So here's the
deal:

 

1.
I've compiled the spandsp 0.0.2pre10 source code successfully and also the
asterisk
    application associated with it.

2.
Receiving a fax at asterisk works fine, at least appears to be working fine.

3.
Sending an outgoing fax from asterisk to the world doesn't work. The sending
out and
    receiving is performed via an E1 interface, not VoIP.

 

  Has
anyone got experience with this package? I really don't want to go back to
hylafax
which is a serious overkill.

 

Nir S






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Re: [Asterisk-Users] Offtopic: improving softphone latency on Linux?

2005-01-17 Thread Bruno Hertz

On Tue, 2005-01-18 at 07:43 +0800, Steve Underwood wrote:

> Latencies that big should not be due to the softphone. They are often 
> due to the sound card driver.

Yeah, it's what I thought, but then, as said, I tried the planetccrma
kernel and drivers, which are supposed to support professional audio
applications. Not much difference, unfortunately. I even tuned pci
latencies to no avail.

My card btw is a soundblaster with ensoniq chip, so any obvious driver
anomaly presumably would soon be filed as a bug, Fedora Core or
otherwise. But, with alsa oss emulation and stuff, it really might be
that latencies just add up, which would after all mean that Linux as a
desktop system still has it's drawbacks.

Anyway, in case you use softphones on Linux, and did compare their
performance with Windows alternatives finding that they can compete, may
I ask what card/driver/kernel version do work for you?

Thanks, Bruno.

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[Asterisk-Users] callers who don't press any keys

2005-01-17 Thread Warren Burstein

I've noticed that some callers listen to our main menu and don't press 
any keys.  I have it set up to restart the menu a few times and 
eventually hang up.  I'm wondering if these are wrong numbers (in that 
case, why don't they hang up) or they really want to speak to someone 
here but don't understand the menu (what's so hard about "for the 
operator, press zero"?).  They couldn't still have rotary phones (or 
phones set to pulse dial), could they?

I've been thinking of changing the menu so that if they don't press any 
keys, the eventually get the operator.

Does anyone have any experience with this?
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Re: [Asterisk-Users] Wait(n) -v- Background(silence/n) ?

2005-01-17 Thread Trevor Peirce

Howard Lowndes wrote:
Will Wait(n) still listen for DTMF input from the caller after there has
been a Background(some-message) prompt, or do I need to use
Background(silence/n) to still listen for DTMF?
 

WaitExten(n) will
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[Asterisk-Users] transfers with zap channel




Ok, I've seen discussion before on doing transfers 
(attended and unattended), there seems to be much confusion over 
it.
 
As things sit, I've been trying (unsuccessfully) to 
do transfers with a zap channel (analog phone attached to TDM400).  I have 
no idea what I'm missing.  My current understanding is that I need to have 
transfer=yes in zapata.conf, do a flash hook, dial the 2nd number, flash hook 
again and we're linked (attended).   Then if I hang up the call will 
be transfered.
 
However, when I try to do this things don't 
work.   Here's what I do:
 
- connection is made between Zap/3 (analog phone) 
and Zap/1 (outside line).
- flash hook to get dialtone (I do get 
dialtone)
- attempt to transfer to extension 7007 - I dial 
7007
- after dialing the 2nd zero, and before dialing 
the 2nd seven, I hear Asterisk announce (seven - zero - one) and then Zap/3 is 
hung up (I get a busy signal).  Zap/1 gets parked.
 
Here's what the log shows:
 
    -- Zap/1-1 answered 
Zap/3-1    -- Attempting native bridge of Zap/3-1 and 
Zap/1-1    -- Started three way call on channel 
1    -- Started music on hold, class 'default', on 
Zap/3-1    -- Attempting native bridge of Zap/3-1 and 
Zap/1-1    -- Starting simple switch on 
'Zap/1-2'    -- Started music on hold, class 'default', on 
Zap/3-1  == Parked Zap/3-1 on 701. Will timeout back to dostuff,7001,1 
in 45 seconds    -- Added extension '701' priority 1 to 
parkedcalls    -- Playing 'digits/7' (language 
'en')    -- Hungup 'Zap/1-1'  == Spawn extension 
(dostuff, 7001, 1) exited non-zero on 
'Parked/Zap/3-1'    -- Stopped music on hold on 
Parked/Zap/3-1    -- Playing 'digits/0' 
(language 'en')    -- Playing 'digits/1' (language 
'en')    -- Parking call to 'Zap/1-2'    
-- Hungup 'Zap/1-2'    -- Stopped music on hold on 
Zap/3-1  == Zap/3-1 got tired of being parked    -- 
Hungup 'Zap/3-1'
 
I'm not sure what I'm missing.  Apparently 
something to do with parked calls, so I must be misunderstanding how do to the 
call transfer.
 
I've also tried enabling Asterisk transfer on the 
channel (exten => 7010,1,Dial(${CORDLESS},20,tT)).
My understanding of this method is that this allows 
one to hit the pound (#) to start a transfer.  Yet pound does 
nothing.  Is it fair to assume that the tT only works on SIP channels, or 
am I missing something else.
 
Any help is much appreciated
 
Paul

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RE: [Asterisk-Users] Re: Any interest in a Canadian Asterisk

[EMAIL PROTECTED] wrote:
> Quoting [EMAIL PROTECTED]:
> 
>>> Would it be considered trolling to start a thread on Cleaning Maple
>>> Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi?
>> 
>> Let's not forget the weekly "tooques and telephony" segment, and a
>> review of the best block heaters for your wi-fi fones.
>> 
> 
> Oh, we're gonna have a good time next Thursday.
> 
> We need to get Molson Canadian to sponsor us and find Bob &
> Doug for the
> event?

Who needs those hosers, eh? 

> By the way, eh. It's hard to get the moose to cooperate. When you put
> the parabolic antenna on his antlers you have to ride him backwards
> when you're leaving your cabin, eh.

That's nuthin'! Try gettin him on the snowmobile, eh? Or makin' him sit
still in the drive-through at Tim Hortons.

I'm tellin' ya . . . 


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[Asterisk-Users] RE: Canadian Content: Telus and Shaw...

Kim Lux wrote:
> I called Telus before Christmas requesting some sort of VOIP
> connection. Here is what I learned:
> 
> a) the guy I was talking to never heard of *

That'll change.

> b) they didn't think there was any way that a PC could
> perform the duties of a PBX

He was probably thinking of the Nortel BCM. It's a PC pretending to be a
PBX, and yeah, it ain't up to the task.

