[Asterisk-Users] Using zap channels for fax

[Chris Mason (Lists)]

I am sending faxes from a standalone fax and going through a
TDM400 ( 2xFXS, 2xFXO ). I don’t want to send over the internet, in fact
the only reason I am going through the card at all is to capture the dialing
for call accounting.

 

Everybody tells me to use G.711 but I don’t see how
you set the codec for a Zap channel. If it is Zap => Zap (Bridging?), is
there a codec at all?

 



Chris Mason

www.anguillaguide.com



 






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RE: [Asterisk-Users] Multiple Servers and One Central Voicemail

[Anton Krall]

That’s could be a problem  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luki
Sent: Sábado, 09 de Abril de 2005 11:51 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple Servers and One Central Voicemail

Anton:

> Unfortunately, the limitation of this method is that you can't 
> differentiate between unavailible and busy messages, however you could 
> get around this by creating a busy voicemail extension as well as an 
> unavailible one (ie, prefix extention with 999 for unavailible or 998 
> for busy).

Your setup looks like it should work just fine. Remember extensions do not
have to be numeric, so you could have an extension pattern _vmb and
_vmu.

My question is: how do you get MWI to work? You know, the shutter tone or
the MWI LED indicator on certain phones?

--Luki
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Re: [Asterisk-Users] Multiple Servers and One Central Voicemail

[Anton Jackson-Smith]

Luki wrote:
Your setup looks like it should work just fine. Remember extensions do
not have to be numeric, so you could have an extension pattern
_vmb and _vmu.
 

Excellent, I didn't realise this
My question is: how do you get MWI to work? You know, the shutter tone
or the MWI LED indicator on certain phones?
 

Unfortunately I haven't considered this at all, as the phones we're 
testing with don't support either. The only thing I can think of is 
getting the voicemail server to provide a stutter dialtone via DISA, but 
that is very likely inappropriate depending on the setup

Anton
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RE: [Asterisk-Users] Multiple Servers and One Central Voicemail

[Anton Krall]

Wow! Another Anton :)

I was thinking something or the sort but you explained it nicely! Thank God
for asterisk and macros!

Thank you for the tips Anton. We Anton's rock :)

Thx man! 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Jackson-Smith
Sent: Sábado, 09 de Abril de 2005 11:27 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple Servers and One Central Voicemail

Anton Krall wrote:

>Guys.
>
>I know how to make 2 asterisk servers dial each other via IAX and such 
>but I was wondering if there is a way to only have 1 centrl voicemail 
>and not have each asterisk have its own voicemails.
>
>Is this possible?
>  
>
Hi Anton,
I'm fairly sure this is possible - I've been looking into setting up
something similar myself.
 From memory, you need to configure one server to dial a specific extension
on the other via IAX which connects to voicemail.
For example:

Voicemail server:
[inbound-iax]
exten => _999,1,Voicemail(u${EXTEN:3})

Other server:
[macro-remoteVM] ; Macro to connect to voicemail on remote system
exten => s,1,Dial(IAX2/user:[EMAIL PROTECTED]/999${ARG1},20)
exten => s,2,Playback(invalid)
exten => s,3,Hangup

[default]
exten => 1234,1,Dial(SIP/user)
exten => 1234,2,Macro(remoteVM, ${EXTEN})
exten => 1234,3,Hangup

Basically, if you dial extension 1234 on the other server and noone answers,
the remoteVM macro dials 999+the extension on the voicemail server.
The wildcard on the voicemail server recognises that you want to forward to
voicemail (from the 999 prefix) and calls voicemail on the original
extension (${EXTEN:3} removes the 999 from the front.
Unfortunately, the limitation of this method is that you can't differentiate
between unavailible and busy messages, however you could get around this by
creating a busy voicemail extension as well as an unavailible one (ie,
prefix extention with 999 for unavailible or 998 for busy).

I hope this helps, good luck with your setup, Anton Jackson-Smith
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Re: [Asterisk-Users] Multiple Servers and One Central Voicemail

[Luki]

Anton:

> Unfortunately, the limitation of this method is that you can't
> differentiate between unavailible and busy messages, however you could
> get around this by creating a busy voicemail extension as well as an
> unavailible one (ie, prefix extention with 999 for unavailible or 998
> for busy).

Your setup looks like it should work just fine. Remember extensions do
not have to be numeric, so you could have an extension pattern
_vmb and _vmu.

My question is: how do you get MWI to work? You know, the shutter tone
or the MWI LED indicator on certain phones?

--Luki
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Re: [Asterisk-Users] SPA and NAT traversal

[Michael D Schelin]

The only way you ll be able to call extension to extension is if  
Asterisk is on the same node behind the nat. like the extensions or if 
each extension is on a different node.  I run a proxie server and have 
ran through this problem many time. I bet you can call out bound to the 
outside world just fine from every extension. .

Eric Wieling wrote:
Jim Sturtevant wrote:
I was hoping someone might help me diagnose a NAT issue with an 
SPA-2000 and
my * server. 
My SPA is behind a NAT accessing a server which is also behind a NAT 
but SIP
and RTP ports are forwarded to it.

My SPA can successfully register.  It can call another extension 
which is
inside the * local net and the inside phone can call the SPA.  But, no
speech path either way.  I have NAT=YES and the two invite parameters 
are
set to NO.

I'm desperately trying to get your sip.conf file telepathically but 
all I'm getting is images from your Martha Stewart porn collection. 
*shudder*

In addition to nat=yes you also need localnet= and externip= set, as 
shown in sip.conf.sample.


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Re: [Asterisk-Users] Configuring the Sipura for static IP and registering with Asterisk.

[Michael D Schelin]

There is very little difference between configuring a static IP or 
DHCP.  You need the basic 3 things like the IP address, Sub net mask, 
and Gateway address. For DNS use the dns servers address's supplied by 
you ISP.  Make sure you turn on the use DNS setting in the Sipura unless 
you use IP address only. A sip registration failure can be many things. 
Your Service provider should have given you basic settings.  Any one of 
them can be typed in wrong. You must go over each setting. A common 
failure is the auth. password. Also make sure you use a stun server. 
Even if the IP is public, It doesn't hurt anything to use it. Most of 
the time stun servers use port 3478. So your entry should be ip-address:3478

I hope this helps a little  

Mike
Jerry wrote:
OK so now you have an IP address. Did you login and configure the Sipura?
On Apr 7, 2005, at 1:04 AM, Rich Adamson wrote:
 I wish to configure my Sipura with static 
IP. I have set the static
IP, but there is registration failure on doing so. Could you please
tell me how do I go about configuring my Sipura for static IP and 
register it successfully
with the Asterisk server.

A few of the spa changes require the box be rebooted. Did you do that?
Can you ping the sipura's ip address?
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Re: [Asterisk-Users] Multiple Servers and One Central Voicemail

[Anton Jackson-Smith]

Anton Krall wrote:
Guys.
I know how to make 2 asterisk servers dial each other via IAX and such but I
was wondering if there is a way to only have 1 centrl voicemail and not have
each asterisk have its own voicemails.
Is this possible?
 

Hi Anton,
I'm fairly sure this is possible - I've been looking into setting up 
something similar myself.
From memory, you need to configure one server to dial a specific 
extension on the other via IAX which connects to voicemail.
For example:

Voicemail server:
   [inbound-iax]
   exten => _999,1,Voicemail(u${EXTEN:3})
Other server:
   [macro-remoteVM] ; Macro to connect to voicemail on remote system
   exten => s,1,Dial(IAX2/user:[EMAIL PROTECTED]/999${ARG1},20)
   exten => s,2,Playback(invalid)
   exten => s,3,Hangup
   [default]
   exten => 1234,1,Dial(SIP/user)
   exten => 1234,2,Macro(remoteVM, ${EXTEN})
   exten => 1234,3,Hangup
Basically, if you dial extension 1234 on the other server and noone 
answers, the remoteVM macro dials 999+the extension on the voicemail server.
The wildcard on the voicemail server recognises that you want to forward 
to voicemail (from the 999 prefix) and calls voicemail on the original 
extension (${EXTEN:3} removes the 999 from the front.
Unfortunately, the limitation of this method is that you can't 
differentiate between unavailible and busy messages, however you could 
get around this by creating a busy voicemail extension as well as an 
unavailible one (ie, prefix extention with 999 for unavailible or 998 
for busy).

I hope this helps, good luck with your setup,
Anton Jackson-Smith
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RE: [Asterisk-Users] Terrible crackling on analogue line and X100Pcard

[dean collins]

> > [DC]
> >
> > Well mine is legitimate digium
> >
> > And I'm in the usa
> >
> >
> > Here is the output but I have no idea what that means?
> >
> > [EMAIL PROTECTED] root]# cat /proc/interrupts
> >CPU0
> >   0: 490763  XT-PIC  timer
> >   1:  2  XT-PIC  keyboard
> >   2:  0  XT-PIC  cascade
> >   8:  1  XT-PIC  rtc
> >   9:4885971  XT-PIC  wcfxo
> >  10:  34309  XT-PIC  eth0
> >  11:4885856  XT-PIC  wcfxo
> >  12:4886150  XT-PIC  ztdummy, usb-uhci
> >  14:   7662  XT-PIC  ide0
> > NMI:  0
> > ERR:  0
> 
> The digium card used in the US will match the telco impedance specs.
> 
> Your x100p card is on interrupt 9 by itself (not shared with any
> other i/o device). That's a good thing.
> 
> Those two items rule out a number of items in terms of what
> might be causing the crackling noise.
> 
> What * version are you using Dean? Have you tried other versions?
> 
 [DC] 
I use [EMAIL PROTECTED] V 0.8
[DC] this has been an ongoing problem so not sure if it is related to
version (as I've used asterisk head up until about 6 months ago and
version 0.4,0.6 and now 0.8 version of [EMAIL PROTECTED]
[DC] 

Cheers,
Dean 


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[Asterisk-Users] Multiple Servers and One Central Voicemail

[Anton Krall]

Guys.

I know how to make 2 asterisk servers dial each other via IAX and such but I
was wondering if there is a way to only have 1 centrl voicemail and not have
each asterisk have its own voicemails.

Is this possible?

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Re: [Asterisk-Users] Terrible crackling on analogue line and X100Pcard

[Rich Adamson]

> > What country are you in, and does the chipset on the compat card
> > support the telco standards in your country?
> 
> I'm in the UK. The card was bought in the UK, but from Ebay, so I suppose it 
> could have originated from anywhere. The card dials and answers calls 
> without a problem, so it must be doing *something* right.

Okay. There are two fairly small integrated circuits on the board.
Can you post the part numbers on those chips?

> The card reports itself as:
> 
> 00:02.0 Communication controller: Individual Computers - Jens Schoenfeld 
> Intel 537
> 
> When the wcfxo module loads, dmesg reports:
> 
> Zapata Telephony Interface Registered on major 196
> PCI: Found IRQ 11 for device :00:02.0
> Uhhuh. NMI received. Dazed and confused, but trying to continue
> You probably have a hardware problem with your RAM chips
> wcfxo: DAA mode is 'FCC'
> Found a Wildcard FXO: Generic Clone
> Registered tone zone 4 (United Kingdom)
> 
> The 3rd and 4th lines are suspicous, but I've no idea what they mean. Does 
> it refer to the system RAM or some sort of special RAM on the card? What is 
> NMI?

NMI = non maskable interrupt (or somthing like that). Those messages
would suggest there is a problem with that card and the wcfxo driver.
 
The "mode is FCC" is saying the zaptel drivers are assuming a
card that matches US telco standards. Again, without the chip set
numbers, I can't tell if that card will work correctly in the UK.
If it does not support UK standards, not likely you'll ever get
the echo to go away.

> > If the chipset doesn't match your telco standards, there is a high
> > probability you won't get rid of the echo. If it does match, then try
> > echotraining=800
> > echocancel=yes
> 
> I already use those parameters in zapata.conf, they make no difference :(

Okay, then there is about a 90% chance the card's chip set was
designed for the US telco standards. I'll be able to tell more
once you post those part numbers. 

Are you using a "opermode=UK" or anything like that in /etc/zaptel.conf?
 
> > Regarding the crackling noise, have you checked for shared
> > interrupts (cat /proc/interrupts)?
> 
> This is the output:
> 
>CPU0
>   0:  211266080  XT-PIC  timer
>   2:  0  XT-PIC  cascade
>   7: 488230  XT-PIC  eth0
>  10:2113812  XT-PIC  eth1
>  11:  211520617  XT-PIC  aacraid, wcfxo
>  14: 11  XT-PIC  ide0
> NMI:  1
> ERR: 60
> 
> It's sharing an interrupt with the RAID controller. I did try to separate 
> the interrupts when I installed the card, but any combination other than 
> that automatically assigned by the BIOS caused the Linux kernel to fail to 
> even uncompress at boot time, much less boot the system, which struck me as 
> a pretty alarming failure.

It would appear you have several interrupts that aren't being used.
Have you tried looking at the bios setup to see if you can disable
any unused interrupts (like 3 for com1 port)?

If there is nothing in the bios relative to configuring interrupts, then
you only choice is to move the card to other slots in hope of
finding one that assigns a different interrupt.

