date:20070912


Is there a way to generate an old-fashioned dial tone with Asterisk?

I'm thinking of one that sounds like:

http://www.seg.co.uk/telecomm/dialtone.wav

-- 
Phil Reynolds
  o   mail: [EMAIL PROTECTED]
|L_ \  / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/  Waltham 66, Emley Moor 69, Droitwich 79, Windows 95


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Re: [asterisk-users] Generating an old-fashioned dialtone

2007-09-12 Thread Clayton Milos

- Original Message - 
From: Phil Reynolds [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 8:57 AM
Subject: [asterisk-users] Generating an old-fashioned dialtone

 Is there a way to generate an old-fashioned dial tone with Asterisk?

 I'm thinking of one that sounds like:

 http://www.seg.co.uk/telecomm/dialtone.wav

 -- 
 Phil Reynolds

As far as I know dialtone with SIP can only be generated on the handsets.
We're using Cisco 7960's with SIP firmware on them and they generate a 
dialtone.

-Clay

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Re: [asterisk-users] Linux-HA and Asterisk

2007-09-12 Thread Chris Mason (Lists)

Mike Clark wrote:

 Yes, the Asterisk boxes were on private addresses. The Polycoms are also 
 behind a NAT. Yes, I tried using externip in sip.conf and this allowed 
 registration, and calls to be placed, but no audio. Unfortunately, 
 Polycom does not support STUN.

Your problem is not Linux-HA, it looks like that is fully functional.
Your problem is the same one many people come across. You can't put
Polycom phones behind NAT, it won't work.
If you have to have the phones behind NAT, which I advise against, use
Linksys which probably work better, and use a SIP aware NAT device.
Better still, put the phones on the same network as the Asterisk PBX and
say goodbye to your problems.

-- 
Chris Mason
Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670
International:  (305) 704-7249 Fax: (815)301-9759
Yahoo IM only: [EMAIL PROTECTED]

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Re: [asterisk-users] IAX2 NAT issues

2007-09-12 Thread Tim H. Panton

I guess that you just need to add a rule to your simple router's config
that permits udp 4569 from asterisk outbound to any IP address.

Tim

- Original Message -
From: Perssy Llamosas [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Tuesday, September 11, 2007 11:06:09 PM (GMT) Europe/London
Subject: [asterisk-users]  IAX2 NAT issues

Hello,

I am playing around with IAX2 and I have encountered a problem trying to 
setup an asterisk box through NAT using IAX2.

This is the problem:

Asterisk box = Advanced Firewall = Internet = User's router = User

The user can register, the server can answer, calls can be made.

Asterisk box = Very simple router = Internet = User's router = User

User's packet reach the server, the server cannot reply because the udp 
connection is lost, several RX retry TX retry, user is unable to call.

In both cases the firewall and the router are forwarding the port 4569 
to Asterisk, user's router is not forwarding anything, the user has 
qualify=yes to maintain the connection open but the very simple router 
will drop the connection before Asterisk can reply to the packet.

So I ask the list: Is there a way to overcome this problem? Udp 
connection timeout in Asterisk? Should I get a new router?

Thanks,

PLL.

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Re: [asterisk-users] 56k modem configuration

2007-09-12 Thread Tobias Wolf

Matthew Fredrickson schrieb:
 Andrea Spadaccini wrote:
 Ciao Matthew,

 I would be very surprised if chan_modem actually works... I don't think 
 I've *ever* seen it setup before.
 Well.. So there's no hope to make that modem work with Asterisk, right?
 
 Unless someone speaks otherwise, I would say that the most accurate 
 answer is, your mileage may vary, but don't hope for a lot :-)
 

I dont't think that chan_modem was designed for *analog modems*. See here:

http://www.voip-info.org/wiki/index.php?page=Asterisk+Modem+channels

Anyway it seems to be depreceated.

Cheers,

-- 

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  Leiter Softwareentwicklung / Kommunikationslösungen

  Evision GmbH



  Wittekindstr. 105

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  Fax: +49 (0)231 - 47790 500

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Re: [asterisk-users] Generating an old-fashioned dialtone

2007-09-12 Thread Gordon Henderson

On Wed, 12 Sep 2007, Clayton Milos wrote:

 - Original Message -
 From: Phil Reynolds [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Sent: Wednesday, September 12, 2007 8:57 AM
 Subject: [asterisk-users] Generating an old-fashioned dialtone

 Is there a way to generate an old-fashioned dial tone with Asterisk?

 I'm thinking of one that sounds like:

 http://www.seg.co.uk/telecomm/dialtone.wav

 As far as I know dialtone with SIP can only be generated on the handsets.
 We're using Cisco 7960's with SIP firmware on them and they generate a
 dialtone.

Zap channels don't though, and DISA also generates an internal dialtone, 
so if you arranged things such that lifting the handset auto-dialled a 
number on a SIP phone which connected you to DISA, then you could fiddle 
with the runes in indications.conf ...

However producing the olde purr might be challenging and you'll be 
after a phone with buttons A and B next ;-)

Gordon

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Re: [asterisk-users] Generating an old-fashioned dialtone


Quoting Clayton Milos [EMAIL PROTECTED]:
 Is there a way to generate an old-fashioned dial tone with Asterisk?