> c) they told me they didn't have any VOIP connections, but
> then told me that they would supply and connect Nortel PBXs using H323

Yep. The name of the game is customer lock-in.

> d) they would not supply me with an H323 connection for *.

More to the point, they don't have the processes in place to make it
happen.

> I don't have time to discuss this in detail, I just thought
> I'd share it based on the chat in the CDN list discussion.
> 
> We are going with babytel.  I'll advise how that works when
> it is up and running, hopefully next week.

I don't see an option to connect Asterisk to them.

> BTW: Shaw is supposed to start supplying VOIP on a separate
> network from their high speed network.  Here is the news clip:
> 
>
http://www.canoe.ca/NewsStand/CalgarySun/News/2005/01/14/898082-sun.html

> I find this interesting because several people have told me they are
> using Shaw's high speed Internet service as the backbone of their VOIP
> system. (Extreme is supposed to work even better.)  

> I wonder if Telus is going to block the SIP ports on their ADSL
network

I'm wondering what the CRTC is planning with respect to VoIP.

Whatever they do, they'll probably miss IAX entirely, so no worries.

> ?  I wonder if Shaw will ?  (Telus presently blocks the SMPT port so
> that you MUST you their mail server.)  

> I wonder if shaw or telus people lurk on this site.

If you mean people that work for those companies, then sure (but for the
most part they love it the same as any of us). If you mean
decision-makers, I would say generally no. Asterisk is not yet on
anyone's radar scope, so it won't be considered a threat. That'll happen
when it starts stealing market-share.

Jim.

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Re: [Asterisk-Users] iaxtel - -- Format for call is ADPCM

2005-01-17 Thread Rich Adamson

> 
> Why is it switching me to Codec: ADPCM?
> 
> PS. It seems to me iaxtel has a problem with connection today, can
> anybody confirm it?

I just tried to place a call via iaxtel and watched the packets with
ethereal. The iaxtel server is very very slow to respond to _any_
packet, indicating its not feeling very well. Could not complete
the call at all, and 'iax2 show registry' indicates instability as
well.



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Re: [Asterisk-Users] Passing PIN Numbers

2005-01-17 Thread Rene Kluwen

Title: Passing PIN Numbers

This is a long shot, I am not sure if it will solve
your problem:
Did you try to change dtmfmode in
sip.conf?
Rene Kluwen
Chimit

- Original Message -
From:
Michael Di Martino
To: asterisk-users@lists.digium.com

Sent: Friday, January 14, 2005 3:53
PM
Subject: [Asterisk-Users] Passing PIN
Numbers

To All If anyone can shed any light on this
it would be greatly appreciated. My phones are unable to enter pins numbers correctly when
required by the party they are calling.
For example I was given an
outside number to attend conference bridge. After the call was connected it
required me to enter a 4 digit PIN. Now here is the problem whenever I enter a
pin it is received twice. For example if the PIN is 1234 they receive it as
12341234.
Any ideas what could be
wrong?
BTW we are using SNOM 190 ip phones
(sip)
Regards, Michael DiMartino Director of MIS The telx Group, Inc. 17 State St, 33rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603

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Re: [Asterisk-Users] Voice Mail Notification

2005-01-17 Thread Rene Kluwen




Alternatively,
What I (we) do personally: In stead of having * 
call my cellphone, it sends an MMS message with the message audio as 
content.
 
Rene Kluwen
Chimit

  - Original Message - 
  From: 
  Mike 
  Boger Jr 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Friday, January 14, 2005 4:14 
  AM
  Subject: [Asterisk-Users] Voice Mail 
  Notification
  
  Hi,
   
  Here's the deal. When someone leaves me a 
  voicemail message I want Asterisk to call me on my cellphone by dialing my 
  cellphone number and tell me I have a message. Is this possible? Can anyone 
  cite examples? Most commercial voicemail systems produced in the last 10 years 
  can do this. Any help would be much appreciated.
   
  Regards,
   
  Mike
  
  

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Re: [Asterisk-Users] SMS Gateway

2005-01-17 Thread Rene Kluwen

There's lots.
www.clickatell.com is one of them.
Google for "sms gateway" and you will find a bunch - especially in the
paid-add section.

Rene Kluwen
Chimit

- Original Message -
From: "Brian C. Fertig" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, January 13, 2005 7:01 PM
Subject: [Asterisk-Users] SMS Gateway


Does anyone know of any companies where I can interconnect with for SMS?



.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office

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Re: [Asterisk-Users] China direct route

2005-01-17 Thread Steve Underwood

Direct VoIP in and out of China is illegal there, and is generally 
blocked by the great fire wall of China. Try an IAX connection. I don't 
think they have blocked those yet. :-)

Steve
mohammad wrote:
HI;
 
 
We need China direct route over H323.
plz contcat me offline
 
MSN: [EMAIL PROTECTED] 
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Re: [Asterisk-Users] Wait(n) -v- Background(silence/n) ?

In my AGI script I made the next trick:
   $digit = $AGI->get_data("vm-enter-num-to-call-then-pound", 15000, 1);
   while ( $digit eq 0 or $digit )
   {
   $phoneNum .= $digit;
   $digit = $AGI->get_data("empty", 7000, 1);
   }
where file empty.gsm have 0 byte length.
It works like a charm for me.
All the Best!
Sergey.
Howard Lowndes wrote:
Will Wait(n) still listen for DTMF input from the caller after there has
been a Background(some-message) prompt, or do I need to use
Background(silence/n) to still listen for DTMF?
 

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Re: [Asterisk-Users] Offtopic: improving softphone latency on Linux?

2005-01-17 Thread Steve Underwood

Hi Bruno,
Latencies that big should not be due to the softphone. They are often 
due to the sound card driver. The driver for the Yamaha sound chip in my 
Vaio, for example, has massive latency like you are seeing. There seems 
to be no way to configure that driver to stop it. With the good drivers 
you can configure the sound card's buffering - small buffers/low 
latency, or big buffers/high latency. The Yamaha driver accepts those 
commands, but it seems to have no effect on actual operation. You may 
find the latency is dramatically different if you try a different sound 
card or a different machine.