There is at least a better then 50% chance sharing the interrupt
between the wcfxo driver and the raid controller (#11) is causing
at least some of the crackling noise. You might try establishing
a call and do a large file copy (to exercise the disk) to see if
disk activity causes the noise.
 

> > Go to /usr/src/zaptel directory and run
> > ./zttest
> > Do you get something close to 100% over some period of time?
> 
> Yep:
> 
> # ./zttest
> Opened pseudo zap interface, measuring accuracy...
> 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
> 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
> 
> Is that good?

Yes, that looks good.

> > What version of asterisk are you running?
> 
> 1.0.7 plus Zaptel of the same version.

You might try going back to an earlier version (or cvs head) to
see if that has any impact on the noise.


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[Asterisk-Users] Confusion re; 407 Proxy Authentication Required

[Kevin Mayer]

I am having one hell of a time configuring 2 Asterisk boxes on a VPN
to create a distributed PBX. Pretty much everything is working as I
need it to work, except for one small thing. For some reason, if I
specify a 'secret' for any of the SIP phones on one of the Asterisk
boxes, then I get a "407 proxy authentication required" error when
trying to make a call from that phone.

To put this in perspective, SIP phones on the second Asterisk box can
make and receive calls A-OK. SIP phones on the problematic box (for
.conf files are very similar!) can log in just fine ('sip show peers'
shows that they're registered) but any attempt to make a call gives me
a 407. The funny thing is, these phones are able to receive calls no
problems, it's just the making calls. Additionally, if I remove the
'secret=' line for the relevant extension, then I get no such errors
when trying to make a call.

Me & the 2 Asterisk boxes are all on a VPN. As far as I can tell, the
relevant settings in sip.conf are exactly the same (apart from the
extension numbers) on the 2 boxes, yet they behave so differently. I
have spent quite a few hours trying to debug this and searching
through web sites and mailing lists but just can't figure it out. If
anyone can point me in the right direction, I would greatly appreciate
that!

Regards,
Kevin
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RE: [Asterisk-Users] Terrible crackling on analogue line and X100Pcard

[Rich Adamson]

> > What country are you in, and does the chipset on the compat card
> > support the telco standards in your country?
> > 
> > If the chipset doesn't match your telco standards, there is a high
> > probability you won't get rid of the echo. If it does match, then try
> >  echotraining=800
> >  echocancel=yes
> > 
> > Regarding the crackling noise, have you checked for shared
> > interrupts (cat /proc/interrupts)?
> > 
> > If you run "cat /proc/interrupts" every ten seconds, do you see
> > calculated interrupt values of about 1,000?
> > 
> > Go to /usr/src/zaptel directory and run
> >  ./zttest
> > Do you get something close to 100% over some period of time?
> > 
> > What version of asterisk are you running?
> > 
> 
> [DC] 
> 
> Well mine is legitimate digium
> 
> And I'm in the usa
> 
> 
> Here is the output but I have no idea what that means?
> 
> [EMAIL PROTECTED] root]# cat /proc/interrupts
>CPU0
>   0: 490763  XT-PIC  timer
>   1:  2  XT-PIC  keyboard
>   2:  0  XT-PIC  cascade
>   8:  1  XT-PIC  rtc
>   9:4885971  XT-PIC  wcfxo
>  10:  34309  XT-PIC  eth0
>  11:4885856  XT-PIC  wcfxo
>  12:4886150  XT-PIC  ztdummy, usb-uhci
>  14:   7662  XT-PIC  ide0
> NMI:  0
> ERR:  0

The digium card used in the US will match the telco impedance specs.

Your x100p card is on interrupt 9 by itself (not shared with any
other i/o device). That's a good thing.

Those two items rule out a number of items in terms of what
might be causing the crackling noise.

What * version are you using Dean? Have you tried other versions?


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RE: [Asterisk-Users] Terrible crackling on analogue line and X100Pcard

[dean collins]

> 
> What country are you in, and does the chipset on the compat card
> support the telco standards in your country?
> 
> If the chipset doesn't match your telco standards, there is a high
> probability you won't get rid of the echo. If it does match, then try
>  echotraining=800
>  echocancel=yes
> 
> Regarding the crackling noise, have you checked for shared
> interrupts (cat /proc/interrupts)?
> 
> If you run "cat /proc/interrupts" every ten seconds, do you see
> calculated interrupt values of about 1,000?
> 
> Go to /usr/src/zaptel directory and run
>  ./zttest
> Do you get something close to 100% over some period of time?
> 
> What version of asterisk are you running?
> 

[DC] 

Well mine is legitimate digium

And I'm in the usa


Here is the output but I have no idea what that means?

[EMAIL PROTECTED] root]# cat /proc/interrupts
   CPU0
  0: 490763  XT-PIC  timer
  1:  2  XT-PIC  keyboard
  2:  0  XT-PIC  cascade
  8:  1  XT-PIC  rtc
  9:4885971  XT-PIC  wcfxo
 10:  34309  XT-PIC  eth0
 11:4885856  XT-PIC  wcfxo
 12:4886150  XT-PIC  ztdummy, usb-uhci
 14:   7662  XT-PIC  ide0
NMI:  0
ERR:  0



Cheers,
Dean 

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[Asterisk-Users] Stanaphone - eureka

[dean collins]

EUREKA

 

I finally solved this problem, I dont know why some of the
more experienced people in here haven't answered this question (I guess they
dont use Stanaphone but here it is)

 

The problem isn't in how you register with Stanaphone but with
the AMP config :(

 

in the sip.conf folder is the line

context = from-sip-external ; Send unknown SIP callers to
this context

 

what happens when it goes to this context is

;give external sip users congestion and hangup

exten => _.,1,AbsoluteTimeout(15)

exten => _.,2,Congestion 

exten => _.,3,Hangup

 

I dont know why this is there when it isn't in the iax.conf
folder -  I'm assuming it is to stop random SIP calls.

 

So to make Stanaphone work you can do 1 of 3 things

1/ comment out this line

2/ change this line to context = from-pstn

3/ do what I did, add the following 4 lines to your
extensions.conf file (the ones in the middle

 

;


; Inbound Contexts [from]

; 

 

[from-sip-external]

 

;Stanaphone incoming extension

exten => 91514413,1,Answer

exten => 91514413,2,Goto(from-pstn,s,1) 

exten => 91514413,3,Hangup

 

;give external sip users congestion and hangup

;

exten => _.,1,AbsoluteTimeout(15)

exten => _.,2,Congestion 

exten => _.,3,Hangup

 

 

The extension number is obviously the Stanaphone phone
number

 

I didn't comment over the 'congestion' section in case it
was important.

 

And for the sip trunks configuration section here is the
information I put in but you need to customise to your own

 

 

trunk name  Stanaphone-out

 

allow=ulaw

auth=md5,plaintext

canredirect=no

disallow=all

dtmfmode=rfc2833

fromuser=91514413

host=sip.stanaphone.com

insecure=very

qualify=yes

secret=

type=peer

username=91514413

 

 

User Context    91514413

 

auth=md5

context=from-sip-external

host=sip.stanaphone.com

qualify=3000

secret=*

type=peer

username=91414413

 

Register String
91514413:[EMAIL PROTECTED]/91514413

 

 

Like I said I dont understand the AMP guys set it up this
way, might be something I dont understand but at least it is working.

 

Maybe someone else can explain the reason for hanging up on
SIP calls.

 

 

Cheers,

Dean

 

 

 

 






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Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card

[Eric Wieling]

Andrew Kohlsmith wrote:
On April 9, 2005 08:25 pm, Eric Wieling wrote:
Which specific Digium card does not use the TigerJet chip (as shown in
"lspci")?

TE405P:
05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 
01)

I imagine the TE410 and TE110 are both also similarly lspci'd.
I sit corrected.  The 4-port T-1/E-1 cards do use the Xilinx.  The 
1-port cards and 4-port TDM cards do not.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] Channel bank replacement

[Jerry]

I enjoy using the Adit 600 with the new FXS cards via the controller T1 
interfaces. Works well. I do have concerns with using the CMG card via 
MGCP. Has anyone done this? How is it working?

On Apr 8, 2005, at 12:50 PM, Matt Schulte wrote:
Word of warning, get the version 5 or higher FXS cards with the 
ADIT600,
else you will have echo problems. This is just from personal 
experience.
Supposedly the 5 and higher cards have dynamic impedance adjustment,
it's worth it.

Matt
-Original Message-
From: Peter Hoppe [mailto:[EMAIL PROTECTED]
Sent: Friday, April 08, 2005 12:23 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Channel bank replacement
Thank you so much for your answers already, I really appreciate it!
I have looked into using an Adtran Total Access 750 platform instead,
but got away from that idea after I saw the totally confusing amount of
options of different modules I can buy. The Adit 600 seemed so much
simpler to put together. Also, the Adit 600 had such an excellent
appraisal in the asterisk voip-info - see
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Channel%20Bank
But maybe I need to come back to the Adtran TA750. Unfortunately that
platform seems to only offer 24 fxs ports per unit and I need to buy an
expensive T1 card. I would buy the Digium T1 card - it seems that it is
by far the least expensive card, but $500 is still something. That's 
why

I toyed with the Adit 600 plus cmg card - all I need is a standard
network card on the Asterisk machine.
We have sorely abandoned the idea of using an extensive amount of voip
phones on the property, as we are not a homogenous office setup (ppl
also live on the property).
This solution would mean
* putting in an entire new cat5 network. I would be the person who 
would

have to put it all in place - When would I be finished? In 2 years? 4
years? 10 years?
* lots of admin hassle to enable all the phones / add new phones /
remove phones
* users can't easily extend stations at end points. With two wire phone
they simply switch one parallel to the existing one - no admin hassle /
extra hubs etc.
* two wire technology enables us to buy almost any phone available.
* security concerns with the SIP protocol. See
http://secunia.com/advisories/8169/ as an example
* users potentially plugging their laptops into the voip sockets and
browsing/downloading away => lots of setup/admin hassle with the
firewall (how do you block Kazaa?)
* Phones potentially breaking when users unplug power during firmware
download. For example, this is an issue with the Grandstream phone.
The only alternative that seems feasible at the moment would be
* a different channel bank than the adit 600 or
* a voip gateway that multiplexes many fxs ports into one ethernet
connection. But before I would go down that route I would have to be
absolutely sure that the SIP conforms to the standard, the upgrades are
free and the fxs ports are compatible with uk standard two wire phones.
I found that some two wire phones actually use 4 wires - confusing
* a bank of ATAs (handytone 286 or similar). I *really* don't like that
solution, as it is a bad botch job and throws lots of issues like which
REN they have, many power supplies (or one big one). I really ought to
be red in the face for even mentioning that solution. But if nothing
else is available, I would probably have to buy them in bulk, take the
boards out and mount them in a 19'' box together with a hub so I build
my own voip gateway :) maybe it's not so botch after all :) )
For connection to the PSTN: We have three BT lines, and again, we would
not like to move over to a different technology like ISDN. The lines
work for us, and 'if it ain't broke, don't fix it'. We would use three
Sipura SPA-3000 interfaces to connect them to the internal network. The
SPA-3000 is sold in the UK and has the CE approval, so it should 
legally

be ok. I am experimenting with one unit at the moment, and am smacked 
by

the literally hundreds of options it has. But I heard good reports 
about

that one, so I expect it to work well in our setting.

Hi Peter, I'm not sure how you are getting PSTN lines into your * box,

but if
it's not ISDN30, you might want to consider some of the cheap IAX
phones on
the market now rather than trying to soldier on with old analogue kit?
e.g. http://www.iaxtalk.com/product_info.php?cPath=1&products_id=29
Shipping for 30 units and UK power supplies was $340, and with the
weak dollar
right now, that works out at just over 40 quid per phone - I'm sure
there's
movement on the unit price when buying in bulk...
Now remove the need for an Asterisk Quad-E1 / T1 interface card and
you've
dropped the cost by nearly a grand food for thought :)
They also sell a single-ethernet-port version of the phone for $10
less if you
have enough ethernet sockets.
Cheers,
Gavin.


I got an Adtran 600 with 12 X FXO and 12 X FXS cards for $495 from
Penny Doyen [EMAIL PROTECTED] With the strength of the poun

Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card

[Andrew Kohlsmith]

On April 9, 2005 08:25 pm, Eric Wieling wrote:
> Which specific Digium card does not use the TigerJet chip (as shown in
> "lspci")?

TE405P:
05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 
01)

I imagine the TE410 and TE110 are both also similarly lspci'd.

-A.
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Re: [Asterisk-Users] Can I set queue not to hangup?

[Steve Edwards]

I'm aware of the "context=menu" feature in queue.conf.
This feature only works while the caller is "waiting" for an agent.
What I want to do is allow the caller to press "*" during the conversation 
with the agent and exit the queue application without hanging up.

On Mon, 4 Apr 2005, Richard Lyman wrote:
Steve Edwards wrote:
How can I configure "queue()" so that it does not hang up if the caller 
presses "*" to exit the queue?

I want to continue the call so the caller can choose other services.
allow the agent to be able to transfer, then create an exten in that context 
that does what you want. so the agent then presses '#xxx' (xxx = some other 
exten within that context) then the agent goes right back in the queue, and 
the caller goes on to the dialplan logic of that exten.