 I'm thinking of one that sounds like:

 http://www.seg.co.uk/telecomm/dial tone.wav

 As far as I know dialtone with SIP can only be generated on the handsets.
 We're using Cisco 7960's with SIP firmware on them and they generate a
 dialtone.

As far as I know I didn't mention generating it as a dialtone on a SIP  
phone, merely generating the tone.

I can probably put it on Zap phones easily enough if I wish, but I'd  
need to know how to generate it first, and all I am after right now is  
the sound.

-- 
Phil Reynolds
  o   mail: [EMAIL PROTECTED]
|L_ \  / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/  Waltham 66, Emley Moor 69, Droitwich 79, Windows 95


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[asterisk-users] AAI2UUI - how?

2007-09-12 Thread Christophorus Laube

Hi list,

on my asterisk machine I have an E1 (Beronet with chan_misdn) board and 
sip clients connected. I am getting some AAI 
(application-to-application-information, enriched SIP header, similar to 
the SipAddHeader application) from a sip client during the BYE method. I 
want to give this AAI to my ISDN line as UUI (user-to-user-information) 
during ISDN Hangup. Doing that with the SipGetHeader application is not 
possible as this is only allowed on incoming SIP calls. Is there a 
possibility I can customize my cdr in a manner that logs this AAI and I 
can strip that in the hangup extensions from the cdr to set the 
MISDN_USERUSER variable and write UUI?
TIA and Regards, Christophorus

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[asterisk-users] TE405P intermittent yellow alarm

2007-09-12 Thread Richard van der Hoff

Folks,

I really hope you can help me here - I'm beginning to tear my hair out!

About 10 days ago my company moved to a new office. As a result of this,
we've plugged our PBX box, which has happily been running for the last
three years, into our new E1 line. Since then, I've been seeing
intermittent yellow alarms. Obviously, since this was working fine in
the old office, the thing to suspect is the new line - but the telco
(British Telecom) aren't really helping much.

The box is a 2.4GHz Intel box, with a TE405P installed in it (we're only
using one of the spans). I'm using the zaptel drivers version 1.4.4
(I've also tried 1.0.2 with similar results). zaptel.conf has:

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

Essentially it runs fine for a few hours; then zttool reports yellow
alarm and calls can be made neither in nor out. After a while this
clears itself again and all is well for another few hours.

At this point, I'd really like to know what a yellow alarm actually
means. I've read that it indicates that that the other end of the E1 is
in an alarm condition: however BT's terminating unit seems quite happy
with no alarm conditions at all.

So, really hoping that someone can shed some light on what this might
all mean.

Cheers,

Richard


-- 
Richard van der Hoff [EMAIL PROTECTED]
Project Manager
Tel: +44 (0) 845 666 7778
http://www.mxtelecom.com


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Re: [asterisk-users] TDM400P periodic sound clicks on FXS

2007-09-12 Thread Costa Tsaousis

Tzafrir Cohen wrote:
 How about calls from either the card or the trunk to an echo test
 extension? to a local SIP/IAX phone?

   
It seems that the FXS slots do no have the issue with local VoIP phones.

Any ideas?



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[asterisk-users] TDM2400P: Power alarm error on boot

2007-09-12 Thread gincantalupo

Hi,
I have an Asterisk PBX equipped with (a Sangoma PRI card and) a Digium 
TDM2400P.
I got this error inside /var/log/messages:Power alarm on module 
8, resetting!
I rebooted the PBX and this time I got:Power alarm on module 
7, resetting!

Please, does anybody know what it means? can it be a TDM2400P broken module?

Thank you.

Giorgio.

-- 

_
Giorgio Incantalupo, mailto:[EMAIL PROTECTED]
FGA srl - http://www.fgasoftware.com -
[EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172  


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Re: [asterisk-users] Linux-HA and Asterisk

2007-09-12 Thread Mike Clark

Chris Mason (Lists) wrote:
 Mike Clark wrote:

   
 Yes, the Asterisk boxes were on private addresses. The Polycoms are also 
 behind a NAT. Yes, I tried using externip in sip.conf and this allowed 
 registration, and calls to be placed, but no audio. Unfortunately, 
 Polycom does not support STUN.
 

 Your problem is not Linux-HA, it looks like that is fully functional.
 Your problem is the same one many people come across. You can't put
 Polycom phones behind NAT, it won't work.
 If you have to have the phones behind NAT, which I advise against, use
 Linksys which probably work better, and use a SIP aware NAT device.
 Better still, put the phones on the same network as the Asterisk PBX and
 say goodbye to your problems.

   
Thanks Chris. Unfortunately, these solutions aren't an option. I guess I 
was hoping someone had found the silver bullet or some undocumented 
Asterisk feature that solved the issue. Back to the drawing board.

Mike Clark

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Re: [asterisk-users] TDM400P (TDM22P) and aux power.