Regards,
Steve
Bruno Hertz wrote:
Hi folks
last weekend, I tried Windows Messenger first time and was stunned by
the little latency it gives. Until now, I've been using softphones on
Linux exclusively, like iaxcomm, linphone and sjphone, and they all give
me about 1, at times even 2 secs delay. Whereas Messenger really seems
to be in the millisec range.
Of course, I'm now curious why there is that difference. Clearly,
Messenger is more tightly integrated with the OS and accordingly tuned.
So where does this time go? Kernel? Application level? Web searches seem
to suggest that sound latency generally is a problem on Linux, so I
tried the low latency kernel from
http://ccrma.stanford.edu/planetccrma/software/planetccrma.html
(there are two kernels, actually, where I only got the stable version to
boot - bleeding edge didn't do on my machine).
Still, that kernel did not really improve things in a noticeable way.
Question hence: did some of you guys experience and investigate this
same issue? Any recommendations or hints how to make VoIP even more
enjoyable on linux?
I wouldn't care that much if I was the only affected party, but of
course whomever I call will also suffer from those delays, so as the
staunch Linux advocate I've been so far I'd really like to show better
performance ...
Thanks, Bruno.
 

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[Asterisk-Users] Wait(n) -v- Background(silence/n) ?

2005-01-17 Thread Howard Lowndes

Will Wait(n) still listen for DTMF input from the caller after there has
been a Background(some-message) prompt, or do I need to use
Background(silence/n) to still listen for DTMF?

-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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[Asterisk-Users] iaxtel - -- Format for call is ADPCM

When I try to call iaxtel it goes to codec ADPCM even though I have
define in iax.conf gsm

Call accepted by 69.73.19.178 (format ADPCM)
-- Format for call is ADPCM

My settings:
[general]
port=4569

register => :[EMAIL PROTECTED]
bandwidth=high
jitterbuffer=no
tos=lowdelay

[voipjet]
type=peer
host= xxx.xxx.xxx.xx
secret= xxx
auth=md5
notransfer=yes
context=incoming
disallow=all ; Prevent all codecs...
allow = ulaw ; ...except G.711 ulaw

[iaxtel]
type=friend
host=iaxtel.com
secret=
auth=rsa
context=incoming
inkeys=iaxtel
disallow=all
allow=gsm

Why is it switching me to Codec: ADPCM?

PS. It seems to me iaxtel has a problem with connection today, can
anybody confirm it?

-- 
#Joseph
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RE: [Asterisk-Users] DIDs anywhere but here?

2005-01-17 Thread Mike Sander

We have DID's in 5 Australian cities for $5 per month.

Mike Sander
Operations Manager

Suite 4 / 38-48 Waterloo St
Surry Hills N.S.W 2010
Phone:(02) 8307 8877
Fax:(02)93182254
Mobile:0401 010 289
Email: [EMAIL PROTECTED]
Website: www.corporatebankinginternational.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Tuesday, 18 January 2005 3:46 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] DIDs anywhere but here?

Are there affordable DIDs (preferably IAX) available anywhere outside
the US?  I want to use it to meet ICANN requirements for providing a
valid phone number, yet pre-empting some of the telemarketing calls my
domain registrations generate.  (Yes, I asked a similar question about
900# availability before).  I'd prefer to have a number somewhere
outside the NANP, preferably an asian country.  This number will
(obviously) be low-volume (minutes/month at the most), and shouldn't
cost more than a couple of bucks.  Maybe a list member knows and/or is
using one?

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No virus found in this incoming message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005
 

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Re: [Asterisk-Users] Canadian Content: Telus and Shaw...

2005-01-17 Thread Kim Lux


So what stops this from happening a year from now, leaving our VOIP
system high and dry for a decent broadband connection ?

Matt: do you do SIP on broadband with "Telcom"  Has their latency
addition wrecked your connection ?

Thanks.  



On Tue, 2005-01-18 at 12:03 +1300, Matt Riddell wrote:
> Kim Lux wrote:
> > That is a good point.  I never thought of doing that. 
> > 
> > They could kill SIP connections by subtly delaying any upload traffic
> > streams, ie introducing latency or jitter.  
> 
> LOL!  As our New Zealand Telecom's Provider/Internet Provider has done. 
>   A little background - in New Zealand we have one company really - 
> Telecom.  They provide the Internet and the telephone services.  A 
> couple of weeks ago they added 30ms of latency to all connections and no 
> one batted an eyelid.
> 
> While playing counterstrike on foreign servers (Australia mainly) I 
> regularly get jitter of 300-400ms (i.e. the game starts at 120ms ping 
> and at times rises above 500ms).
> 
> Here the regulatory power is called the commerce commission.  They were 
> about to release a finding that Telecom (which was originally state 
> owned) had to unbundle their lines.  However, about one or two weeks 
> before the finding was released, they changed their mind.  Much to the 
> detriment of New Zealand internet...
> 
> So, once again, the big boys get to keep their toys, while we can but 
> beg for quality service...
> 
-- 
Kim Lux,  Diesel Research Inc.


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Re: [Asterisk-Users] Canadian Content: Telus and Shaw...

Kim Lux wrote:
That is a good point.  I never thought of doing that. 

They could kill SIP connections by subtly delaying any upload traffic
streams, ie introducing latency or jitter.  
LOL!  As our New Zealand Telecom's Provider/Internet Provider has done. 
 A little background - in New Zealand we have one company really - 
Telecom.  They provide the Internet and the telephone services.  A 
couple of weeks ago they added 30ms of latency to all connections and no 
one batted an eyelid.

While playing counterstrike on foreign servers (Australia mainly) I 
regularly get jitter of 300-400ms (i.e. the game starts at 120ms ping 
and at times rises above 500ms).

Here the regulatory power is called the commerce commission.  They were 
about to release a finding that Telecom (which was originally state 
owned) had to unbundle their lines.  However, about one or two weeks 
before the finding was released, they changed their mind.  Much to the 
detriment of New Zealand internet...

So, once again, the big boys get to keep their toys, while we can but 
beg for quality service...

--
Cheers,
Matt Riddell
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[Asterisk-Users] Looking for Asterisk termination in Russia

2005-01-17 Thread Vitalie Apostu

I would like to make inlimited call to russia in exchange to USA.

Any idea are welcome.

Thanks.

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RE: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

[EMAIL PROTECTED] wrote:
> On January 17, 2005 04:47 pm, Jim Van Meggelen wrote:
>> LOL. I hadn't thought of it that way. Little vignettes amidst the
>> commercials?
> 
> Exactly -- It's precisely why I hang around on linux-elitists
> and a couple
> other oddball lists...  a good 90% of what's there is crap
> but man when
> something good comes by...  wowza.

It sure can be time consuming trying to keep up, though.

>> Just because the volume isn't there? That might be a good thing, ya
>> know - have a list with, say, one or two messages a day, on average.
> 
> True, but that's why I like looking at sineapps now and again -- they
> sometimes focus on things that I've not even seen, it's interesting
> reading...  but I slug it out on the list well, just to slug
> it out.  :-)

I think we are all a bit masochistic on this list . . . all 10,000 plus
of us . . .