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Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline   [EMAIL PROTECTED]Fax: +1-760-731-3000
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Re: [Asterisk-Users] Configuring the Sipura for static IP and registering with Asterisk.

[Jerry]

OK so now you have an IP address. Did you login and configure the 
Sipura?

On Apr 7, 2005, at 1:04 AM, Rich Adamson wrote:
 I wish to configure my Sipura with static 
IP. I have set the static
IP, but there is registration failure on doing so. Could you please
tell me how do I go about configuring my Sipura for static IP and 
register it successfully
with the Asterisk server.

A few of the spa changes require the box be rebooted. Did you do that?
Can you ping the sipura's ip address?
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Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card

[Eric Wieling]

Andrew Kohlsmith wrote:
On April 9, 2005 02:13 pm, Eric Wieling wrote:
izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can
handle. Anybody had his hands on this card or knows some details ?
Please God, if you can hear me, don't let them use a TigerJet chipet.

I don't think they will; their quad T1/E1/J1 have no such POS on them.
Which specific Digium card does not use the TigerJet chip (as shown in 
"lspci")?

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] Terrible crackling on analogue line and X100Pcard

[Stuart Ford]

Rich Adamson wrote ...
What country are you in, and does the chipset on the compat card
support the telco standards in your country?
I'm in the UK. The card was bought in the UK, but from Ebay, so I suppose it 
could have originated from anywhere. The card dials and answers calls 
without a problem, so it must be doing *something* right.

I didn't *mean* to cheap out over this - I tried to buy a genuine Digium 
part, but they don't seem to do it any more and I can't find it for sale 
anywhere. The Ebay vendor claimed it was 100% compatible.

The card reports itself as:
00:02.0 Communication controller: Individual Computers - Jens Schoenfeld 
Intel 537

When the wcfxo module loads, dmesg reports:
Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 11 for device :00:02.0
Uhhuh. NMI received. Dazed and confused, but trying to continue
You probably have a hardware problem with your RAM chips
wcfxo: DAA mode is 'FCC'
Found a Wildcard FXO: Generic Clone
Registered tone zone 4 (United Kingdom)
The 3rd and 4th lines are suspicous, but I've no idea what they mean. Does 
it refer to the system RAM or some sort of special RAM on the card? What is 
NMI?

If the chipset doesn't match your telco standards, there is a high
probability you won't get rid of the echo. If it does match, then try
echotraining=800
echocancel=yes
I already use those parameters in zapata.conf, they make no difference :(
Regarding the crackling noise, have you checked for shared
interrupts (cat /proc/interrupts)?
This is the output:
  CPU0
 0:  211266080  XT-PIC  timer
 2:  0  XT-PIC  cascade
 7: 488230  XT-PIC  eth0
10:2113812  XT-PIC  eth1
11:  211520617  XT-PIC  aacraid, wcfxo
14: 11  XT-PIC  ide0
NMI:  1
ERR: 60
It's sharing an interrupt with the RAID controller. I did try to separate 
the interrupts when I installed the card, but any combination other than 
that automatically assigned by the BIOS caused the Linux kernel to fail to 
even uncompress at boot time, much less boot the system, which struck me as 
a pretty alarming failure.

If you run "cat /proc/interrupts" every ten seconds, do you see
calculated interrupt values of about 1,000?
I don't know what you mean here.
Go to /usr/src/zaptel directory and run
./zttest
Do you get something close to 100% over some period of time?
Yep:
# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
Is that good?
What version of asterisk are you running?
1.0.7 plus Zaptel of the same version.
Thanks
Stuart 

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Re: [Asterisk-Users] Any opinions on quality/service of Teliax?

[Eric Wieling]

Brian McSpadden wrote:
On Apr 9, 2005 5:03 PM, Bellows, Jared <[EMAIL PROTECTED]> wrote:
I'm looking into the TelIAX pay-as-you-go plan.  I'm assuming that they charge incoming calls minutes as well?  Is there the $0.02 connection fee for the incoming call as well?

That's the only thing they do that I could do without. But, for the
service they provide, I'll gladly pay it.
___
With Teliax I noticed that the delay between the Dial command running 
and me hearing the ringback tone is unusually long.  Not TERRIBLE, 
just unusual for a VoIP connection.  More like the delay when dialing 
out of an analog port, but they don't use analog ports.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] Any opinions on quality/service of Teliax?

[Brian McSpadden]

On Apr 9, 2005 5:03 PM, Bellows, Jared <[EMAIL PROTECTED]> wrote:
> I'm looking into the TelIAX pay-as-you-go plan.  I'm assuming that they 
> charge incoming calls minutes as well?  Is there the $0.02 connection fee for 
> the incoming call as well?


That's the only thing they do that I could do without. But, for the
service they provide, I'll gladly pay it.
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Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card

[Andrew Kohlsmith]

On April 9, 2005 02:13 pm, Eric Wieling wrote:
> izo wrote:
> > I just checked digium's site. Looks like next big thing is coming to town
> > DS3 on single card. Would be nice to know how many channels it can
> > handle. Anybody had his hands on this card or knows some details ?
>
> Please God, if you can hear me, don't let them use a TigerJet chipet.

I don't think they will; their quad T1/E1/J1 have no such POS on them.

-A.
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RE: [Asterisk-Users] Using manager interface to play aanouncments in aMeetMe

[mattf]

I've wondered about this as well. I suggest posting a bug to the bug tracker
and see if you can get a clarification or better yet, get someone to fix
this. It would be nice to override the clearing of the vars for Local
channels.

MATT---


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Friday, April 08, 2005 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin aMeetMe


A little more googling and wiki browsing shows that the default
behaviour
of the Local channel is to dump the variables.  According to the wiki I
can append /n to the channel identifier to preserve variables, but this
does
not seem to be working.  I'm running 1.0.7.

Can anyone lend me a clue?

Thanks,
Dan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin
Sent: Friday, April 08, 2005 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin aMeetMe

I've run into a snag.  I make extensive use of dynamic conferences
and thought it would be a no-brainer to pass the conference number
with the manager interface.  Looking at the wiki and sample code,
I thought I had it right, but no joy.

PHP manager call-
$res = $as->connect();
if (!$res){ echo 'Error connection to the manager!'; exit();}
$res = $as->send_request('Originate', array('Channel' =>
'Local/[EMAIL PROTECTED]', 'Context' => 'mm-announce', 'Exten' => '',
'Priority' => '1', 'Variable' => 'confNo=$confNo'));
$res = $as->disconnect();

extensions.conf-
 
[mm-announce]
exten => 9998,1,Answer
exten => 9998,2,noop,${confNo} ;to test, it was meetme(${confNo})
exten => 9998,3,Hangup

exten => ,1,Answer
exten => ,2,Playback(this-conf-will-end-in-5-minutes)
exten => ,3,Hangup

Console output-
  == Manager 'MeetMe' logged on from 127.0.0.1
-- Executing Answer("Local/[EMAIL PROTECTED],2", "") in new
stack
-- Executing NoOp("Local/[EMAIL PROTECTED],2", "") in new stack
-- Executing Hangup("Local/[EMAIL PROTECTED],2", "") in new
stack
-- Executing Answer("Local/[EMAIL PROTECTED],1", "") in new
stack
-- Executing Playback("Local/[EMAIL PROTECTED],1",
"this-conf-will-end-in-5-minutes") in new stack
-- Playing 'this-conf-will-end-in-5-minutes' (language 'en')
  == Manager 'MeetMe' logged off from 127.0.0.1

So it appears that my variable ${confNo} is not being set, or at least
honored. 

Any thoughts?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin a MeetMe

Hello,

Here's jsut a simple manager Action to send, make sure that you have an
extension set up to play the message(exten => 1234,1,Playback(file)) and
that's the extension that will be called from the meetme room. Also,
make
sure that that extension calls in to the meetme room extension with the
'q'
flag so that noone hears the welcome and leaving tone.

exten => 1234,1,Answer
exten => 1234,2,Playback(out_of_time)
exten => 1234,3,Hangup


Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: default
Exten: 1234
Priority: 1


where 78600051 is the exten to get to your meetme room.

Let me know if you have any questions,

MATT---


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 7:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin a MeetMe


A sample would be great.  I'm hoping that the Official MeetMe2
will have provisions for this, but until then I'll have a
fully functional scheduler.

Dan 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 3:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin a MeetMe

just create an extension that plays the message and hangs up and use the
manager interface to drop it into the meetme room.

Let me know if you would like an example and I'll whip one up.

We do this kind of thing in astGUIclient to play DTMF tones
automatically in
meetme rooms.

MATT---


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 6:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Using manager interface to play aanouncments
in a MeetMe


I am wrapping up a PHP addon script to my scheduling
framework and have it properly tracking and closing
conferences.

I need to play an announcement into the room that the
conference will end soon.  I haven't found a great way
to do that.  One way that I have thought of, but wou

Re: [Asterisk-Users] unlimited iax termination

[Paul]

Rich Adamson wrote:
Sure you can, in most cases. Just check the fine print in their
service agreements, or whatever else they publish. If its not
their, call them as a prospective customer. If they don't answer,
then why bother to do business with them as that's going to be
about the same level of service you get after signing up.
 

You shouldn't have to call them. If you email them and always provide a 
phone number, you are allowing them to choose the most convenient way to 
answer your questions. If they are non-responsive you just include all 
the email inquiries somewhere within your website and let the viewer 
decide. If  I read your six unanswered email inquiries to a vendor, all 
I need to see is that they were composed well and went unanswered.

Very important to find out the minimum, increment and rounding methods 
used for billing calls. From what I have seen so far, it is usually not 
found on the website or faq when it really sucks. So email them and see 
how quick they respond when they know the answer is not the one you want 
to hear.

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Re: [Asterisk-Users] Terrible crackling on analogue line and X100P card

[Rich Adamson]

> I'm experiencing terrible trouble with crackling and noise on an
> analogue line connected to an X100P (compatible) card. I've checked the
> line with a normal analogue phone and it works fine, clear as a bell,
> but any outgoing or incoming calls to Asterisk are almost completely
> drowned out by loud crackling.
> 
> I've attempted to adjust the RX and TX gains, but to no avail. There's
> also an echo, but only one way. I'm assuming this is a separate issue so
> I've not done much to investigate that, but I may be wrong so if it is
> related does anyone have any suggestions?
> 
> I never had this trouble with ISDN, but then I wouldn't would I? :)
> 
> If anyone can wave a magic wand, or at the very least point me to a
> website where I can get my own magic wand, please let me know.

What country are you in, and does the chipset on the compat card
support the telco standards in your country?

If the chipset doesn't match your telco standards, there is a high
probability you won't get rid of the echo. If it does match, then try
 echotraining=800
 echocancel=yes

Regarding the crackling noise, have you checked for shared
interrupts (cat /proc/interrupts)?

If you run "cat /proc/interrupts" every ten seconds, do you see
calculated interrupt values of about 1,000?

Go to /usr/src/zaptel directory and run
 ./zttest
Do you get something close to 100% over some period of time?

What version of asterisk are you running?


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RE: [Asterisk-Users] unlimited iax termination

[Rich Adamson]

> I am trying to put together a matrix. Please send me links, corrections,
> additions, flames, etc.
> 
> http://www.geekgazette.com/index.php?option=com_content&task=view&id=25&Item
> id=26

Go look at the list on digium's site, free world dialup's site,
the wiki, google, etc.


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Re: [Asterisk-Users] unlimited iax termination

[Rich Adamson]

> >>Serves you right for offering a bait and switch deal. If you are selling
> >>"unlimited" that's what it should be. Why would you be surprised if someone
> >>wants to use the unlimited feature?
> >>What's wrong with selling a "1000 minutes for $10" plan? I guess you are
> >>afraid someone will then offer an "unlimited" plan and take all the
> >>business! So you all offer unlimited, even though you can't deliver it and
> >>hide the real details in the fine print. So much for truth in marketing.
> >>There's laws to protect us from this kind of marketing, it's a shame they
> >>aren't used more often.
> >>
> >>
> >
> >Unfortunately, the marketing profession (world wide) has gotten to the
> >point of "how can we stretch the wording to influence a buy decision 
> >without outright lying". I'd swear a prereq for filling any marketing
> >position is for one to have experience selling used cars.
> >
> >Read the fine print for...
> > - satellite TV (HD, first months different rate, termination fee)
> > - car leases (front-end and back-end fees, milage limits)
> > - telephone company (home vs business line cost)
> > - breakfast food
> > - cisco phones (actual cost for a new working legal "sip" phone)
> > - cellular usage plans
> > - attorney's fees
> > - TV ads (the first 100 callers get two for the price of one)
> > - stock brokers (influenced by back door commissions)
> > - unlimited voip plans (411 & 1-900 costs, international calls)
> >
> >It's become common practice to state one thing and place limits (or
> >conditions) on that statement within the fine print. And, our legal 
> >system(s) seem to support that fine-print approach.
> >
> >  
> >
> 
> Can we make a list of what is the limit for each provider?