2007-09-12 Thread Joe Acquisto



 On 9/5/2007 at 10:56 AM, Jason Parker [EMAIL PROTECTED] wrote:
 Joe Acquisto wrote:
 I need to ask, to refresh, is the aux power connector on the TDM400P card 
 *only* to power the ringer on any 
 analog phones/devices on the system?  
 
 Can I still use this board, to terminate POTS lines and use all SIP 
 Phones?
 
 Due to circumstances, I end up with a 1u server that has no aux power 
 connectors available.  I have to use this server, so am considering 
 abandoning the analog phones and using all SIP.  
 
 IIRC, the aux power *is* only to power ringers.
 
 joe a.
 
 
 Correct, it is to provide the ringing voltage on the FXS modules.  For 
 systems
 without internal molex connectors available, there is another option.  
 Digium
 has created an externally powered supply that can be used with these cards.
 
 http://www.digium.com/en/products/hardware/analogpwr.php 

Thanks.  I don't know how I missed this when posted, but, better late than 
never.

joe a. 


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Re: [asterisk-users] Linux-HA and Asterisk

2007-09-12 Thread Eric ManxPower Wieling

Polycoms work just fine behind NAT.

Mike Clark wrote:
 Chris Mason (Lists) wrote:
 Mike Clark wrote:

   
 Yes, the Asterisk boxes were on private addresses. The Polycoms are also 
 behind a NAT. Yes, I tried using externip in sip.conf and this allowed 
 registration, and calls to be placed, but no audio. Unfortunately, 
 Polycom does not support STUN.
 
 Your problem is not Linux-HA, it looks like that is fully functional.
 Your problem is the same one many people come across. You can't put
 Polycom phones behind NAT, it won't work.
 If you have to have the phones behind NAT, which I advise against, use
 Linksys which probably work better, and use a SIP aware NAT device.
 Better still, put the phones on the same network as the Asterisk PBX and
 say goodbye to your problems.

   
 Thanks Chris. Unfortunately, these solutions aren't an option. I guess I 
 was hoping someone had found the silver bullet or some undocumented 
 Asterisk feature that solved the issue. Back to the drawing board.

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[asterisk-users] fax and answer machine detection for outgoing call on DIVA card

2007-09-12 Thread lemmel lemmel


   Hello,
I need to detect both fax and answer machine, and it should be valuable that 
the detection will be run by the Diva card itself. So :
- I read Diva Documentation, and I found that the Diva could send some 
specific DTMF, if I had [..] enabled [this functionnality] by the 
application for a designated controller through a manufacturer request 
command 9 [...], but I didn't figure how to activate it ; has someone an 
idea ?
- I read my capi.conf, and found the faxdetect parameter, which enable 
faxdetection and redirection to EXTEN fax for incoming and/or outgoing 
calls, but I didn't succeed to perform that[1] ; has someone an idea about 
it ?

[1] I altered my capi.conf file, and put the fax entension just after the 
Dial, and called a fax machine, but nothing happens. The capi log is in the 
attachment.

_
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Re: [asterisk-users] Linux-HA and Asterisk

2007-09-12 Thread Dovid B

Eric,
Try 5 polycoms behind the same NAT router. Let me know when you grab a drink 
;)

- Original Message - 
From: Eric ManxPower Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 2:43 PM
Subject: Re: [asterisk-users] Linux-HA and Asterisk

 Polycoms work just fine behind NAT.

 Mike Clark wrote:
 Chris Mason (Lists) wrote:
 Mike Clark wrote:

 Yes, the Asterisk boxes were on private addresses. The Polycoms are 
 also
 behind a NAT. Yes, I tried using externip in sip.conf and this allowed
 registration, and calls to be placed, but no audio. Unfortunately,
 Polycom does not support STUN.

 Your problem is not Linux-HA, it looks like that is fully functional.
 Your problem is the same one many people come across. You can't put
 Polycom phones behind NAT, it won't work.
 If you have to have the phones behind NAT, which I advise against, use
 Linksys which probably work better, and use a SIP aware NAT device.
 Better still, put the phones on the same network as the Asterisk PBX and
 say goodbye to your problems.

 Thanks Chris. Unfortunately, these solutions aren't an option. I guess I
 was hoping someone had found the silver bullet or some undocumented
 Asterisk feature that solved the issue. Back to the drawing board.

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Re: [asterisk-users] SIP Debugging to separate log file

2007-09-12 Thread Dovid B

- Original Message - 
From: Jason Martin [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, September 06, 2007 4:58 PM
Subject: [asterisk-users] SIP Debugging to separate log file

 Hello, I'm working with our SIP provider to nail down some call quality 
 issues
 we're having, and they've asked me to provide SIP debug log files from our
 asterisk server. Is there a way to make asterisk 1.4 output only SIP
 debugging to a specific log file? Or it is best just to use tcpdump?

 Thank you!
 -- 

Try using Ngrep

ngrep -t -W byline -d any -w SIP ID port 5060

Where SIP ID is the id of your sip account. It should give you everything 
you need. 