>> It was a policy at our company that any new product implementation
>> would always require Technical Support be involved until several
>> engineers, technicians and installers were comfortable with it. I
>> hope I always remember the lessons learned from getting new
>> products, and having to develop training and implementation
>> practices. 
> 
> ...
> 
>> Seems that making mistakes is actually a fantastic (albeit
>> uncomfortable) way to learn. I sometimes wonder if I unconsciously
>> muck things up at first as a rite of passage.
> 
> We have a similar policy here and it really helps people
> understand why things
> are done a certain way when they have to field some of the
> customer calls
> themselves.  "right" and "wrong" take on new nuances that
> they would have
> otherwise been oblivious and even belligerent towards.
> 
>> Nobody knows a thing so well as those who can expertly break it.
> 
> That sounds very close to "As soon as you make something
> idiot-proof along
> comes a better class of idiot."  :-)

One of my favorites is (to paraphase Douglas Adams):

"It's easy to be blinded to the essential uselessness of these things by
the sense of satisfaction you get from making them work at all"

>> Would it be considered trolling to start a thread on Cleaning Maple
>> Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi?
> 
> Let's not forget the weekly "tooques and telephony" segment,
> and a review of
> the best block heaters for your wi-fi fones.

Snoms and Snowmobiles?
Ice Phishing?
Tim Hortel?




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Re: [Asterisk-Users] Offtopic: improving softphone latency on Linux?

2005-01-17 Thread Bruno Hertz

On Mon, 2005-01-17 at 16:51 -0500, Steve Kann wrote:

> What softphone are you using on Linux?
> 
> > iaxcomm, linphone and sjphone, and they all give
> 
> If you use an iaxclient-based softphone on linux as root, it runs with 
> RT priority, and pretty low latency

Hmmm, on my side I can't say it makes much of a difference for iaxcomm.
It does improve sound quality though, since running iaxcomm non root
produces pretty crackling audio, for whatever reasons. Altogether, I
find that sjphone performs best, regarding quality as well as latency,
where Windows/WM still seems to play in a different league.

Thanks, Bruno.

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Re: [Asterisk-Users] Euro ISDN and Caller ID (Sweden)

2005-01-17 Thread Peter Svensson

On Mon, 17 Jan 2005, Daniel Nyström wrote:

> Do anyone have experiences with Euro ISDN in Sweden?

Yes, it works. We have not connected with Telia though, only other 
operators.

> Does CallerID work properly? Both in and out.

Yes.

> Do anyone know of a reseller for Digium cards and/or CarrierAccess Adit
> 600 in Sweden or Europe (EU)?

Telappliant in England sells the Digium cards. There should be quite a few 
others. You can try asking Digium. I have no idea about the Adit channel 
banks.

Peter

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Re: [Asterisk-Users] Re: Media Path Optimization & NAT

2005-01-17 Thread Rich Adamson

> >>I have a bunch of setups where an Asterisk system with a public IP 
> >>doubles as a router/gateway/firewall for a set of phones on a private 
> >>network.
> >>
> >>We're using external SIP providers.
> >>
> >>Everything works quite nicely.
> >>
> >>Now, I would very much like to remove the "canreinvite=no" from the 
> >>provider's definition on sip.conf, but doing so causes Asterisk to send 
> >>a re-invite to the provider pointing to a private IP. I thought that 
> >>correct localnet entries would solve this...
> >>
> >>Is what I'm after even possible? Am I looking in the wrong place?
> > 
> > By changing to canreinvite=yes, are you expecting the asterisk box to
> > act as a router, passing rtp traffic from your sip provider through
> > the box to a sip phone with a private address (without passing
> > asterisk code in the middle of the rtp session)?
> 
> No, sorry. I'm looking for Asterisk to not issue the re-invites if the 
> two devices can't see each other. Think of mobile users who are often 
> behind the corporate firewall but also travel. I'm trying to avoid 
> having the media path be "user->corporate lan->pstn provider". I want it 
> to be "user->pstn provider".
> 
> If not to accomplish this, what are the localnet configuration entries for?

Okay, maybe I'm a little dense here; is this what you're looking for?

When a user is in the office, his phone registers with asterisk, and he
places calls through asterisk to the sip provider. But, when he's
out of the office, he takes his phone with him, and you are wanting
him to make use of the canreinvite=yes to allow his phone to connect
directly to the sip provider avoiding asterisk (from an rtp perspective). 
Is that right?

If not, then I'm not at all understanding 'exactly' what you're trying
to accomplish. Maybe a better example would help.


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Re: [Asterisk-Users] X100P Unstable.

2005-01-17 Thread alexb


Hi ,

How did you go with this problem?

We have had our * box working for over
6 months now with out a problem. then all of a sudden we now have the n
same problem that you have/had?

Any solutions?

Thanks 

Alex Broad





Jefferson Carvalho <[EMAIL PROTECTED]>

Sent by: [EMAIL PROTECTED]
29/09/2004 02:19 PM



Please respond to
Asterisk Users Mailing List - Non-Commercial Discussion 





To
Asterisk Users Mailing List
- Non-Commercial Discussion 


cc



Subject
[Asterisk-Users] X100P Unstable.








Hello All ,

In some ocasions i´m getting a problem with my X100P board.
I´m trying to trace tre problem , but i didn´t find a possible
answer.

-> I get those messages when trying to use Zap Channel

Sep 29 14:15:46 WARNING[-1094796368]: chan_sip.c:2107 sip_new: Unable to

allocate channel structure
Sep 29 14:15:46 NOTICE[-1094796368]: chan_sip.c:7283 handle_request: 
Unable to create/find channel

When i try to make a call to the line gives me a busy signal.

The problem only solves when i reboot the PC. :(

Best Regards,

-Jefferson Carvalho
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Re: [Asterisk-Users] Canadian Content: Telus and Shaw...

2005-01-17 Thread Kim Lux

That is a good point.  I never thought of doing that. 

They could kill SIP connections by subtly delaying any upload traffic
streams, ie introducing latency or jitter.  



On Mon, 2005-01-17 at 17:11 -0500, Sergey Kuznetsov wrote:
> If they will do it, you are welcome to write the letter to CRTC and 
> other governmental agencies
> for uncompetitive behavior.
> I think it should work.
> 
-- 
Kim Lux,  Diesel Research Inc.