Sure you can, in most cases. Just check the fine print in their
service agreements, or whatever else they publish. If its not
their, call them as a prospective customer. If they don't answer,
then why bother to do business with them as that's going to be
about the same level of service you get after signing up.



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Re: [Asterisk-Users] Any opinions on quality/service of Teliax?

[Brian Dingman]

Yes and yes.

On Apr 9, 2005 6:03 PM, Bellows, Jared <[EMAIL PROTECTED]> wrote:
> I'm looking into the TelIAX pay-as-you-go plan.  I'm assuming that they 
> charge incoming calls minutes as well?  Is there the $0.02 connection fee for 
> the incoming call as well?
> 
> Thanks,
> Jared
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RE: [Asterisk-Users] Any opinions on quality/service of Teliax?

[Bellows, Jared]

I'm looking into the TelIAX pay-as-you-go plan.  I'm assuming that they charge 
incoming calls minutes as well?  Is there the $0.02 connection fee for the 
incoming call as well?
 
Thanks,
Jared



From: [EMAIL PROTECTED] on behalf of Mohit Muthanna
Sent: Fri 4/8/2005 12:30 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Any opinions on quality/service of Teliax?



I've been using TelIAX for a while now.

Outside of the occasional network hiccup, they certainly are one of
the better providers out there. Their customer support is also very
responsive.

Highly recommended.

Mohit.

On Apr 8, 2005 12:32 PM, Brian McSpadden <[EMAIL PROTECTED]> wrote:
> On Apr 8, 2005 9:06 AM, Jacob Cazzell <[EMAIL PROTECTED]> wrote:
> > Looking at alternative VoIP providers and I found Teliax.  One of the
> > features listed on their pay-as-you-go plan is unlimited
> > incoming/outgoing connections.
> >
> > I am working on setting up a conference calling system for some of our
> > traveling salepeople to call into for their weekly staff meetings.
> > Right now our phone system limits the number of connected conf callers
> > - this would be a perfect fit.
> >
> > There are so many VoIP providers out there, it's tough to know who's
> > good and who's not.  Any insight on Teliax is apprecaited!
> >
>
> I also have been using them for a month or so. I'll have to say,
> they're great. Very responsive support, great selection of DIDs, and
> good voice quality. I'd definitely recommend them.
>
> Brian
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--
Mohit Muthanna [mohit (at) muthanna (uhuh) com]
"There are 10 types of people. Those who understand binary, and those
who don't."
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Re: [Asterisk-Users] Terrible crackling on analogue line and X100P card

[Paul]

On Sat, 9 Apr 2005, Stuart Ford wrote:
 

Dear all ...
I'm experiencing terrible trouble with crackling and noise on an
analogue line connected to an X100P (compatible) card. I've checked the
line with a normal analogue phone and it works fine, clear as a bell,
but any outgoing or incoming calls to Asterisk are almost completely
drowned out by loud crackling.
I've attempted to adjust the RX and TX gains, but to no avail. There's
also an echo, but only one way. I'm assuming this is a separate issue so
I've not done much to investigate that, but I may be wrong so if it is
related does anyone have any suggestions?
I never had this trouble with ISDN, but then I wouldn't would I? :)
If anyone can wave a magic wand, or at the very least point me to a
website where I can get my own magic wand, please let me know.
Thanks
Stuart
   

Maybe a filter or isolation transformer is needed. Maybe both. Ask a ham 
radio enthusiast about such things. Maybe we have one reading this list?

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Re: [Asterisk-Users] Netlogic inbound DID issue

[James Taylor]

I've seen this with @home.
Either "trunk" (under amp) and then dial(sip/trunk_name/extension)
or
Dial(IAX2/user_name:[EMAIL PROTECTED]/s)
James
On Fri, 18 Mar 2005 07:08:17 -0600, Matt Schulte <[EMAIL PROTECTED]>  
wrote:

Per Mike's issue here, we're noticing this problem with older versions
of Asterisk (it would seem?), and especially distrib [EMAIL PROTECTED] As
he stated we're seeing 'No Authority Found' coming from the clients, in
[EMAIL PROTECTED] we get see the No Authority found on the server, and the 
client
sees absolutely nothing.
What's strange is I personally run CVS-head at my house, dated 11/10/04,
it has no problems at all.
If anyone has info on this please help, it's killing us :D
Matt
-Original Message-
From: Mike Clark [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 17, 2005 11:02 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Netlogic inbound DID issue
Anyone out there using NetLogic DIDs? And have inbound working? I got
outbound working, but no joy so far with inbound. Here are the relevant
parts from my conf files:
iax.conf
[general]
tos=lowdelay
jitterbuffer=no
register => username:[EMAIL PROTECTED]
[netlogic]
type=friend
host=dynamic
context=sourcekit-main
auth=plaintext
username=
secret=
disallow=all
allow=ulaw
allow=all
extensions.conf
[sourcekit-sip]
exten => 101,1,Dial(SIP/SK-101,20)
exten => 101,2,Voicemail(u101)
exten => 101,102,Voicemail(b101)
exten => 101,103,Hangup
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
[sourcekit-main]
include=>sourcekit-sip
exten => +19193233010,1,GoTo(sourcekit-sip,101,1)
exten => _1NXXNXX,1,SetCallerID(9193233010)
exten => _1NXXNXX,2,Dial(IAX2/netlogic/${EXTEN})
exten =>
_1NXXNXX,3,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN})
exten => _1NXXNXX,4,Hangup
[netlogic]
include=>sourcekit-main
and, thr debug output from * CLI:
Asterisk Ready.
*CLI> iax2 debug
IAX2 Debugging Enabled
*CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX
Subclass: NEW
   Timestamp: 00017ms  SCall: 00030  DCall: 0 [206.80.70.49:4569]
   VERSION : 2
   CALLED NUMBER   : +19193233010
   Unknown IE 045  : Present
   CALLING NUMBER  : +13362150564
   CALLING PRESNTN : 0
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   LANGUAGE: en
   CALLED CONTEXT  : netlogic
   USERNAME: username
   FORMAT  : 4
   CAPABILITY  : 2097151
   ADSICPE : 2
   DATE TIME   : 175199382
Ignoring unknown information element 'Unknown IE' (45) of length 1 Mar
17 12:35:19 NOTICE[21100]: chan_iax2.c:5419 socket_read: Rejected
connect at
tempt from 206.80.70.49, who was trying to reach '[EMAIL PROTECTED]'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
   Timestamp: 00018ms  SCall: 2  DCall: 00030 [206.80.70.49:4569]
   CAUSE   : No authority found
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--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas  75503
903-793-1956
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Re: [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems

[Dan Perik]

Pardon my answering myself (and for the long post).  But I do have it
sort of working, and I come back with information on the GS HT-488, as
well as questions related to SIP / DTMF issues.

The GS HT-488 acts as a PSTN pass through device for 4 rings.  If the
phone attached to the FXS port hasn't picked up by 4 rings, it will by
default "answer", and you're at an internal (*) dial tone.  You can also
configure the HT-488 to dial a specific extention, which it will then do
instead of dropping you at an internal dial tone.  From there you can
obviously do what ever you want with the call.  (It would be nice if you
could configure and/or disable the # rings before it switches over to
VoIP.  Maybe that will be something they will add to a firmware update
someday.) 

For dialing out, you set up an extention for the FXO port, and dial
that.  It will ring once, and then present you with the PSTN line, dial
tone and all.   From there you (should be) are able to dial out. 

Now, here is my problem and question.  Both the FXS and FXO ports are
set up to use SIP INFO for DTMF.   You would think that when you have
dialed the FXO port, and are at the PSTN dial tone, the HT-488 will
translate the SIP DTMF INFO passed through to the FXO port as audible
DTMF on the PSTN line.  This is not the case.  So I really can't make
outgoing calls yet.  Now, I can change the FXS line to send DTMF in
audio, which works, but I figure that sending DTMF in audio is not
ideal.  So I'm trying to "translate" the SIP DTMF INFO to DTMF
in-audio.  I've tried a few combinations of SipDTMFMode(inband) (trying
to do a DTMF style translation, I guess), and
Dial(SIP/gs1-FXO,10,D() ), but can't get it to work.  

Should I just suck it up and keep the FXS port using DTMF in-audio, or
is there a way to get SIP DTMF INFO translated to DTMF tones in audio in
the Dial settings for the FXO extension?

Thanks!
Dan

Dan Perik wrote:

>I just got my shiny new Grandstream HandyTone-488 today.  My goal is to
>use it to allow incoming/outgoing calls to PSTN using my normal ole'
>phone as usual.  I will be switching over to using BroadVoice as my main
>phone #, but want that to be as seemless of a switchover as possible
>(for the wife and kids, and for people needing to call us).
>
>I've got the following working:
>
>FXS -> * ( and then -> BroadVoice )
>( BroadVoice -> ) * -> FXS
>FXO -> * ( and then -> FXS )
>
>I don't have this working:
>( FXS -> ) * -> FXO
>
>In other words, I can't seem to call out on my PSTN line from Asterisk.
>
>  
>
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Re: [Asterisk-Users] "s" extension doesn't work with ata

[Scott Nelson]

On Apr 8, 2005, at 9:40 PM, Drew Einhorn wrote:
...But how do we get the intial prompt to play
on an ATA?
On many ATAs you can have it do a "hot-line" dial -- start a call when 
the phone is picked up.  Perhaps you can have your ATA dial 
"@servername" (no phone number, just the @ sign and the server name).

If you do this, this makes the phone act like a regular phone -- you 
pick up the phone and it is connected to the asterisk server, rather 
than the primary way most ATAs work -- you dial a number and it 
connects to the server after you are finished dialing.

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RE: [Asterisk-Users] SPA and NAT traversal

[Nabeel Jafferali]

> In your second option using a STUN server would I need to setup my
> own STUN server? 

No, use FWD or xten's STUN servers.

-- 
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698

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RE: [Asterisk-Users] SPA and NAT traversal

[Jim Sturtevant]

In your second option using a STUN server would I need to setup my own STUN
server?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Saturday, April 09, 2005 12:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SPA and NAT traversal

> Thank you for your reply.  There is a wealth of information on the
> wiki, etc.   I turned on RTP debug and the SPA is not sending it's
> public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP
> packets are going nowhere... 

Do I understand your question correctly:

You have an SPA behind NAT1 and * and a second SIP device behind NAT2. Both
devices register, but calls between the devices result in no audio?

If that is the case, you can do one of two things:

- set canreinvite=no for the devices' sip.conf entries, or
- teach both devices to *stop* using their internal IPs for all
communications and remove nat=yes from the entry for the SIP device inside
NAT2.

To set the SPA to give the correct IP, enable STUN, add a STUN server, and
say Yes to "Substitue VIA Addr".

-- 
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698

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Re: [Asterisk-Users] Syntax error near unexpected token 'fi'

[Luki]

> During boot I am getting an error that says the following:
> Syntax error near unexpected token 'f'i'
> /etc/rc3.d/S09zaptel line 92

Maybe you should look at line 92 in that file and see what's up with
it? Or post it here...

--Luki
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[Asterisk-Users] Syntax error near unexpected token 'fi'

[Chuck Bunn]

Hi,
During boot I am getting an error that says the following:
Syntax error near unexpected token 'f'i'
/etc/rc3.d/S09zaptel line 92
Any ideas what might be causing this? I am using Fedora 3 with latest 
Asterisk build

Thanks
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[Asterisk-Users] OT: ManxPower 2005 European Tour

[Eric Wieling]

I've helped a lot of people on the mailing lists and on IRC #asterisk. 
and wanted to let people know that I will be in Europe between May 19 
and June 21.  Stockholm (VON 2005), Brussels (holiday/vacation), 
Amsterdam (holiday/vacation), and Madrid (Astricon).  There are 
several weeks during my trip that I have no current plans for and may 
add other cities to my itinerary.

I'm looking for recommendations for lodging and tourist activities in 
all of the above cities.

I would be interested in meeting Asteriskers for drinks or coffee in 
any of these cities.

I am also looking for employment in Europe.  I would prefer the 
Benelux area, but all serious offers will be considered.  I have 
experience in a number of areas including Asterisk/SIP/IAX (2 yrs), 
Linux (10 yrs), WAN/Frame/T-1/DSL (10 yrs), and more.  I can do 
limited programming in C, Perl and PHP.

I am a citizen of the USA and want to relocate to Europe.
Eric Wieling
[EMAIL PROTECTED]
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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RE: [Asterisk-Users] SPA and NAT traversal

[Nabeel Jafferali]

> Thank you for your reply.  There is a wealth of information on the
> wiki, etc.   I turned on RTP debug and the SPA is not sending it's
> public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP
> packets are going nowhere... 

Do I understand your question correctly:

You have an SPA behind NAT1 and * and a second SIP device behind NAT2. Both
devices register, but calls between the devices result in no audio?

If that is the case, you can do one of two things:

- set canreinvite=no for the devices' sip.conf entries, or
- teach both devices to *stop* using their internal IPs for all
communications and remove nat=yes from the entry for the SIP device inside
NAT2.

To set the SPA to give the correct IP, enable STUN, add a STUN server, and
say Yes to "Substitue VIA Addr".