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Re: [asterisk-users] Chan_sip Entry

2007-09-12 Thread Kutman.DK

Hello,
 
Yes, I also believe that this is some sort of codec issue.  Here is my sip.conf 
file:
 
[201]?xml:namespace prefix = o ns = urn:schemas-microsoft-com:office:office 
/

type=friend

;secret=201

record_out=Adhoc

record_in=Adhoc

qualify=no

port=5060

nat=no

host=dynamic

dtmfmode=rfc2833

dial=SIP/201

context=from-internal

canreinvite=no

callerid=device 201

 

[202]

type=friend

;secret=202

record_out=Adhoc

record_in=Adhoc

qualify=no

port=5060

nat=no

host=dynamic

dtmfmode=rfc2833

dial=SIP/202

context=from-internal

canreinvite=no

callerid=device 202
 
Note: The secret is commented out so that there is no authentication when 
registering with the Jain-Sip phones.
 
Thanks,
 
 

-Original Message-
From: Gerald A [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 11, 2007 5:12 PM
To: Kutman [EMAIL PROTECTED](Mat) DAEPM(RCS)@Ottawa-Hull
Subject: Re: [asterisk-users] Chan_sip Entry


Hi again,


On 9/11/07, [EMAIL PROTECTED]  [EMAIL PROTECTED]  mailto:[EMAIL PROTECTED]  
wrote: 


I am trying to get to Jain Sip softphones to call one another via an Asterisk 
server.  When I call from phone 1 to phone 2 there is audio transmission both 
ways, but when I call from phone 2 to phone 1 I don't get audio transmission 
and reception both ways.  When I look at the asterisk log file it has an entry 
which says: 
Oooh, format changed to 2.


Usually this is a codec selection problem. Are both Jain's the same version?

Maybe posting your sip.conf for the phones might help.

Thanks, 
Gerald.
 

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Re: [asterisk-users] Flash IDE

2007-09-12 Thread Juan Sandro


 You could read the archives from a week or 2 ago under the heading: Build 
 your own appliance
 
Yap... read it, thanks
  I use these deices, but I unload them entirely into RAM.
 
Fine.. I though about that too but what about:
 
- if power fails?
- how/when to write changes to DOM?
 If you're sticking a normal disctibution on it, I'd suggest dumping the  
 DOM and getting a laptop type IDE/SATA drive and using that instead. It's  
 not silent, but will be very quiet.
 
Yeah, the trouble is that device we bough do not have space for HDs.. :)
 
 
_
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Re: [asterisk-users] Linux-HA and Asterisk

2007-09-12 Thread Mike Clark

Eric ManxPower Wieling wrote:
 Polycoms work just fine behind NAT.
   
Yep, we have lots of Polycoms behind NAT working fine with Asterisk 
servers on *public* IPs. However, with the HA cluster, we had the 
Asterisk servers NATed in a Linux-HA cluster and in that configuration, 
the Asterisk servers seem to only send RTP packets to the phones private 
address.
 Mike Clark wrote:
   
 Chris Mason (Lists) wrote:
 
 Mike Clark wrote:

   
   
 Yes, the Asterisk boxes were on private addresses. The Polycoms are also 
 behind a NAT. Yes, I tried using externip in sip.conf and this allowed 
 registration, and calls to be placed, but no audio. Unfortunately, 
 Polycom does not support STUN.
 
 
 Your problem is not Linux-HA, it looks like that is fully functional.
 Your problem is the same one many people come across. You can't put
 Polycom phones behind NAT, it won't work.
 If you have to have the phones behind NAT, which I advise against, use
 Linksys which probably work better, and use a SIP aware NAT device.
 Better still, put the phones on the same network as the Asterisk PBX and
 say goodbye to your problems.

   
   
 Thanks Chris. Unfortunately, these solutions aren't an option. I guess I 
 was hoping someone had found the silver bullet or some undocumented 
 Asterisk feature that solved the issue. Back to the drawing board.
 

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Re: [asterisk-users] Flash IDE

2007-09-12 Thread Bill Seddon

Does it have to be a flash device?  I have an 8GB flash drive that is
really a small hard disk that plugs into and is powered by a USB port.
The device is 3cmx3cmx0.5cm, silent, fast and wears out like a hard disk
not flash memory.  It doesn't stick out (and so get knocked off) because
the USB connecter is hinged.  Something like this might be better for
your (though more expensive).

 

On the other hand, flash devices able to accommodate the needs of
Asterisk cost almost nothing.  Why not use two devices one that is
updated in real-time and that is backed up periodically (say overnight).
If one begins to fail then you can switch over the periodic back up and
used a new device for backup.

Bill 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan
Sandro
Sent: 12 September 2007 13:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Flash IDE

 


 You could read the archives from a week or 2 ago under the heading:
 Build your own appliance
 
Yap... read it, thanks

 
 I use these deices, but I unload them entirely into RAM.
 
Fine.. I though about that too but what about:
 
- if power fails?
- how/when to write changes to DOM?


 If you're sticking a normal disctibution on it, I'd suggest dumping
the 
 DOM and getting a laptop type IDE/SATA drive and using that instead.
It's 
 not silent, but will be very quiet.
 