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[Asterisk-Users] Re: Any interest in a Canadian Asterisk

2005-01-17 Thread David Cook

Quoting [EMAIL PROTECTED]:

> > Would it be considered trolling to start a thread on Cleaning Maple
> > Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi?
>
> Let's not forget the weekly "tooques and telephony" segment, and a
> review of
> the best block heaters for your wi-fi fones.
>

Oh, we're gonna have a good time next Thursday.

We need to get Molson Canadian to sponsor us and find Bob & Doug for the
event?

By the way, eh. It's hard to get the moose to cooperate. When you put
the parabolic antenna on his antlers you have to ride him backwards
when you're leaving your cabin, eh.

dbc.
--
David Cook

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Re: [Asterisk-Users] Canadian Content: Telus and Shaw...

2005-01-17 Thread Brandon Patterson

The CRTC is the biggest joke in the world. Ten people sit on their ass
making decisions for 30+ million and no one has ever done anything
to remove them from power. Heck even HBO is illegal in Canada. Why?
Because they few that run the country want the many to remain their captive
audience. We can go on all day about the CRTC - they represent the interests
of the large companies only.

Branson Patterson


> If they will do it, you are welcome to write the letter to CRTC and
> other governmental agencies
> for uncompetitive behavior.
> I think it should work.
>
>
> All the Best!
> Sergey.

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Re: [Asterisk-Users] Agent Status on FOP

2005-01-17 Thread steve szmidt

On Monday 10 January 2005 04:10 pm, Richard Lyman wrote:
> Joe Dennick wrote:
> > The hype and documentation for the last couple of releases of the Flash
> > Operator Panel claim that the Panel can be configured to either change
> > the LED for a phone, or the name of a phone to indicate when that phone
> > is logged into a queue.  I've tried on two different versions (0.18 and
> > 0.19) on two different systems to get this feature to work, and have been
> > completely unsuccessful.  Any hints you can provide would be greatly
> > appreciated.
> >
> > Thank you!
> >
> > Joe Dennick
> > [EMAIL PROTECTED]
>

You may of course join the list for the panel, this being for asterisk and 
all.

[EMAIL PROTECTED]

-- 

Steve Szmidt

"They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety."
Benjamin Franklin
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RE: [Asterisk-Users] ntp Server and Zultys 4X4

2005-01-17 Thread Ronald Hartmann

That was the trick.

Thanks for the assistance

Have you had success getting the park button to work with Asterisk?

Ron

-Original Message-
From: Bruce Komito [mailto:[EMAIL PROTECTED] 
Sent: Monday, January 17, 2005 11:05 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] ntp Server and Zultys 4X4

That was the hint I needed.  Try adding this to your dhcp.conf:

option time-offset -480

(-480 is for PST, -420 is mountain, etc.)



Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815


On Mon, 17 Jan 2005, Ronald Hartmann wrote:

> I have been reading the RFC http://www.faqs.org/rfcs/rfc2132.html on
> this and I think the issue may be related to the setting of the Time
> Offset
>
> 3.4. Time Offset
>
>The time offset field specifies the offset of the client's subnet
in
>seconds from Coordinated Universal Time (UTC).  The offset is
>expressed as a two's complement 32-bit integer.  A positive offset
>indicates a location east of the zero meridian and a negative
offset
>indicates a location west of the zero meridian.
>
>The code for the time offset option is 2, and its length is 4
octets.
>
> Code   LenTime Offset
>+-+-+-+-+-+-+
>|  2  |  4  |  n1 |  n2 |  n3 |  n4 |
>+-+-+-+-+-+-+
>
> Once I have time to play with this I will check it out.. any
> feedback is appreciated.
>
> Ron
>
> -Original Message-
> From: Bruce Komito [mailto:[EMAIL PROTECTED]
> Sent: Monday, January 17, 2005 9:38 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
> Discussion
> Subject: Re: [Asterisk-Users] ntp Server and Zultys 4X4
>
> For what it's worth, I'm working with Zultys trying to solve this
exact
> same problem.  So far, they've told me to take an ethernet trace,
> because
> they claim the DHCP option 42 isn't being sent, but I know this is not
> the
> case, because the phone knows the time, just not the time zone.  There
> is
> a setting in the general section of the config file called timezone,
> which
> defaults to -480 (minutes off of GMT), but this setting only seems to
> control the value that you are prompted with when the phone boots.
>
> If I get a solution, I'll let you know.
>
> Bruce Komito
> High Sierra Networks, Inc.
> www.servers-r-us.com
> (775) 236-5815
>
>
> On Mon, 17 Jan 2005, Ronald Hartmann wrote:
>
> > Good Day List,
> >
> > I have my asterisk box setup to be an ntp server, and my zultys
> > 4X4 phone  is able to get the time, however
> > I must first select the TimeZone Offset and then it will use the
> > time setting from my server.
> >
> > This is a hassle because every time the phone reboots the user
> > must answer this question and as you can imagine
> > End users do not know what to do and since the phone is not
> > booted they can not call helpdesk..
> >
> > Is there anyway to fix this.  Please excuse my ignorance if this
> > is an ntp server option I am unaware of.
> >
> > ron
> >
> >
> > ___
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> >
> > This message has been categorized as "Legitimate" by Bayesian
> Analyzer.
> > If you do not agree, please click on the link below to train the
> Analyzer.
> >
>
http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-
> 01-17%5C566bc776c215431faea5578aee92675a&C=2
> >
> > --
> >
>
---
> > This message has been inspected by DynaComm i:mail
> >
>
---
> >
> >
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[Asterisk-Users] China direct route

2005-01-17 Thread mohammad




HI;
 
 
We need China direct route over H323.
plz contcat me offline
 
MSN: [EMAIL PROTECTED]
 
 
Regards
Mohammad
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[Asterisk-Users] Re: Media Path Optimization & NAT

2005-01-17 Thread Adam Sherman

Rich Adamson wrote:
I have a bunch of setups where an Asterisk system with a public IP 
doubles as a router/gateway/firewall for a set of phones on a private 
network.

We're using external SIP providers.
Everything works quite nicely.
Now, I would very much like to remove the "canreinvite=no" from the 
provider's definition on sip.conf, but doing so causes Asterisk to send 
a re-invite to the provider pointing to a private IP. I thought that 
correct localnet entries would solve this...

Is what I'm after even possible? Am I looking in the wrong place?
By changing to canreinvite=yes, are you expecting the asterisk box to
act as a router, passing rtp traffic from your sip provider through
the box to a sip phone with a private address (without passing
asterisk code in the middle of the rtp session)?
No, sorry. I'm looking for Asterisk to not issue the re-invites if the 
two devices can't see each other. Think of mobile users who are often 
behind the corporate firewall but also travel. I'm trying to avoid 
having the media path be "user->corporate lan->pstn provider". I want it 
to be "user->pstn provider".