-- 
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698

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Re: [Asterisk-Users] SPA and NAT traversal

[Eric Wieling]

Jim Sturtevant wrote:
Thank you for your reply.  There is a wealth of information on the wiki,
etc.   I turned on RTP debug and the SPA is not sending it's public IP it is
sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere...
nat=yes makes Asterisk use the public IP that is inserted by the far 
side NAT router instead of the private IP the SIP device puts in the 
packet.

Perhaps there is a problem in your sip.conf that is causing the SPA's 
packets not to match anything.

"sip show peers" will tell you if Asterisk is seeing the public or the 
private IP of the far end SPA.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] FWD no longer doing IAX?

[r00t]

Hi,

On Apr 9, 2005 2:57 PM, Scott Wolfe <[EMAIL PROTECTED]> wrote:
> I used the iax section of
> http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
> to try and help me get this going. 

I followed the directions below, and things are still working. You
must activate iax through fwd. Check this page:

http://www.freeworlddialup.com/content/view/full/1501

> I can login to my account via thier web page so it would seem that the
> account is set up. Here is the string I am using in iax.conf. 
> register => 64:[EMAIL PROTECTED] 

register => 64:[EMAIL PROTECTED] 

My line is:
register => 64:[EMAIL PROTECTED]

iax2.fwdnet.net...

Give the instructions at the above link a go and let me know if that
works for you.

Cheers
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RE: [Asterisk-Users] SPA and NAT traversal

[Jim Sturtevant]

Thank you for your reply.  There is a wealth of information on the wiki,
etc.   I turned on RTP debug and the SPA is not sending it's public IP it is
sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere...


The SPA is behind a NAT and traversing the public IP network to get to the *
server.  It is successfully registering, thus I can ring a phone registered
locally to the * server.

I made sure localnet=192.168.2.9/255.255.255.0 (my local cfg for *)  and
externip=65.87.x.x (which is the public IP of my * server).  The * server is
behind a NAT as well with the 5060 and 16384-32767 UDP ports open.  

Based on RTP debug it appears the RTP packets are making it to the * server,
the problem is the return address is the internal NAT address of the SPA
192.168.1.100 and not it's public address.

Are you willing to share your Martha collection or are you going to keep it
to yourself? :-)


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Saturday, April 09, 2005 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA and NAT traversal

Jim Sturtevant wrote:

> I was hoping someone might help me diagnose a NAT issue with an SPA-2000
and
> my * server.  
> 
> My SPA is behind a NAT accessing a server which is also behind a NAT but
SIP
> and RTP ports are forwarded to it.
> 
> My SPA can successfully register.  It can call another extension which is
> inside the * local net and the inside phone can call the SPA.  But, no
> speech path either way.  I have NAT=YES and the two invite parameters are
> set to NO.

I'm desperately trying to get your sip.conf file telepathically but 
all I'm getting is images from your Martha Stewart porn collection. 
*shudder*

In addition to nat=yes you also need localnet= and externip= set, as 
shown in sip.conf.sample.


-- 
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card

[Remco Barende]

like it says, the equivalent of 20 E1's or 28 T1's
and I guess you know how many channels a E1 or T1 PRI is
On Sat, 9 Apr 2005, izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can handle.
Anybody had his hands on this card or knows some details ?
regards
m.
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Re: [Asterisk-Users] FWD no longer doing IAX?

[Carlos Chavez]

On Sat, 9 Apr 2005 11:57:20 -0700, Scott Wolfe wrote
> Last night I signed up for a FWD account and was 

hoping to use iax to connect thier server. I have been unable to connect as of 

yet. I get 
a:

>  

> Registration of '64' rejected: Registration 

Refused.

>  

> I used the iax section of http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD 
to 

try and help me get this 
going.

>  

> iax2 show peers gives 
me.

> Name/Username    

Host 
    

Mask 

Port  

Status
> fwd-gw/64    65.39.205.121   (S) 
 
255.255.255.255  4569  

OK (76 
ms)

>  

> I can login to my account via thier web page so 
it 

would seem that the account is set up. Here is the string I am using in 

iax.conf. 


> register => 

64:[EMAIL PROTECTED]

>  

> I set off a message to thier tech support but I 
am 

just wondering if any of you were having any 
problems.

>  

> -Scott

>  


    You have to activate IAX support by hand inside your FWD account.  Go into their webpage and find the option to activate IAX, after you select it it should take about half an hour until you can use IAX.

-- 


Carlos Chavez 


Director de Tecnología 


Telecomunicaciones Abiertas de México S.A. de C.V. 


Tel: +52-55-91169161 Ext 
2001





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RE: [Asterisk-Users] How to change language using manager interface?

[Thorben Jensen]

| 
| >How do I change the language when I do commands from the manager
| interface?
| >It seems that if I originate a call to a mailbox it will always speak
| >English. I have set the language to "da" in sip.conf general context, but
| it
| >still speaks English. I have no problems when using a phone, everything
| is
| >in Danish.
| >
| >
| 
| You have to use SetLanguage(da) in you dialplan.
| 
| >Is there a manager interface command to change language?
| >
| >Thorben

Hi,

I am looking for a way to change language from the manager interface, I
don't have a problem with the dialplan.

Thorben


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[Asterisk-Users] AgentLogin to MeetMe conference?

[Steve Edwards]

How can I configure AgentLogin to connect the agent to a MeetMe 
conference?

Or, can I achieve similar functionality through other means?
Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline   [EMAIL PROTECTED]Fax: +1-760-731-3000
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[Asterisk-Users] CallerID name lookup AGI script

[Jim Meehan]

Hi all,

My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote
an AGI script that does the following:

1) If it's a toll free number (800|888|877|866), set the CallerID name to 
"TollFree Caller"
2) Use curl to look up the number in Google phonebook
3) If a business listing, set the CallerID name to business name, as is.
4) If it's a residential listing, reverse the listing so it's last name first,
then set the CallerID name to that.
5) If there's no match in Google phonebook, look up the NPA/NXX on 
www.areacodedownload.com and set the CallerID name to "@ST RATECENTER" where 
"ST" is the two-letter state abbreviation, and "RATECENTER" is the name of 
telco rate center in that state.

Thought some of you might find this AGI script useful, so I'm including it
below.  It requires the Asterisk::AGI perl module.  

There are other reverse phone lookup sources that are more complete than
Google's, but they are harder to screen scrape.  Also, I probably could have
made this a little cleaner if I used the Google API rather than screen
scraping with curl/perl.  Please feel free to take a shot at making any of
those modifications.

Here's a snippet from my extensions.conf where it gets called:

exten => s,1,AGI(callerid.agi|${CALLERIDNUM})
exten => s,2,SetCallerId,"${googlename} <${CALLERIDNUM}>"
exten => s,3,Dial(${PHONES},30,r)
exten => s,4,Answer
exten => s,5,Wait(2)
exten => s,6,Voicemail(u3001)
exten => s,7,Hangup


And here's the script:

#!/usr/bin/perl

use Asterisk::AGI;

$AGI = new Asterisk::AGI;

$number = $ARGV[0];

if ($number =~ m/(800|888|877|866)\d{7}/) {
  $AGI->set_variable('googlename', "\"TollFree Caller\"");
  exit 0;
}

open(RESULTS, "/usr/bin/curl -s -m 2 -A Mozilla/4.0 http://www.google.com/search
?q=phonebook:$number |");

while () {
  if (m/Residential Phonebook/) {
$reverse = 1;
@fields = split(/>/);
  }
  if (m/Business Phonebook/) {
@fields = split(/>/);
  }
  if (m/did not match any/) {
@digits = split(//, $number);
$npa = $digits[0] . $digits[1] . $digits[2];
$nxx = $digits[3] . $digits[4] . $digits[5];
open(LOCATION, "/usr/bin/curl -s -m 2 -A Mozilla/4.0 http://www.areacodedown
load.com/$npa/$nxx/ |");
while () {
  if (m/>State;
$line =~ m/\"\#CACACA\">\w* (\w\w)<\/td>/;
$name = "[EMAIL PROTECTED]";
  }
  if (m/>Rate Center;
$line =~ m/\"\#CACACA\">((\w|\s)*)<\/td>/;
$name = $name . " " . $1;
  }
}
$AGI->set_variable('googlename', "\"$name\"");
exit 0;
  }
}

@result = split(/-/, $fields[35]);
chop($result[0]);
if ($reverse) {
  @words = split(/ /, $result[0]);
  $last = pop(@words);
  unshift(@words, "$last,");
  foreach $word (@words) {
$name = $name . $word . " ";
  }
}
if ($reverse == 0) {
  $name = $result[0];
}

$AGI->set_variable('googlename', "\"$name\"");
 
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[Asterisk-Users] FWD no longer doing IAX?

[Scott Wolfe]

Last night I signed up for a FWD account and was 
hoping to use iax to connect thier server. I have been unable to connect as of 
yet. I get a:
 
Registration of '64' rejected: Registration 
Refused.
 
I used the iax section of http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD to 
try and help me get this going.
 
iax2 show peers gives me.
Name/Username    
Host 
Mask 
Port  
Statusfwd-gw/64    65.39.205.121   (S)  
255.255.255.255  4569  
OK (76 ms)
 
I can login to my account via thier web page so it 
would seem that the account is set up. Here is the string I am using in 
iax.conf. 
register => 
64:[EMAIL PROTECTED]
 
I set off a message to thier tech support but I am 
just wondering if any of you were having any problems.
 
-Scott
 
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Re: [Asterisk-Users] Terrible crackling on analogue line and X100P card

[Sascha Ferley]

I am currently trying to solve this problem aswell with a TDM400p card and
going out the FXO port to the PSTN ..
If anyone runs into a solution, would be great news.
T

On Sat, 9 Apr 2005, Stuart Ford wrote:

> Dear all ...
>
> I'm experiencing terrible trouble with crackling and noise on an
> analogue line connected to an X100P (compatible) card. I've checked the
> line with a normal analogue phone and it works fine, clear as a bell,
> but any outgoing or incoming calls to Asterisk are almost completely
> drowned out by loud crackling.
>
> I've attempted to adjust the RX and TX gains, but to no avail. There's
> also an echo, but only one way. I'm assuming this is a separate issue so
> I've not done much to investigate that, but I may be wrong so if it is
> related does anyone have any suggestions?
>
> I never had this trouble with ISDN, but then I wouldn't would I? :)
>
> If anyone can wave a magic wand, or at the very least point me to a
> website where I can get my own magic wand, please let me know.
>
> Thanks
>
> Stuart
>
>
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Re: [Asterisk-Users] Asterisk as protocol conventer beetwen SIP and H.323

[Adam Rybak]

Cytowanie Sahil Gupta <[EMAIL PROTECTED]>:
> > [...]
> Hi,
> Try the OH323 implementation, we found it works better.  Everyone has
> different experiences oviously..
>
Thanks, just compiled oh323 0.6.5. But still don't know how force asterisk to
act as protocol converter.

Regards, Adam

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Re: [Asterisk-Users] SPA and NAT traversal

[Eric Wieling]

Jim Sturtevant wrote:
I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and
my * server.  

My SPA is behind a NAT accessing a server which is also behind a NAT but SIP
and RTP ports are forwarded to it.
My SPA can successfully register.  It can call another extension which is
inside the * local net and the inside phone can call the SPA.  But, no
speech path either way.  I have NAT=YES and the two invite parameters are
set to NO.
I'm desperately trying to get your sip.conf file telepathically but 
all I'm getting is images from your Martha Stewart porn collection. 
*shudder*

In addition to nat=yes you also need localnet= and externip= set, as 
shown in sip.conf.sample.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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[Asterisk-Users] Hardware dimesioning issues

[David John Walsh]

I sent this earlier today.  I didn't see my copy of the mail arrive back.

Does anyone know if I am supposed to get back any of my posts or is
there a setting I need to change.

If it has been reflected properly this morning, please accept my
applogies for the re-send.

David

--
Hello

I am in the process of putting together a short term calling card
solution that is rapidly deployable for charity events, and would
apreciate some guidence on hardware dimensioning for the solution

I have a test system running on an old P3 laptop, so in principle the
solution works : It is configured as follows:

Latest CVS of asterisk (well as of about 3 weeks ago)
AreskiCC as the card solution
Latest RPM of PostgreSQL
Latest RPM of apache
Latest RPM of php / pgphp
4 SIP accounts for the phones
1 SIP account with 4 concurrent calls for the "lines"
Sipura 1001's as the ATA, DTMF phones on the end.

It has a simple extension.conf

User dials  -> runs DeadAGI(Areskicc.php)
User goes on to enter PIN, phone number and then is connected (subject
to credit and b-number being availible)

The only difference between this test system and the production system
is the number of lines.  I need it to be able to run 80 extensions and
therefor 80 lines (presented by SIP)

How large should the processor, memory etc be - could anyone suggest a
Dell / similar system that would be good for our needs.

I don't need any zaptel hardware, as the places this is going to (its
intended to be movable - not mobile per-se but movable) will only have
outside internet connections, a local SIP provider is helping us which
is why its SIP both sides.