Yeah, the trouble is that device we bough do not have space for HDs.. :)
 

 



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Re: [asterisk-users] Flash IDE

2007-09-12 Thread Juan Sandro


 So basically it seems that given a large enough flash drive with decent  
 wear levelling the lifetime should be completely ample...  ...Thats the 
 theory anyway.  I feel quite bullish about the whole thing, but I think I 
 would avoid  the *really* discounted cheapo flash drives since they may not 
 have the  correct wear levelling. Decent brand names should be fine though 
 (and  you can google for details on their specs)
 
 
Hi
 
Yeah we contacted a distributor of PQI flash memory. They sent us a wear 
leveling formula.
 
Here it is:
 
Example: A 256MB flash device writing 128KB data into flash device the formula 
as below: ( With wear leveling ).
DOM Lifetime ( Theory ) = (256MB-100MB)*100K*0.95 / (128KB/sec) *60*60*24= 
1340.06 days
# “0.95” : After the flash being format the capacity might lower than the 
certain capacity.# “60*60*24” : 86400 writing times per day. 1 time /sec# 
”100MB” : The size of your OS  AP. In this case we set it as 
“100MB”#“128KB/sec”: The data size that writing onto flash device per second.# 
”100K” : The limitation of flash memory’s P/E cycles.
 
P.S. : NAND type flash (Small block) : 1 Block = 32 page * 512byte = 16KBNAND 
type flash (Large block): 1 Block = 64 page * 2Kbyte = 128KBIf the data less 
than 128KB we recommend you still calculate it with 128KB.
 
 
NOW.. I am truly confused :)
 
 
Juan
 
 
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Re: [asterisk-users] Linux-HA and Asterisk

2007-09-12 Thread Jerry Jones

How about 20+ on a Qwest DSL modem hitting our server? Works great.


On Sep 12, 2007, at 7:23 AM, Dovid B wrote:

 Eric,
 Try 5 polycoms behind the same NAT router. Let me know when you  
 grab a drink
 ;)

 - Original Message -
 From: Eric ManxPower Wieling [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Wednesday, September 12, 2007 2:43 PM
 Subject: Re: [asterisk-users] Linux-HA and Asterisk


 Polycoms work just fine behind NAT.

 Mike Clark wrote:
 Chris Mason (Lists) wrote:
 Mike Clark wrote:


 Yes, the Asterisk boxes were on private addresses. The Polycoms  
 are
 also
 behind a NAT. Yes, I tried using externip in sip.conf and this  
 allowed
 registration, and calls to be placed, but no audio. Unfortunately,
 Polycom does not support STUN.

 Your problem is not Linux-HA, it looks like that is fully  
 functional.
 Your problem is the same one many people come across. You can't put
 Polycom phones behind NAT, it won't work.
 If you have to have the phones behind NAT, which I advise  
 against, use
 Linksys which probably work better, and use a SIP aware NAT device.
 Better still, put the phones on the same network as the Asterisk  
 PBX and
 say goodbye to your problems.


 Thanks Chris. Unfortunately, these solutions aren't an option. I  
 guess I
 was hoping someone had found the silver bullet or some undocumented
 Asterisk feature that solved the issue. Back to the drawing board.

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Re: [asterisk-users] Flash IDE

2007-09-12 Thread Jon Pounder


there is tons of information about linux and flash drives on the  
nslu2-linux.org and the openwrt sites.

main points :

- disable swap
- disable atime
- disable most logging

once the drive is not being written to then it will last a long time.




Quoting Bill Seddon [EMAIL PROTECTED]:

 Does it have to be a flash device?  I have an 8GB flash drive that is
 really a small hard disk that plugs into and is powered by a USB port.
 The device is 3cmx3cmx0.5cm, silent, fast and wears out like a hard disk
 not flash memory.  It doesn't stick out (and so get knocked off) because
 the USB connecter is hinged.  Something like this might be better for
 your (though more expensive).



 On the other hand, flash devices able to accommodate the needs of
 Asterisk cost almost nothing.  Why not use two devices one that is
 updated in real-time and that is backed up periodically (say overnight).
 If one begins to fail then you can switch over the periodic back up and
 used a new device for backup.

 Bill



 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Juan
 Sandro
 Sent: 12 September 2007 13:46
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Flash IDE




 You could read the archives from a week or 2 ago under the heading:
 Build your own appliance

 Yap... read it, thanks


 I use these deices, but I unload them entirely into RAM.

 Fine.. I though about that too but what about:

 - if power fails?
 - how/when to write changes to DOM?


 If you're sticking a normal disctibution on it, I'd suggest dumping
 the
 DOM and getting a laptop type IDE/SATA drive and using that instead.
 It's
 not silent, but will be very quiet.