If not to accomplish this, what are the localnet configuration entries for?
Thanks for your response,
A.
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[Asterisk-Users] Multiple Line Caller Id With Polycom IP500

2005-01-17 Thread Matt Gibson

Greetings,
I'm wondering if it's possible to display line breaks with caller ID 
display.

I have the Polycom ip500 phone, and what I am trying to accomplish is
instead of the phone saying 'Incoming call from: name/number'
i want it to appear on the phone like this
Incoming Call from:
Menu Context last in
Name
Number
I tried using \n and \\n between the variables (${VAR} \\n ${VAR2}) but 
that did not work, not that I really expected it to.

It is displaying the correct information, just no spacing between the 
output on the display of the phone.

Does anybody know if this is possible with the Polycom phones, or even 
possible at all?

This is how I'm attempting to do it:
[macro-cidrewrite]
exten => s,1,NoOp
exten => s,2,SetVar(REWRITECALLS=${MACRO_CONTEXT} ${CALLERIDNAME} 
${CALLERIDNUM})
exten => s,3,SetCIDName(${REWRITECALLS})
exten => s,4,Dial(${ARG2},15)
exten => s,5,Voicemail(u${ARG1})
exten => s,6,Hangup
exten => s,103,Voicemail(b${ARG1})
exten => s,104,Hangup

Thanks,
Matt
--
Matt Gibson
VOIP Administrator
NJ Tech Solutions
PSTN: 1.877.999.4678 ex. 6400
FWD: 472645
IAXTEL: 1.700.761.1828
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[Asterisk-Users] VOIP CONNECTION, NO AUDIO AT THE OTHER END, NEWBIE

2005-01-17 Thread chawki hammoud

I INSTALLED FEDORA CORE AND ASTERISK YESTERDAY. THEN I
MADE A THIRD PARTY PC-PHONE CALL THROUGH VOIPJET AND
IT WENT FINE. TODAY I TRIED TO MAKE A CALL AGAIN,
PEOPLE AT THE OTHER END CAN'T HEAR ANY THING. I TESTED
MY SOUND CARD AND IT'S WORKING PROPERLY. IT SEEMS MY
CALL IS GETTING LOST SOMEWHERE IN ASTERISK AND ISN'T
LEAVING MY SERVER.
PLEASE GIVE ME ANY ADVISE OF WHAT I AM DOING WRONG.
YOUR HELP IS GREATLY APPRECIATED. 



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Re: [Asterisk-Users] Canadian Content: Telus and Shaw...

If they will do it, you are welcome to write the letter to CRTC and 
other governmental agencies
for uncompetitive behavior.
I think it should work.

All the Best!
Sergey.
Kim Lux wrote:
I called Telus before Christmas requesting some sort of VOIP connection.
Here is what I learned:
a) the guy I was talking to never heard of *
b) they didn't think there was any way that a PC could perform the
duties of a PBX
c) they told me they didn't have any VOIP connections, but then told me
that they would supply and connect Nortel PBXs using H323
d) they would not supply me with an H323 connection for *.
I don't have time to discuss this in detail, I just thought I'd share it
based on the chat in the CDN list discussion. 

We are going with babytel.  I'll advise how that works when it is up and
running, hopefully next week. 

BTW: Shaw is supposed to start supplying VOIP on a separate network from
their high speed network.  Here is the news clip:
http://www.canoe.ca/NewsStand/CalgarySun/News/2005/01/14/898082-sun.html
I find this interesting because several people have told me they are
using Shaw's high speed Internet service as the backbone of their VOIP
system. (Extreme is supposed to work even better.) 

I wonder if Telus is going to block the SIP ports on their ADSL
network ?  I wonder if Shaw will ?  (Telus presently blocks the SMPT
port so that you MUST you their mail server.)
I wonder if shaw or telus people lurk on this site.
 

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Re: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?





I would be interested in this list as well.
I have an positive experience how to get License Class A from CRTC.
As well as I am interested to talk about LNP portability.


All the Best!
Sergey.

Andrew Kohlsmith wrote:

  On January 17, 2005 04:47 pm, Jim Van Meggelen wrote:
  
  
LOL. I hadn't thought of it that way. Little vignettes amidst the
commercials?

  
  
Exactly -- It's precisely why I hang around on linux-elitists and a couple 
other oddball lists...  a good 90% of what's there is crap but man when 
something good comes by...  wowza.

  
  
Just because the volume isn't there? That might be a good thing, ya know
- have a list with, say, one or two messages a day, on average.

  
  
True, but that's why I like looking at sineapps now and again -- they 
sometimes focus on things that I've not even seen, it's interesting 
reading...  but I slug it out on the list well, just to slug it out.  :-)


  
  
It was a policy at our company that any new product implementation would
always require Technical Support be involved until several engineers,
technicians and installers were comfortable with it. I hope I always
remember the lessons learned from getting new products, and having to
develop training and implementation practices.

  
  
...

  
  
Seems that making mistakes is actually a fantastic (albeit
uncomfortable) way to learn. I sometimes wonder if I unconsciously muck
things up at first as a rite of passage.

  
  
We have a similar policy here and it really helps people understand why things 
are done a certain way when they have to field some of the customer calls 
themselves.  "right" and "wrong" take on new nuances that they would have 
otherwise been oblivious and even belligerent towards.

  
  
Nobody knows a thing so well as those who can expertly break it.

  
  
That sounds very close to "As soon as you make something idiot-proof along 
comes a better class of idiot."  :-)

  
  
Would it be considered trolling to start a thread on Cleaning Maple
Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi?

  
  
Let's not forget the weekly "tooques and telephony" segment, and a review of 
the best block heaters for your wi-fi fones.

  
  
Does that mean I'm right and you're wrong?

  
  
Yes... oh, wait...  Aughhh!

-A.
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[Asterisk-Users] Queue and Normal Transfer.

2005-01-17 Thread John Bittner

Hi,

Does anyone know how to get the normal transfer button to
work when transferring a queue call.
There seems to be a bug in app_queue that prevents calls
from reaching agents. If a call is directed to an agent, and
that agent transfers the call using the transfer facility on
Cisco phones the call is disconnected. If the agent uses #
transfers it works but the agents do not want to do blind
transfers. Sometimes they also forget to use # and hang-up
on calls.