Thank you for your time on this matter

David
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Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card

[Eric Wieling]

izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can handle. 
Anybody had his hands on this card or knows some details ?
Please God, if you can hear me, don't let them use a TigerJet chipet.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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[Asterisk-Users] unlimited iax termination

[Jeff Glassman]

Message: 11
Date: Sat, 9 Apr 2005 08:21:16 -0700
From: "Kerry Garrison" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] unlimited iax termination
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain;   charset="us-ascii"

I am trying to put together a matrix. Please send me links, corrections,
additions, flames, etc.

http://www.geekgazette.com/index.php?option=com_content&task=view&id=25&;
Item
id=26

-Kerry

Great idea, I would like to see one for DID/800 incoming also.

Jeff


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[Asterisk-Users] DS3000P - 20 E1 capacity on single card

[izo]

I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can handle. 
Anybody had his hands on this card or knows some details ?

regards
m.
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[Asterisk-Users] Asterisk Dual Servers

[Juan Luis Moyano]

Hi all, I am trying to set up two asterisk servers (SrvA and SrvB), and 
what I want to get done is that if I dial 1X on SrvB the call must be 
routed to extension X on SrvA and if I dial 2X on SrvA the call must be 
routed to extension X on SrvB. I've read the www.voip-info.org wiki 
abouta sterisk dual servers but couldn't succeed on get it working. 
Perhaps someone that has a working dialplan similar to what I want to do 
could post his config files or explain what to do. Thanks in advance.

--
Juan Luis Moyano
[EMAIL PROTECTED]
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[Asterisk-Users] SPA and NAT traversal

[Jim Sturtevant]

I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and
my * server.  

My SPA is behind a NAT accessing a server which is also behind a NAT but SIP
and RTP ports are forwarded to it.

My SPA can successfully register.  It can call another extension which is
inside the * local net and the inside phone can call the SPA.  But, no
speech path either way.  I have NAT=YES and the two invite parameters are
set to NO.

Thoughts?


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Re: [Asterisk-Users] Asterisk as protocol conventer beetwen SIP and H.323

[Sahil Gupta]

Hi,
Try the OH323 implementation, we found it works better.  Everyone has 
different experiences oviously..

Cheers,
Sahil
On Sat, 9 Apr 2005, Adam Rybak wrote:
Hello,
  have successfully installed Asterisk 1.o with H.323 driver and made
configuration:
GW (Hardware)-> GnuGK -> Asterisk
and i call into asterisk from the PSTN network and it's work fine, but i need to
make conversion from SIP small gateways to H.323. I need to make configuration
like that:
(Normal Phones -> SIP Gateways ->) x many -> Asterisk -> GnuGK (H.323) ->
Gateway (H.323)
SIP Gateway and H.323 Gateway supports g.729 - i need the g729 codec into
Asterisk? Can i mark sip gateways that i will can see on h.323 gateway witch
from SIP gateway it comes?
Can you write sample configs for me?
Im Asterisk newbie :)
Regards,
Adam Rybak
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Re: [Asterisk-Users] how to pass G723.1

[Chetan Sarva]

Kamran Ahmad wrote:
hello
how to pass G723.1 to other side is there any
softphone using g723.1. i want to use G723.1 in my
voice communication.
 

Microsoft Netmeeting can use G723.1
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[Asterisk-Users] Asterisk as protocol conventer beetwen SIP and H.323

[Adam Rybak]

Hello,

   have successfully installed Asterisk 1.o with H.323 driver and made
configuration:
GW (Hardware)-> GnuGK -> Asterisk

and i call into asterisk from the PSTN network and it's work fine, but i need to
make conversion from SIP small gateways to H.323. I need to make configuration
like that:

(Normal Phones -> SIP Gateways ->) x many -> Asterisk -> GnuGK (H.323) ->
Gateway (H.323)
SIP Gateway and H.323 Gateway supports g.729 - i need the g729 codec into
Asterisk? Can i mark sip gateways that i will can see on h.323 gateway witch
from SIP gateway it comes?

Can you write sample configs for me?

Im Asterisk newbie :)

Regards,
Adam Rybak
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Re: [Asterisk-Users] Running a Marco from the dial command

[Chris]

Oh my gosh!   I've been staring so long at it that I didn't even see my 
typo. I was not talking about *8.I am using the prefix of 8 instead of 
9.   Like 8401234.

Regards,

Chris

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

Sent: Saturday, April 09, 2005 11:49 AM
Subject: Re: [Asterisk-Users] Running a Marco from the dial command


> > [marco-voicerec]
> > exten => s,1,noop(${ARG1})
> > exten => s,2,Background(custom/recordwarn)
> 
> A nice thought, to name macros for Mark, "marco". Won't work in the
> dialplan though.
> 
> Also, *8 is usually used for picking up a ringing phone. See features.conf.
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Re: [Asterisk-Users] unlimited iax termination

[Ronald Wiplinger]

Kerry Garrison wrote:
I am trying to put together a matrix. Please send me links, corrections,
additions, flames, etc.
http://www.geekgazette.com/index.php?option=com_content&task=view&id=25&Item
id=26
-Kerry
 

Kerry,
you did a great job, ...  (I made a bookmark of it!!!)
However, I wanted to find the real definition of "unlimited" of each 
provider.

BTW, NuFone is missing.
A suggestion to the list: Can you add the web site, please?
Again, a wonderful creation. I am sure when the providers will see it, 
they will try to get the best place,  ;-)

bye
Ronald
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Re: [Asterisk-Users] Terrible crackling on analogue line and X100P card

[Damian Funnell]

Forgot to mention - we are using
an IBM xSeries 206 Server, so the Dell riser card may not be the issue
if we are having the same problem.
FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz


Damian Funnell wrote:

  
  I have a very similar problem
that
I have been grappling with for a while.  I've got a genuine TDM400P
with four FXS ports and am using an Eicon Server quad BRI ISDN (using
CAPI) for external calls. 
  
To date we have had no luck at all in diagnosing this problem as we too
have periodic problems where the crackling occurs only sometimes, but
affects all calls that are in progress (including those using the
TDM400P and SIP calls that do not).  Asterisk does not report any
problems when this problem occurs, but it is sufficiently bad to force
everyone to terminate in-progress calls (at which time everything works
fine again).
  
Appreciate hearing if you guys find a resolution to the problem that
you are having, as we have had zero luck so far.
  
  
  
  
dean collins wrote:
  
I have a similar situation but it seems to vary from call to call
sometimes.

Using 2 digium genuine x100p's in a dell with riser card.

I'm wondering if it is something to do with the riser because it doesn't
seem to matter if I swap various cords, positions, etc.


Cheers,
Dean 




  

  -Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]] On Behalf Of Stuart Ford
Sent: Saturday, April 09, 2005 9:57 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Terrible crackling on analogue line and


X100P
  

  card

Dear all ...

I'm experiencing terrible trouble with crackling and noise on an
analogue line connected to an X100P (compatible) card. I've checked


the
  

  line with a normal analogue phone and it works fine, clear as a bell,
but any outgoing or incoming calls to Asterisk are almost completely
drowned out by loud crackling.

I've attempted to adjust the RX and TX gains, but to no avail. There's
also an echo, but only one way. I'm assuming this is a separate issue


so
  

  I've not done much to investigate that, but I may be wrong so if it is
related does anyone have any suggestions?

I never had this trouble with ISDN, but then I wouldn't would I? :)

If anyone can wave a magic wand, or at the very least point me to a
website where I can get my own magic wand, please let me know.

Thanks

Stuart


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Re: [Asterisk-Users] Call rejected by XXX: No authority found

[Wilson Pickett]

> My first szenario connects two servers via IAX2. One is static IP the second
> is a nated dnyamic host. I could register the dynamic host succesfully at
> the static one. Routing calls to it with my dialplan gets denied/rejected
> due to missing authority on the remote side. I REALLY put this up several
> times different as regarded by many examples from voip.org. Still it doesn't
> work. Please anyone give me hints where I could find some useful
> information.
Looking at CLI of the asterisk behing NAT, do you see anything trying
to authenticate when you call from another box? How would the asterisk
behind NAT ever know anyone was calling it without forwarding port
4569 to it (or at least allowing traffic inside on it)? One of the
boxes has  to begin a dialogue, and that would be the one behind NAT.
I'm not betting my life on the above, it's just a free opinion which
I'm sure 10,000 people will jump in and correct if necessary.
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Re: [Asterisk-Users] Running a Marco from the dial command

[Wilson Pickett]

> [marco-voicerec]
> exten => s,1,noop(${ARG1})
> exten => s,2,Background(custom/recordwarn)

A nice thought, to name macros for Mark, "marco". Won't work in the
dialplan though.

Also, *8 is usually used for picking up a ringing phone. See features.conf.
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Re: [Asterisk-Users] "s" extension doesn't work with ata

[Eric Wieling]

Drew Einhorn wrote:
The ATA generates it's own dialtone, and waits for
the user to dial a number, before sending anything
to the * box.  So one of the first examples in the
in the Brief Introduction to Dialplans from
Vol. 1 of the Asterisk Documentation Project.
[incoming]
exten => s,1,Answer()
exten => s,2,Playback(goodbye)
exten => s,3,Hangup()
does not work.  The ATA generates a Dialtone
and waits for the user to dial, then as soon
as the user presses some keys.  The ATA sends
that extension was not found in [incoming]
This example is elaborated into a simple example
IVR.
But how do we get the intial prompt to play
on an ATA?
In MY extensions.conf I have a comment above [incoming] that says 
something like "Calls without a destination number land here, usually 
from the PSTN".

"s" is ONLY EVER called when Asterisk doesn't know what number was 
dialed.  This (generally) only happens if a call is coming in on an 
ANALOG port, or if the call is coming in on a T-1/E-1 port that does 
not have DID/DDI service on it.

An IP Phone or ATA normally send the number dialed to Asterisk and 
therefore if you dial 5551212 then the ATA will send the call to exten 
=> 5551212,1,Blah(

Now if your ATA is not sending the correct numbers or not waiting for 
you to finish dialing then the problem is with ATA and NOT Asterisk. 
You didn't bother to tell us what ATA you are using, so I can't really 
give you any more advice.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] Dialing With Backgound Music

[Eric Wieling]

Ugur GUNCER wrote:
How can play music when is clients phone ringing in dial progress.
Usually you read the documentation.
At the Asterisk CLI do a "show applications" to show you what Asterisk 
apps are available.  Also see musiconhold.conf.sample in the Asterisk 
source directory (in the configs directory).

To see detailed help for a specific application, like Dial, do "show 
application dial".  Pay special attention to the "m" option to Dial.

Don't worry about the "t" option at this time (and don't use that 
option).  The "t" option is actually a good one.  If someone tells you 
to use it, you can pretty much assume they are a newbie and should 
take their advice with a grain of salt.  "t" and "T" are only for a 
SPECIFIC type of call transfer and most people don't need it.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] How to change language using manager interface?

[Guy Decarpentrie]

Thorben Jensen a écrit :
How do I change the language when I do commands from the manager interface?
It seems that if I originate a call to a mailbox it will always speak
English. I have set the language to "da" in sip.conf general context, but it
still speaks English. I have no problems when using a phone, everything is
in Danish.
 

You have to use SetLanguage(da) in you dialplan.
Is there a manager interface command to change language?
Thorben
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Re: [Asterisk-Users] Terrible crackling on analogue line and X100P card

[Damian Funnell]

I have a very similar problem that
I have been grappling with for a while.  I've got a genuine TDM400P
with four FXS ports and am using an Eicon Server quad BRI ISDN (using
CAPI) for external calls. 

To date we have had no luck at all in diagnosing this problem as we too
have periodic problems where the crackling occurs only sometimes, but
affects all calls that are in progress (including those using the
TDM400P and SIP calls that do not).  Asterisk does not report any
problems when this problem occurs, but it is sufficiently bad to force
everyone to terminate in-progress calls (at which time everything works
fine again).

Appreciate hearing if you guys find a resolution to the problem that
you are having, as we have had zero luck so far.




dean collins wrote:

  I have a similar situation but it seems to vary from call to call
sometimes.

Using 2 digium genuine x100p's in a dell with riser card.

I'm wondering if it is something to do with the riser because it doesn't
seem to matter if I swap various cords, positions, etc.


Cheers,
Dean 




  
  
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]] On Behalf Of Stuart Ford
Sent: Saturday, April 09, 2005 9:57 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Terrible crackling on analogue line and

  
  X100P
  
  
card

Dear all ...

I'm experiencing terrible trouble with crackling and noise on an
analogue line connected to an X100P (compatible) card. I've checked

  
  the
  
  
line with a normal analogue phone and it works fine, clear as a bell,
but any outgoing or incoming calls to Asterisk are almost completely
drowned out by loud crackling.

I've attempted to adjust the RX and TX gains, but to no avail. There's
also an echo, but only one way. I'm assuming this is a separate issue

  
  so
  
  
I've not done much to investigate that, but I may be wrong so if it is
related does anyone have any suggestions?

I never had this trouble with ISDN, but then I wouldn't would I? :)

If anyone can wave a magic wand, or at the very least point me to a
website where I can get my own magic wand, please let me know.

Thanks

Stuart


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RE: [Asterisk-Users] Terrible crackling on analogue line and X100Pcard

[Stuart Ford]

Dean Collins wrote ...