 Yeah, the trouble is that device we bough do not have space for HDs.. :)




 

 Invite your mail contacts to join your friends list with Windows Live
 Spaces. It's easy! Try it!
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 aspxmkt=en-us





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[asterisk-users] Problems with Asterisk behind a firewall

2007-09-12 Thread Christian

Hi all,
I have set up Asterisk and I am able to register with my SIP provider and 
receive calls.
When I try to register with Asterisk from outside I can place calls but tthe 
other person can't hear me.
Have opened port 5060 UDP as well as port 1 to 2 UDP. Any ideas?
Thanks,
Christian

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Re: [asterisk-users] Linux-HA and Asterisk

2007-09-12 Thread Jon Pounder

Quoting Jerry Jones [EMAIL PROTECTED]:

 How about 20+ on a Qwest DSL modem hitting our server? Works great.

yeah but how many have call paths open at once ? just sitting there on  
hook you could probably have hundreds and still be fine. and of the  
call paths open how many reinvited and are talking over the lan  
directly to each other ?







 On Sep 12, 2007, at 7:23 AM, Dovid B wrote:

 Eric,
 Try 5 polycoms behind the same NAT router. Let me know when you
 grab a drink
 ;)

 - Original Message -
 From: Eric ManxPower Wieling [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Wednesday, September 12, 2007 2:43 PM
 Subject: Re: [asterisk-users] Linux-HA and Asterisk


 Polycoms work just fine behind NAT.

 Mike Clark wrote:
 Chris Mason (Lists) wrote:
 Mike Clark wrote:


 Yes, the Asterisk boxes were on private addresses. The Polycoms
 are
 also
 behind a NAT. Yes, I tried using externip in sip.conf and this
 allowed
 registration, and calls to be placed, but no audio. Unfortunately,
 Polycom does not support STUN.

 Your problem is not Linux-HA, it looks like that is fully
 functional.
 Your problem is the same one many people come across. You can't put
 Polycom phones behind NAT, it won't work.
 If you have to have the phones behind NAT, which I advise
 against, use
 Linksys which probably work better, and use a SIP aware NAT device.
 Better still, put the phones on the same network as the Asterisk
 PBX and
 say goodbye to your problems.


 Thanks Chris. Unfortunately, these solutions aren't an option. I
 guess I
 was hoping someone had found the silver bullet or some undocumented
 Asterisk feature that solved the issue. Back to the drawing board.

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 _/_/_/  _/  _/ _/_/_/  _/  _/_/
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Re: [asterisk-users] Flash IDE

2007-09-12 Thread Juan Sandro


 I've had CF units fail in service, but it's true that reliability is 
 increasing, especially as they get bigger.  I would recommend going with 
 the largest CF you can afford.  -Stephen-
 
Thanks Stephen...
 
 
That makes sence now if wear leveling is used.
 
 
Juan
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Re: [asterisk-users] Linux-HA and Asterisk

2007-09-12 Thread Eric ManxPower Wieling

Now why would that cause a problem if it is a decent NAT router?

What specific issue did you have?  Obviously NAT increases the 
complexity of an Asterisk and phone deployment, but it does work.

Dovid B wrote:
 Eric,
 Try 5 polycoms behind the same NAT router. Let me know when you grab a drink 
 ;)
 
 - Original Message - 
 From: Eric ManxPower Wieling [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, September 12, 2007 2:43 PM
 Subject: Re: [asterisk-users] Linux-HA and Asterisk
 
 
 Polycoms work just fine behind NAT.

 Mike Clark wrote:
 Chris Mason (Lists) wrote:
 Mike Clark wrote:


 Yes, the Asterisk boxes were on private addresses. The Polycoms are 
 also
 behind a NAT. Yes, I tried using externip in sip.conf and this allowed
 registration, and calls to be placed, but no audio. Unfortunately,
 Polycom does not support STUN.

 Your problem is not Linux-HA, it looks like that is fully functional.
 Your problem is the same one many people come across. You can't put
 Polycom phones behind NAT, it won't work.
 If you have to have the phones behind NAT, which I advise against, use
 Linksys which probably work better, and use a SIP aware NAT device.
 Better still, put the phones on the same network as the Asterisk PBX and
 say goodbye to your problems.


 Thanks Chris. Unfortunately, these solutions aren't an option. I guess I
 was hoping someone had found the silver bullet or some undocumented
 Asterisk feature that solved the issue. Back to the drawing board.
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Re: [asterisk-users] Linux-HA and Asterisk

2007-09-12 Thread Eric ManxPower Wieling

THAT is an issue with externip= and localnet= not being correct.

Mike Clark wrote:
 Eric ManxPower Wieling wrote:
 Polycoms work just fine behind NAT.
   
 Yep, we have lots of Polycoms behind NAT working fine with Asterisk 
 servers on *public* IPs. However, with the HA cluster, we had the 
 Asterisk servers NATed in a Linux-HA cluster and in that configuration, 
 the Asterisk servers seem to only send RTP packets to the phones private 
 address.
 Mike Clark wrote:
   
 Chris Mason (Lists) wrote:
 
 Mike Clark wrote:

   
   
 Yes, the Asterisk boxes were on private addresses. The Polycoms are also 
 behind a NAT. Yes, I tried using externip in sip.conf and this allowed 
 registration, and calls to be placed, but no audio. Unfortunately, 
 Polycom does not support STUN.
 