Is there anyway to fix app_queue to get the normal transfer
buttons working. I am will to pay for this fix.

Let me know

Thanks

John Bittner
Simlab.net



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Re: [Asterisk-Users] Having trouble with T405P and PPP: ZT_SPANCONFIG failed

2005-01-17 Thread Ben Greear

Eric Wieling aka ManxPower wrote:
Adam Goryachev wrote:
On Fri, 2005-01-14 at 14:38 -0800, Ben Greear wrote:
Hello!
I am trying to set up multi-link PPP using two T100P cards in one
machine, and 1 T405P card (the 4-port one) in another machine.  I have
previously been able to get PPP working between the two T100P cards
in separate machines
The 4-port card seems to be my problem currently.  I am trying to use 
the tor2
driver (from a fresh CVS download this morning).  When I load the 
driver (or run ztcfg)
I get this error:

Start with the wct4xxp driver instead... That should get you closer ...

Or he can read the README in the Zaptel directory, which lists what card 
needs what driver.  Then he can be sure he's loading the right driver.
Yep, I should have found that.  Was looking at a text page I got with the
T100P card which unfortunately didn't have any mention of the 4-port NIC.
At any rate, I gave the wct4xxp a try.  Doesn't look too good to me:
[EMAIL PROTECTED] zaptel]# modprobe wct4xxp
Found TE410P at base address fe5dec00, remapped to f8a74c00
TE410P version c018009b, burst ON
Tried to load  into 0003, but got  instead
Tried to load 35523800 into , but got  instead
Tried to load 35523000 into 0001, but got  instead
Tried to load 07fc07fc into 0002, but got  instead
Tried to load 004a into 000a, but got  instead
Tried to load 044a into 000a, but got  instead
Tried to load 0001044a into 000a, but got  instead
Tried to load 044a into 000a, but got  instead
Tried to load 004a into 000a, but got  instead
Tried to load  into 000a, but got  instead
Tried to load 004a into 000a, but got  instead
Tried to load 844a into 000a, but got  instead
Tried to load  into 000a, but got  instead
FALC version: 00ff, Board ID: 0f
Reg 0: 0x
Reg 1: 0x
Reg 2: 0x
Reg 3: 0x
Reg 4: 0x
Reg 5: 0x
Reg 6: 0x
Reg 7: 0x
Reg 8: 0x
Reg 9: 0x
Reg 10: 0x
Found a Wildcard: Wildcard TE410P-Xilinx
ZT_SPANCONFIG failed on span 1: No such device or address (6)
FATAL: Error running install command for wct4xxp
Any ideas on this one?
Thanks,
Ben
--
Ben Greear <[EMAIL PROTECTED]>
Candela Technologies Inc  http://www.candelatech.com
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[Asterisk-Users] Canadian Content: Telus and Shaw...

2005-01-17 Thread Kim Lux

I called Telus before Christmas requesting some sort of VOIP connection.
Here is what I learned:

a) the guy I was talking to never heard of *
b) they didn't think there was any way that a PC could perform the
duties of a PBX
c) they told me they didn't have any VOIP connections, but then told me
that they would supply and connect Nortel PBXs using H323
d) they would not supply me with an H323 connection for *.

I don't have time to discuss this in detail, I just thought I'd share it
based on the chat in the CDN list discussion. 

We are going with babytel.  I'll advise how that works when it is up and
running, hopefully next week. 

BTW: Shaw is supposed to start supplying VOIP on a separate network from
their high speed network.  Here is the news clip:

http://www.canoe.ca/NewsStand/CalgarySun/News/2005/01/14/898082-sun.html

I find this interesting because several people have told me they are
using Shaw's high speed Internet service as the backbone of their VOIP
system. (Extreme is supposed to work even better.) 

I wonder if Telus is going to block the SIP ports on their ADSL
network ?  I wonder if Shaw will ?  (Telus presently blocks the SMPT
port so that you MUST you their mail server.)

I wonder if shaw or telus people lurk on this site.


-- 
Kim Lux,  Diesel Research Inc.


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Re: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

2005-01-17 Thread Andrew Kohlsmith

On January 17, 2005 04:47 pm, Jim Van Meggelen wrote:
> LOL. I hadn't thought of it that way. Little vignettes amidst the
> commercials?

Exactly -- It's precisely why I hang around on linux-elitists and a couple 
other oddball lists...  a good 90% of what's there is crap but man when 
something good comes by...  wowza.

> Just because the volume isn't there? That might be a good thing, ya know
> - have a list with, say, one or two messages a day, on average.

True, but that's why I like looking at sineapps now and again -- they 
sometimes focus on things that I've not even seen, it's interesting 
reading...  but I slug it out on the list well, just to slug it out.  :-)

> It was a policy at our company that any new product implementation would
> always require Technical Support be involved until several engineers,
> technicians and installers were comfortable with it. I hope I always
> remember the lessons learned from getting new products, and having to
> develop training and implementation practices.

...

> Seems that making mistakes is actually a fantastic (albeit
> uncomfortable) way to learn. I sometimes wonder if I unconsciously muck
> things up at first as a rite of passage.

We have a similar policy here and it really helps people understand why things 
are done a certain way when they have to field some of the customer calls 
themselves.  "right" and "wrong" take on new nuances that they would have 
otherwise been oblivious and even belligerent towards.

> Nobody knows a thing so well as those who can expertly break it.

That sounds very close to "As soon as you make something idiot-proof along 
comes a better class of idiot."  :-)

> Would it be considered trolling to start a thread on Cleaning Maple
> Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi?

Let's not forget the weekly "tooques and telephony" segment, and a review of 
the best block heaters for your wi-fi fones.

> Does that mean I'm right and you're wrong?

Yes... oh, wait...  Aughhh!

-A.
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Re: [Asterisk-Users] Media Path Optimization & NAT

2005-01-17 Thread Rich Adamson

> (This message is not a dumb NAT question!)
> 
> I have a bunch of setups where an Asterisk system with a public IP 
> doubles as a router/gateway/firewall for a set of phones on a private 
> network.
> 
> We're using external SIP providers.
> 
> Everything works quite nicely.
> 
> Now, I would very much like to remove the "canreinvite=no" from the 
> provider's definition on sip.conf, but doing so causes Asterisk to send 
> a re-invite to the provider pointing to a private IP. I thought that 
> correct localnet entries would solve this...
> 
> Is what I'm after even possible? Am I looking in the wrong place?

Let's see if I'm reading this correctly.