> Using 2 digium genuine x100p's in a dell with riser card.

> I'm wondering if it is something to do with the riser because 
> it doesn't seem to matter if I swap various cords, positions, etc.

Right, that's interesting. My card too is in a Dell (2550) with a riser
card. That's a pig! Has *anything* you've done improved it at all?

Stuart



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Re: [Asterisk-Users] Dialing With Backgound Music

[Giovanni Miano]

Dial(SIP/100,30,tm)

On Apr 9, 2005 5:50 PM, Ugur GUNCER <[EMAIL PROTECTED]> wrote:
> 
> How can play music when is clients phone ringing in dial progress.
> 
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RE: [Asterisk-Users] Terrible crackling on analogue line and X100P card

[dean collins]

I have a similar situation but it seems to vary from call to call
sometimes.

Using 2 digium genuine x100p's in a dell with riser card.

I'm wondering if it is something to do with the riser because it doesn't
seem to matter if I swap various cords, positions, etc.


Cheers,
Dean 




> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Stuart Ford
> Sent: Saturday, April 09, 2005 9:57 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Terrible crackling on analogue line and
X100P
> card
> 
> Dear all ...
> 
> I'm experiencing terrible trouble with crackling and noise on an
> analogue line connected to an X100P (compatible) card. I've checked
the
> line with a normal analogue phone and it works fine, clear as a bell,
> but any outgoing or incoming calls to Asterisk are almost completely
> drowned out by loud crackling.
> 
> I've attempted to adjust the RX and TX gains, but to no avail. There's
> also an echo, but only one way. I'm assuming this is a separate issue
so
> I've not done much to investigate that, but I may be wrong so if it is
> related does anyone have any suggestions?
> 
> I never had this trouble with ISDN, but then I wouldn't would I? :)
> 
> If anyone can wave a magic wand, or at the very least point me to a
> website where I can get my own magic wand, please let me know.
> 
> Thanks
> 
> Stuart
> 
> 
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[Asterisk-Users] Dialing With Backgound Music

[Ugur GUNCER]

How can play music when is clients phone ringing in dial progress.




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[Asterisk-Users] How to change language using manager interface?

[Thorben Jensen]

How do I change the language when I do commands from the manager interface?
It seems that if I originate a call to a mailbox it will always speak
English. I have set the language to "da" in sip.conf general context, but it
still speaks English. I have no problems when using a phone, everything is
in Danish.

Is there a manager interface command to change language?

Thorben


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Re: [Asterisk-Users] Terrible crackling on analogue line and X100P card

[Bill Ford]

It Stuart...Wonder if We're long lost cousins or something...Name here
is Bill Ford...

Anyway...It sounds like a "mechanical" problem. Maybe something as
simple as dirty contacts on the RJ-11 on the X100P. You say you've
checked the line...but have you replaced the cable from the demark to
the serverYou might try pulling the X100 and reseating it, also
clean the contacts on the card's rj.

On Apr 9, 2005 8:56 AM, Stuart Ford <[EMAIL PROTECTED]> wrote:
> Dear all ...
> 
> I'm experiencing terrible trouble with crackling and noise on an
> analogue line connected to an X100P (compatible) card. I've checked the
> line with a normal analogue phone and it works fine, clear as a bell,
> but any outgoing or incoming calls to Asterisk are almost completely
> drowned out by loud crackling.
> 
> I've attempted to adjust the RX and TX gains, but to no avail. There's
> also an echo, but only one way. I'm assuming this is a separate issue so
> I've not done much to investigate that, but I may be wrong so if it is
> related does anyone have any suggestions?
> 
> I never had this trouble with ISDN, but then I wouldn't would I? :)
> 
> If anyone can wave a magic wand, or at the very least point me to a
> website where I can get my own magic wand, please let me know.
> 
> Thanks
> 
> Stuart
> 
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RE: [Asterisk-Users] unlimited iax termination

[Kerry Garrison]

I am trying to put together a matrix. Please send me links, corrections,
additions, flames, etc.

http://www.geekgazette.com/index.php?option=com_content&task=view&id=25&Item
id=26

-Kerry


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RE: [Asterisk-Users] unlimited iax termination

[Chris Mason (Lists)]

Folks,
Let's try trimming the replies. I'm sick of wading through 100 lines of
reply to find a single line comment.

Chris Mason
www.anguillaguide.com


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Re: [Asterisk-Users] fax pass through on te410p

[Henry Devito]

I had the same problem at one site.  We could not receive faxes with spandsp 
reliably.  Our solution that seems to have worked with no problems so far 
was to use a SPA-2000 to a fax machine.

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

Sent: Saturday, April 09, 2005 7:16 AM
Subject: Re: [Asterisk-Users] fax pass through on te410p


Ok - point taken  - but we're running Asterisk as a SIP/PSTN gateway and 
we
don't seem to have any other noticable problems, ok fax is more sensitive.
We've tried different versions of spandsp and it does not fix anything, ok
perhaps this shows problem is not spandsp - so where/how to start looking
for a fix - any pointers anyone.

If your hardware isn't getting clean data to spandsp, why should it be
able to get clean data to a hylafax box? Unless you fix the config
problem that stops spandsp working, there is no reason to expect a
pass-through to a modem bank and hylafax to work.
Regards,
Steve
Kevin Brennan wrote:
> We are using spandsp but find it unusable in a commercial environment,
> we are looking at changing to a dedicated hylafax server using an
> eicon diva PRI/E1-30 via asterisk. We know the server on it's own is a
> reliable config our only uncertainty is how good Asterisk is at
> handling pass through fax on a te410p. Has anybody got good/bad
> experience with similar setup ?
>
> Br/Kevin Brennan
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Re: [Asterisk-Users] unlimited iax termination

[Ronald Wiplinger]

Rich Adamson wrote:
Serves you right for offering a bait and switch deal. If you are selling
"unlimited" that's what it should be. Why would you be surprised if someone
wants to use the unlimited feature?
What's wrong with selling a "1000 minutes for $10" plan? I guess you are
afraid someone will then offer an "unlimited" plan and take all the
business! So you all offer unlimited, even though you can't deliver it and
hide the real details in the fine print. So much for truth in marketing.
There's laws to protect us from this kind of marketing, it's a shame they
aren't used more often.
   

Unfortunately, the marketing profession (world wide) has gotten to the
point of "how can we stretch the wording to influence a buy decision 
without outright lying". I'd swear a prereq for filling any marketing
position is for one to have experience selling used cars.

Read the fine print for...
- satellite TV (HD, first months different rate, termination fee)
- car leases (front-end and back-end fees, milage limits)
- telephone company (home vs business line cost)
- breakfast food
- cisco phones (actual cost for a new working legal "sip" phone)
- cellular usage plans
- attorney's fees
- TV ads (the first 100 callers get two for the price of one)
- stock brokers (influenced by back door commissions)
- unlimited voip plans (411 & 1-900 costs, international calls)
It's become common practice to state one thing and place limits (or
conditions) on that statement within the fine print. And, our legal 
system(s) seem to support that fine-print approach.

 

Can we make a list of what is the limit for each provider?
The unlimited plans (regardless of which itsp) are no different. 

Technical folks are educated to define things in a clear & concise
manner, black or white, if... then..., spend time to find and
remove "exceptions" to any spec, works or doesn't work, etc.  
Marketing is almost the complete opposite, and none of us (as 
individuals) can change that with the exception of individually 
electing not to buy. Your line drawn in the sand will likely be 
at a different location then the next person's. So, it's simply 
buyer be ware!
 


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[Asterisk-Users] Terrible crackling on analogue line and X100P card

[Stuart Ford]

Dear all ...

I'm experiencing terrible trouble with crackling and noise on an
analogue line connected to an X100P (compatible) card. I've checked the
line with a normal analogue phone and it works fine, clear as a bell,
but any outgoing or incoming calls to Asterisk are almost completely
drowned out by loud crackling.

I've attempted to adjust the RX and TX gains, but to no avail. There's
also an echo, but only one way. I'm assuming this is a separate issue so
I've not done much to investigate that, but I may be wrong so if it is
related does anyone have any suggestions?

I never had this trouble with ISDN, but then I wouldn't would I? :)

If anyone can wave a magic wand, or at the very least point me to a
website where I can get my own magic wand, please let me know.

Thanks

Stuart


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Re: [Asterisk-Users] Asterisk Memory Requirements

[Tony Hoyle]

Eric Rees wrote:
MemTotal:  2074808 kB
MemFree:417420 kB
Buffers: 39396 kB
Cached:1547124 kB
SwapCached:  0 kB
Active: 471180 kB
That's a total memory usage for the entire OS of only 107MB: 
(Total-Free)-Cached.

Tony
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Re: [Asterisk-Users] UK ISDN with Asterisk

[John Daragon]

Henry Owens wrote:
John,
Thanks very much for the detailed response, that sounds pretty much like
what i'm looking for (1x BT ISDN2e and 1x analogue). Are you using one
of the Digium 4 port BRI cards, or what hardware are you using?
I'm using an AVM Fritz card with chan_capi.  They're pretty cheap on 
eBay if you're suffering sticker shock...  Of course, they're not as 
efficient as the active ones, but they're a lot cheaper and you already 
own the PC, I guess.

It would be my intention to use the ISDN primarily for incoming, and
VoIP for outgoing to cut costs, and increase functionality. You
mentioned your PSTN number is routed to you via IAX; can that number be
included in local directories?
H... Probably not.  I'm using vioptalk.org's Prepay Silver which 
allocates a geographic number. But it's *voiptalk's* number as far as 
the network is concerned, so I've no idea how you'd get it into a 
directory. I chose this because, although the company is ex-directory, I 
want people to be able to phone back so I'll show the geographic number 
in outgoing CLI. Some providers allow you to specify your own CLI on 
outgoing SIP, if that's any use.

I don't think the analogue CLI should be a problem, since the ISDN
should be taking most of the incoming calls. Does CLI work ok on the ISDN?
Oh, yes.  Of course, it's not *quite* the same. Here's an example :
Analogue CLI :01460 234068
ISDN CLI*:441460234068
So if you want to call people back, you're going to have to play with 
extensions.conf...

* Yes, I *know* that's not what it's really called...
At this point i would intend to use only 1 ISDN card, so i'll cross the
multiple card bridge when (and if) i come to it.
OK.
Drop me an email if you need any help (bearing in mind that paying 
clients get first dibs on my time!).

jd
--
John Daragon  [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127
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RE: [Asterisk-Users] Asterisk Memory Requirements

[Eric Rees]

MemTotal:  2074808 kB
MemFree:417420 kB
Buffers: 39396 kB
Cached:1547124 kB
SwapCached:  0 kB
Active: 471180 kB
Inactive:  1131508 kB
HighTotal: 1179392 kB
HighFree:   233536 kB
LowTotal:   895416 kB
LowFree:183884 kB
SwapTotal: 2031608 kB
SwapFree:  2031368 kB
Dirty: 332 kB
Writeback:   0 kB
Mapped:  37696 kB
Slab:43616 kB
Committed_AS:   126244 kB
PageTables:   1192 kB
VmallocTotal:   106488 kB
VmallocUsed:  3104 kB
VmallocChunk:   103104 kB
HugePages_Total: 0
HugePages_Free:  0
Hugepagesize: 2048 kB

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cameron
Schaus
Sent: Saturday, April 09, 2005 1:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Memory Requirements

On Fri, Apr 08, 2005 at 07:01:08PM -0500, Eric Rees wrote:
> I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB
> of memory.  This is serving about 75 sip clients, Polycom500's and
> 600's.  We are running into problems with the memory.  Asterisk, right
> now, is using about 1.8GB of system memory.  I am using Asterisk
1.0.7,
> Zaptel 1.0.7 with Digiums TE410 1xT1 RBS and 1xT1 PRI, Libpri 1.0.7 on
> Fedora Core 3.  My question; is this normal or do I need more memory
or
> is there a more serious underlying problem.

How are measuring Asterisk memory usage?  You're not counting the
memory consumed by the filesystem cache, are you?

Cam

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Re: [Asterisk-Users] oh323 on @homeasterisk

[Mike Sander]

Can you please detail the steps you have taken to successfully compile this 
on @home asterisk?

Regards
Mike
- Original Message - 
From: "CM Rahman Jr." <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 

Sent: Saturday, April 09, 2005 4:09 PM
Subject: [Asterisk-Users] oh323 on @homeasterisk


Anybody here added oh323 to @homeasterisk?  I have compiled and add the
oh323. I am wondering if anybody able to add the oh323 under web interface
AMP? If anybody did it or know how to do it, please let me know. It has
option for sip, IAX.. why not add h323 !!
Thanks
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr.

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[Asterisk-Users] Shorewall settings?

[Alexander Fitterling]

I use following settings in shorewall:

(for connections established to the firewall)

ACCEPT netfwudp   4569,5060,1:2

(all outgoing connections are permitted)

Someone, please, comment on that to attest! I appreciate...