 
 Your problem is not Linux-HA, it looks like that is fully functional.
 Your problem is the same one many people come across. You can't put
 Polycom phones behind NAT, it won't work.
 If you have to have the phones behind NAT, which I advise against, use
 Linksys which probably work better, and use a SIP aware NAT device.
 Better still, put the phones on the same network as the Asterisk PBX and
 say goodbye to your problems.

   
   
 Thanks Chris. Unfortunately, these solutions aren't an option. I guess I 
 was hoping someone had found the silver bullet or some undocumented 
 Asterisk feature that solved the issue. Back to the drawing board.
 
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Re: [asterisk-users] Different Networks

2007-09-12 Thread Mike Hammett

*bump*


-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com


- Original Message - 
From: Mike Hammett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Friday, September 07, 2007 3:25 PM
Subject: Re: [asterisk-users] Different Networks


 If it has nothing to do with Asterisk, then why does every other device 
 work
 as its supposed to?

 An MGCP ATA routes out that interface.
 A laptop routes out that interface.
 That server traceroutes out that interface.

 Asterisk doesn't link up.


 -
 Mike Hammett
 Intelligent Computing Solutions
 http://www.ics-il.com


 - Original Message - 
 From: Erik Anderson [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Friday, September 07, 2007 3:06 PM
 Subject: Re: [asterisk-users] Different Networks


 On 9/6/07, Mike Hammett [EMAIL PROTECTED] wrote:

 I have multiple upstreams in my office.  The primary upstream is having
 some
 issues with latency\jitter.  I want to move the VoIP traffic to another
 interface.

 I have the router set to send all traffic destined for local networks
 out
 the respective interfaces.  Traffic destined to the Internet goes out 
 one
 of
 the upstreams.

 I can do this on a per-IP basis and have successfully done so in testing
 on
 my laptop and a couple other machines.  I also have it in production for
 an
 ATA.

 I also switch all devices to use another upstream with the failure of 
 the
 primary ISP.

 Again, this works with everything but the Asterisk server.

 The internal Asterisk server cannot connect to the Asterisk server out 
 on
 the public Internet.  How do I investigate this?

 Mike - there's no reason this routing problem would have anything to
 do with asterisk itself.Have you tried running links (or another
 text web browser) on the asterisk server to see if you're able to get
 traffic past the gateway?  Do you have the default gateway and/or
 routing tables configured correctly on the asterisk server?

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Re: [asterisk-users] Problems with Asterisk behind a firewall

2007-09-12 Thread lemmel lemmel

I don't have enough experiment to help you, but I can suggest you this :
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html

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Re: [asterisk-users] TDM2400P: Power alarm error on boot

2007-09-12 Thread gincantalupo

Hi all,
I solved the problem changing the module.

Giorgio


gincantalupo wrote:
 Hi,
 I have an Asterisk PBX equipped with (a Sangoma PRI card and) a Digium 
 TDM2400P.
 I got this error inside /var/log/messages:Power alarm on module 
 8, resetting!
 I rebooted the PBX and this time I got:Power alarm on module 
 7, resetting!

 Please, does anybody know what it means? can it be a TDM2400P broken module?

 Thank you.

 Giorgio.

   

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[asterisk-users] Wanted: VoIP Engineer for Warsaw !

2007-09-12 Thread laurent schweizer

Peoplefone AG offers Voice over IP(VoIP) services with exceptional rates.
Peoplefone is a certified partner of
Siemenshttp://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1sdc_sid=1113876080;and
AVM/FRITZ!Box http://www.fritz-shop.ch/. Due to our rapid growth, for our
new Polish subsidiary we are currently seeking for:





VOIP SPECIALIST

Place of work: Warsaw

*Requirements:*

   - Graduation from college or university with a Bachelor's degree
   (preferably IT)
   - Experience with PHP
   - Practical knowledge of C and C++
   - Practical knowledge of Mysql and Postgresql
   - Linux experience
   - Knowledge of IP Networks, UDP, TCP
   - Experience with tools for Network analysis like Ethereal
   - VoIP basic knowledge, VoIP servers, configuration of devices
   - Fluency in English
   - Ability to interact with individuals and groups at all levels
   - Detail oriented and analytical
   - Strong verbal and written communication skills



Knowledge of Perl, Java, Asterisk / SER / OPENSER, ability to configure
routers, Cisco or Patton gateways, knowledge of SIP and STUN protocol,
knowledge of NAT problems, of outbandProxy, knowledge of monitoring tools
like Cactus, Nagios, MRTG or high availability tools like DRBD, Hearthbeat
would be an additional asset.



Interested individuals are requested to send their resume to:

[EMAIL PROTECTED]


In your application please put the following statement:
I hereby agree for processing the following personal information strictly
for recruitment purposes in accordance with the regulation regarding the
protection of personal data passed on the following date: 29.08.97r. Dz.U nr
133 poz. 883.
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Re: [asterisk-users] Linux-HA and Asterisk

2007-09-12 Thread Ove Aursand

I am also using Polycom behind NAT without problems. But my asterisk box is not natted (so I have no externip setting in my asterisk).
My setup is Asterisk-internet-nat-polycom601/501/430/301

(PS: Hopefully posting in plain text after adding digium.com as a text-only domain in thunderbird :))

Regards,
Ove

Eric "ManxPower" Wieling wrote:

Polycoms work just fine behind NAT.