By changing to canreinvite=yes, are you expecting the asterisk box to
act as a router, passing rtp traffic from your sip provider through
the box to a sip phone with a private address (without passing
asterisk code in the middle of the rtp session)?

If I read the above correctly, best guess is it will be very difficult
if not impossible to accomplish.

The logic behind that guess is... sip reinvites occur on udp 5060 and
you only have one external IP address on the box. If the sip provider
sends reinvite traffic to your extern IP (intending it to go to private
address 192.168.5.6 sip phone), what is going to catch that packet and
decide where to send it? The extern IP with udp 5060 is already in use
by asterisk code.

You might be able to reconfig the asterisk box and map another registered
IP address on its external nic for each internal sip phone. Wouldn't
even care to guess how that might actually work.




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Re: [Asterisk-Users] Offtopic: improving softphone latency on Linux?

2005-01-17 Thread Steve Kann

Bruno Hertz wrote:
Hi folks
last weekend, I tried Windows Messenger first time and was stunned by
the little latency it gives. Until now, I've been using softphones on
Linux exclusively, like iaxcomm, linphone and sjphone, and they all give
me about 1, at times even 2 secs delay. Whereas Messenger really seems
to be in the millisec range.
Of course, I'm now curious why there is that difference. Clearly,
Messenger is more tightly integrated with the OS and accordingly tuned.
So where does this time go? Kernel? Application level? Web searches seem
to suggest that sound latency generally is a problem on Linux, so I
tried the low latency kernel from
http://ccrma.stanford.edu/planetccrma/software/planetccrma.html
(there are two kernels, actually, where I only got the stable version to
boot - bleeding edge didn't do on my machine).
Still, that kernel did not really improve things in a noticeable way.
Question hence: did some of you guys experience and investigate this
same issue? Any recommendations or hints how to make VoIP even more
enjoyable on linux?
 

What softphone are you using on Linux?
If you use an iaxclient-based softphone on linux as root, it runs with 
RT priority, and pretty low latency -- it certainly is much less than 2 
seconds, usually less than 100-200ms or so, depending on the network.

There's certainly room to squeeze down the path through the library, but 
there's isn't that much latency in there..

-SteveK
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Re: FW: [Asterisk-Users] Radius on *

2005-01-17 Thread Mike Tkachuk

http://voipbill.sf.net/asterisk_b2bua_v0.1.tgz
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RE: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

[EMAIL PROTECTED] wrote:
> On January 17, 2005 01:57 pm, Jim Van Meggelen wrote:
>> I'd have a hard time appreciating regulatory challenges in regions
>> that I don't have involvement with, so I assume that the reciprocal
>> is generally true as well. Granted, I shouldn't presume that everyone
>> thinks as I do, but people tend to have a limited interest in
>> subjects that don't directly affect them. I figure technical
>> discussions in Asterisk-Users are nearly always of some value, while
>> regulatory peculiarities mostly are not.
> 
> I agree with you on this, but then again I "put up" with so
> much on the -users
> list that I really don't care to listen to, especially the same tired
> discussions over and over, that I figure some regulatory
> babble might be a
> welcome change.

LOL. I hadn't thought of it that way. Little vignettes amidst the
commercials?

> That and for the half dozen or so Canuck-only threads I can
> think of off the
> top of my head it's hardly worth putting together an entire
> mailing list.

Just because the volume isn't there? That might be a good thing, ya know
- have a list with, say, one or two messages a day, on average.

>> Having said all that, I want to stress that I do not feel that I am
>> right and you are wrong. I not only respect your opinion, but am open
>> to the possibility that you are correct in your assertion.
> 
> hahaha don't worry I've got a pretty thick skin and actually
> enjoy being
> proven wrong...  makes me a better person.

It was a policy at our company that any new product implementation would
always require Technical Support be involved until several engineers,
technicians and installers were comfortable with it. I hope I always
remember the lessons learned from getting new products, and having to
develop training and implementation practices. 

I'd figure I was an expert on the ones that went smoothly, but then six
months later if someone would call me with problems, I'd discover that I
didn't know a thing. As for the ones that gave nothing but trouble, I'd
fear them, assuming I didn't know what I was doing, until it became
clear that more often than not, I knew more about them than the
manufacturer did.

The fewer the troubles, the less knowledge retained; the more painful
the task, the more god-like skills were obtained. It took me many years
to figure this out - I used to think it should be the other way around.

Seems that making mistakes is actually a fantastic (albeit
uncomfortable) way to learn. I sometimes wonder if I unconsciously muck
things up at first as a rite of passage.

Nobody knows a thing so well as those who can expertly break it.

>> For me, however, the separate list seems to have value. It is not
>> because I want to start a Canadian Club(tm), but more that I would
>> love to talk about certain subjects of interest to Canadian Asterisk
>> users, and somehow feel that Asterisk-Users is not the right venue
>> for it.
> 
> Well as I said there's so much fluff in -users already that
> until we really
> become a nuisance I don't see the harm in it at all.  Half of
> the stuff
> belongs on -biz anyway and a full third of the remaining in
> asterisk-newbies-who-refuse-to-research-before-posting...  a
> dozen or so
> threads of stuff that makes no sense to anyone outside of
> Canada wouldn't
> even register as a blip on the screen.

You're probably right. 

Would it be considered trolling to start a thread on Cleaning Maple
Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi?

> Besides my personal bet is that the CRTC is going to just
> mimic what the FCC
> does anyway (c.f. UL vs CSA, wireless regulations, etc.,
> etc.) that it would
> be great to see what the various big boys stateside have to
> say and how they
> interpret what's going on with the FCC and VOIP regulation.
> 
>> It's not a matter of right or wrong; more like personal taste,
>> really. 
> 
> Agreed.

Does that mean I'm right and you're wrong? 

;-P

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Re: [Asterisk-Users] Snom hint for ZAP channels?

2005-01-17 Thread Justin Carlson

no we have a tdm400 at this site does this still apply?

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[Asterisk-Users] Echo on SIP -- not on analog.

2005-01-17 Thread Ken D'Ambrosio

Okay, I'm stumped.  When I call the PSTN (through POTS lines), my analog 
phone phone works fine.  My SIP phones -- a Grandstream and a Polycom -- 
have major echo; roughly a .25 second delay.  Eventually, it goes away, 
which I guess is echo cancellation in action.  But, dammit, why does my 
analog phone work fine?  I've tried myriad CODECs, and various echo 
cancellation settings, to no avail.  Can anyone give me an idea of what 
I should be checking?  [Note that internal calls work fine.]

Thanks,
-Ken
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[Asterisk-Users] X-Ten lite troubles.