A.Fittering


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[Asterisk-Users] sip phone extensions at a remote site

[cmould]

I am in the proscess of integrating a clients remote and head office 
phone systems. Currenty each office has their own PBX and trunk lines. I 
am recommending that they put in an Asterisk server at the Head office 
with a WAN link to the remote office and switch to IP phones.  Trunk 
lines at the remote site would  be returned to the TELCO. External calls 
over the PSTN from the remote office would be routed over the WAN to the 
head office and through Asterisk to the PSTN trunk lines. All phones 
would then become extensions (both remote and head office locations). I 
want Person A in the remote office to dial an extension number and get 
Person B in the head office. What I am unsure about is if person A and 
Person B are both at the remote site and Asterisk PBX is at the head 
office, can A and B talk directly to each pther without traversing the 
WAN link? Has anyone done this before? What is the quality of the call 
if they have? Any information is useful.

begin:vcard
fn:Carey Mould
n:Mould;Carey
org:E2 Systems Limited
adr:237 Old hope Road;;Suite 11 & 12, technology Innovation Centre;Kingston;;Kgn 6;Jamaica
email;internet:[EMAIL PROTECTED]
title:CEO/Consultant
tel;work:(876) 512-2680
x-mozilla-html:FALSE
url:http://www.e2team.com
version:2.1
end:vcard

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[Asterisk-Users] dyndns alias clients: needs to register in iax.conf as well?

[Alexander Fitterling]

One important question i ask my self is whether my asterisk server (it uses
nat, which in public uses a dns alias as well), needs to register itself
(with the register statement in iax.conf) at a host not behind a router?
Would this be mandatory in any case asterisk is behind a router, or can I
setup the peer using host = dnsalias as well, even if I use NAT?

Regards,
A.Fitterling




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[Asterisk-Users] Call rejected by XXX: No authority found

[Alexander Fitterling]

Everyone,

I beg pardon to probably demand help of what had discussed many times,
earlier. But I really stuck and earlier replies couldn't help me out.

My first szenario connects two servers via IAX2. One is static IP the second
is a nated dnyamic host. I could register the dynamic host succesfully at
the static one. Routing calls to it with my dialplan gets denied/rejected
due to missing authority on the remote side. I REALLY put this up several
times different as regarded by many examples from voip.org. Still it doesn't
work. Please anyone give me hints where I could find some useful
information.

Alex Fitterling
 

My second 

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RE: [Asterisk-Users] SIP peer doesn't report busy properly

[Florian Overkamp]

Hi Remco, 

> -Original Message-
> I'm using wengo for my outgoing calls (SIP). However, 
> whenever a number is 
> busy, asterisk plays a message that the number you dialed is not 
> available instead of a busy signal.
> 
> How can I get the 'normal' PSTN tones (like number not in use 
> tone or busy 
> tone etc)
> 
> Or is this not possible with SIP?

Yes, it is possible to signal busy. However, the signalling must be correct
through the entire path, so that also includes a requirement for your
provider (Wengo). You can start by checking what signal they return to you
by checking the DIALSTATUS variable after a Dial.

Best regards,
Florian


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Re: [Asterisk-Users] fax pass through on te410p

[Kevin Brennan]

Ok - point taken  - but we're running Asterisk as a SIP/PSTN gateway and we
don't seem to have any other noticable problems, ok fax is more sensitive.
We've tried different versions of spandsp and it does not fix anything, ok
perhaps this shows problem is not spandsp - so where/how to start looking
for a fix - any pointers anyone.

> If your hardware isn't getting clean data to spandsp, why should it be
> able to get clean data to a hylafax box? Unless you fix the config
> problem that stops spandsp working, there is no reason to expect a
> pass-through to a modem bank and hylafax to work.
>
> Regards,
> Steve
>
>
> Kevin Brennan wrote:
>
> > We are using spandsp but find it unusable in a commercial environment,
> > we are looking at changing to a dedicated hylafax server using an
> > eicon diva PRI/E1-30 via asterisk. We know the server on it's own is a
> > reliable config our only uncertainty is how good Asterisk is at
> > handling pass through fax on a te410p. Has anybody got good/bad
> > experience with similar setup ?
> >
> > Br/Kevin Brennan
>
>
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Re: [Asterisk-Users] How many FXS/FXO ports do I need?

[Rich Adamson]

> I'm new to phone systems and phone wiring and I couldn't find an answer
> to this question on the wiki or google.
> 
> My understanding is that a standard residential/business phone line
> carries the signal over 2 wires.  Your 4-wire RJ11 wiring supports 2
> phone lines.  Given that each line takes 2 wires, and there are 8 wires
> in an FXO port, can I conceivably support 4 phone lines on one FXO port?
> 
> On the phone/FXS side of things, can you also have multiple lines per
> FXS port?
> 
> If I want to hookup 5 phones to my residential phone service with 2
> lines, what # of FXO & FXS ports do I need?
> 
> Thanks for your clarification...

Others have already addressed most of your questions.

You have lots of different choices on how you address 5 phones and
2 lines. A couple choices include:

- use the digium TDM card with two fxo modules (to connect to the
  two pstn lines), and one fxs module (to ring all of the five phones)

- use two Sipura spa-3000 adapters (each adapter can support one pstn
  line and one fxs port, and you decide whether the two fxs ports
  provided on the adapters have one or more of the five phones
  attached to them. eg, business vs home phones.

- replace all of your analog phones with voip sip phones (and ethernet
  wiring). No need for fxs ports.

I kind of like the spa-3000 approach since those adapters allow
you to make a decision on how you want your system to function,
while also allowing you to change your mind and support your phones
in a different way at some later date. Lots of little features built 
into those boxes.

If you use the TDM card approach, any time your asterisk system is
down (for any reason), your phones are all down. That will likely be
a problem for you as you learn how to do things with asterisk, and
some of those "things" require you to stop asterisk & restart it.

If you use the spa-3000 approach, you can configure the boxes to
have all incoming pstn calls ring through to your phones (even when
asterisk is down, or AC power is down).


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RE: [Asterisk-Users] unlimited iax termination

[Rich Adamson]

> Serves you right for offering a bait and switch deal. If you are selling
> "unlimited" that's what it should be. Why would you be surprised if someone
> wants to use the unlimited feature?
> What's wrong with selling a "1000 minutes for $10" plan? I guess you are
> afraid someone will then offer an "unlimited" plan and take all the
> business! So you all offer unlimited, even though you can't deliver it and
> hide the real details in the fine print. So much for truth in marketing.
> There's laws to protect us from this kind of marketing, it's a shame they
> aren't used more often.

Unfortunately, the marketing profession (world wide) has gotten to the
point of "how can we stretch the wording to influence a buy decision 
without outright lying". I'd swear a prereq for filling any marketing
position is for one to have experience selling used cars.

Read the fine print for...
 - satellite TV (HD, first months different rate, termination fee)
 - car leases (front-end and back-end fees, milage limits)
 - telephone company (home vs business line cost)
 - breakfast food
 - cisco phones (actual cost for a new working legal "sip" phone)
 - cellular usage plans
 - attorney's fees
 - TV ads (the first 100 callers get two for the price of one)
 - stock brokers (influenced by back door commissions)
 - unlimited voip plans (411 & 1-900 costs, international calls)

It's become common practice to state one thing and place limits (or
conditions) on that statement within the fine print. And, our legal 
system(s) seem to support that fine-print approach.

The unlimited plans (regardless of which itsp) are no different. 

Technical folks are educated to define things in a clear & concise
manner, black or white, if... then..., spend time to find and
remove "exceptions" to any spec, works or doesn't work, etc.  
Marketing is almost the complete opposite, and none of us (as 
individuals) can change that with the exception of individually 
electing not to buy. Your line drawn in the sand will likely be 
at a different location then the next person's. So, it's simply 
buyer be ware!


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Re: [Asterisk-Users] UK ISDN with Asterisk

[Henry Owens]

John,

Thanks very much for the detailed response, that sounds pretty much like
what i'm looking for (1x BT ISDN2e and 1x analogue). Are you using one
of the Digium 4 port BRI cards, or what hardware are you using?

It would be my intention to use the ISDN primarily for incoming, and
VoIP for outgoing to cut costs, and increase functionality. You
mentioned your PSTN number is routed to you via IAX; can that number be
included in local directories?

I don't think the analogue CLI should be a problem, since the ISDN
should be taking most of the incoming calls. Does CLI work ok on the ISDN?

At this point i would intend to use only 1 ISDN card, so i'll cross the
multiple card bridge when (and if) i come to it.

Thanks again!
Henry.

John Daragon wrote:
> Henry Owens wrote:
>> Hi,
>>
>>
>>>Get yourself a 'Fritz!Card PCI' - also marketed by BT themselves as a
>>>'BT Speedway ISDN' adapter - these seem to be the most cheap and
>>>supported of low-end ISDN2 adapters
>>
>>
>> Will do - they seem pretty inexpensive (even for the BT Speedway card is
>> only about £35). From doing a bit of poking, SuSE 9.1 seems to be the
>> latest OS for which drivers are available. Is anyone using one of these
>> cards successfully, and if so, on SuSE?
>>
>> One more question (and probably a pretty basic one, but i'm not that
>> familiar with PSTNs) - will i need two of these cards in order to use
>> both channels?
>>
>> Looking forward to getting this going now, and much more confident,
>> thanks for your support!
> 
> Henry, hi;
> 
> I'm running a small Asterisk PABX under SuSE 9.1. I have one analogue
> (PSTN) line, a single ISDN2e connection (i.e. 2 channels - one adaptor
> card) and a London PSTN number which gets routed to me via IAX, and I
> support 2 internal SIP phones and 4 internal analogue handsets. DID and
> whatever CLID is called in ISDN work fine. CLI on the analogue line is a
> nightmare because the Digium hardware doesn't support BT's CLI, so I
> have a modem picking that up and inserting it into Asterisk with (so
> far) variable results.  Outgoing calls go either via the landlines, or
> via the Docklands-terminated IAX channel.
> 
> All works pretty well - looks like just the sort of solution your client
> may need. Do be aware that supporting multiple ISDN2e cards might
> problematic. Not impossible, but problematic...
> 
> jd
> 
> --
> 
> 
> John Daragon  [EMAIL PROTECTED]
> argv[0] limited
> Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
> v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127
> 
> 
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Re: [Asterisk-Users] UK ISDN with Asterisk

[John Daragon]

Henry Owens wrote:
Hi,

Get yourself a 'Fritz!Card PCI' - also marketed by BT themselves as a
'BT Speedway ISDN' adapter - these seem to be the most cheap and
supported of low-end ISDN2 adapters

Will do - they seem pretty inexpensive (even for the BT Speedway card is
only about £35). From doing a bit of poking, SuSE 9.1 seems to be the
latest OS for which drivers are available. Is anyone using one of these
cards successfully, and if so, on SuSE?
One more question (and probably a pretty basic one, but i'm not that
familiar with PSTNs) - will i need two of these cards in order to use
both channels?
Looking forward to getting this going now, and much more confident,
thanks for your support!
Henry, hi;
I'm running a small Asterisk PABX under SuSE 9.1. I have one analogue 
(PSTN) line, a single ISDN2e connection (i.e. 2 channels - one adaptor 
card) and a London PSTN number which gets routed to me via IAX, and I 
support 2 internal SIP phones and 4 internal analogue handsets. DID and 
whatever CLID is called in ISDN work fine. CLI on the analogue line is a 
nightmare because the Digium hardware doesn't support BT's CLI, so I 
have a modem picking that up and inserting it into Asterisk with (so 
far) variable results.  Outgoing calls go either via the landlines, or 
via the Docklands-terminated IAX channel.

All works pretty well - looks like just the sort of solution your client 
may need. Do be aware that supporting multiple ISDN2e cards might 
problematic. Not impossible, but problematic...

jd
--
John Daragon  [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127
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[Asterisk-Users] HOW TO SET THE TIME TO DIAL AFTER astcc-accountnum and astcc-phonenum

[wassim darwish]

when a call comes the astcc-accountnum plays and ask
the caller about the card number and after playing
astcc-accountnum a period of time is given for the
caller  to dial his card number but the problem here
is the short of the time given ,and i dont know where
and how can i setup the time. 





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Re: [Asterisk-Users] fax pass through on te410p

[Steve Underwood]

If your hardware isn't getting clean data to spandsp, why should it be 
able to get clean data to a hylafax box? Unless you fix the config 
problem that stops spandsp working, there is no reason to expect a 
pass-through to a modem bank and hylafax to work.

Regards,
Steve
Kevin Brennan wrote:
We are using spandsp but find it unusable in a commercial environment, 
we are looking at changing to a dedicated hylafax server using an 
eicon diva PRI/E1-30 via asterisk. We know the server on it's own is a 
reliable config our only uncertainty is how good Asterisk is at 
handling pass through fax on a te410p. Has anybody got good/bad 
experience with similar setup ?
 
Br/Kevin Brennan

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Re: [Asterisk-Users] UK ISDN with Asterisk

[Henry Owens]

Hi,

Thanks for the tip - is there a better ISDN card (i don't mind paying
extra) for compatibility with Asterisk? Is there any Digium hardware
that will do what i need to do? I'm basically looking for a really
reliable solution, with (relatively) easy setup and good compatibility,
and don't mind paying more for it.

-- Henry.
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