Mike Clark wrote:

Chris Mason (Lists) wrote:

Mike Clark wrote:

Yes, the Asterisk boxes were on private addresses. The Polycoms are also
behind a NAT. Yes, I tried using externip in sip.conf and this allowed
registration, and calls to be placed, but no audio. Unfortunately,
Polycom does not support STUN.

Your problem is not Linux-HA, it looks like that is fully functional.
Your problem is the same one many people come across. You can't put
Polycom phones behind NAT, it won't work.
If you have to have the phones behind NAT, which I advise against, use
Linksys which probably work better, and use a SIP aware NAT device.
Better still, put the phones on the same network as the Asterisk PBX and
say goodbye to your problems.

Thanks Chris. Unfortunately, these solutions aren't an option. I guess I
was hoping someone had found the silver bullet or some undocumented
Asterisk feature that solved the issue. Back to the drawing board.

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[asterisk-users] Direct dialing to correct extension from analog lines

2007-09-12 Thread Lars Bensmann

Hi,

I have a problem with people that are calling from analog lines.

We have a block of numbers 12345 - 0 to -99. Most calls are transmitting
the whole number including the extension. There's no problem with that.

But people calling from analog lines are connected to our asterisk box
as soon as they finish dialing 12345. They don't get a chance to dial an
extension.

Just inserting a WaitExten() does not help, as it does not get any
additional digits. I guess I would have to Answer() the call first, but I
really would like to avoid this as this might generate unnecessary costs
for the caller.

I used 'bri debug span', but I don't see any information coming in after
initiating the call to the base number.

I can set up an extension with the base number to dial a certain
extension, but then the extension the caller wants to dial is lost. If I
don't set up this extension the caller does not get through at all as
asterisk rejects the call. I tried setting autofallthrought=no, but it
didn't make a difference.

How do I tell asterisk to wait for the extension from the PSTN?

I'm using a Xorcom Astribank BRI 8-port with Asterisk
1.2.19-BRIstuffed-0.3.0-PRE-1y-e (from the Xorcom Debian repository).

Thanks in advance,
Lars

-- 
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[asterisk-users] Astribank 32 and Far End Disconnection

2007-09-12 Thread Gleidson Antonio Henriques

Hi all,

   I'm interested in buy a Astribank-32 but i never heard about detection of 
far-end disconnection.
   Does anyone have some experience about that functionality in this 
hardware ?
   Thanks in Advance,

Gleidson Antonio Henriques


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Re: [asterisk-users] Flash IDE

2007-09-12 Thread Gordon Henderson

On Wed, 12 Sep 2007, Juan Sandro wrote:


 You could read the archives from a week or 2 ago under the heading: Build 
 your own appliance

 Yap... read it, thanks
 I use these deices, but I unload them entirely into RAM.

 Fine.. I though about that too but what about:

 - if power fails?

*shrug*

What if the power fails? It reboots, reloads from flash and re-starts.

 - how/when to write changes to DOM?

What I do is have a separate partition on the device and write a TAR file 
of the files that I need to preserve between boots. No different from 
copy run start in a cisco.

I have a 2nd device that is mounted read/write for Voicemail. Less 
critical than the system  configuration device

 If you're sticking a normal disctibution on it, I'd suggest dumping 
 the  DOM and getting a laptop type IDE/SATA drive and using that 
 instead. It's  not silent, but will be very quiet.

 Yeah, the trouble is that device we bough do not have space for HDs.. :)

Make sure it doesn't over heat then...

Gordon

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Re: [asterisk-users] Flash IDE

2007-09-12 Thread Jon Pounder

Quoting Gordon Henderson [EMAIL PROTECTED]:

 On Wed, 12 Sep 2007, Juan Sandro wrote:


 You could read the archives from a week or 2 ago under the   
 heading: Build your own appliance

 Yap... read it, thanks
 I use these deices, but I unload them entirely into RAM.

 Fine.. I though about that too but what about:

 - if power fails?

 *shrug*

 What if the power fails? It reboots, reloads from flash and re-starts.

 - how/when to write changes to DOM?

 What I do is have a separate partition on the device and write a TAR file
 of the files that I need to preserve between boots. No different from
 copy run start in a cisco.

 I have a 2nd device that is mounted read/write for Voicemail. Less
 critical than the system  configuration device

 If you're sticking a normal disctibution on it, I'd suggest dumping
 the  DOM and getting a laptop type IDE/SATA drive and using that
 instead. It's  not silent, but will be very quiet.

 Yeah, the trouble is that device we bough do not have space for HDs.. :)

 Make sure it doesn't over heat then...

I think most of the commercial voicemail solutions use harddrives, so  
does stuff like tivo for that matter. I wouldn't put vm on flash, os  
and config yes.

  How quiet does this place need to be ? don't the users all have  
computers already ? is the a/c silent too ? Just in my personal office  
there is so much background hum it masks out external noise almost  
completely, but its not an objectionably loud level, ie no one I have  
been on the phone with has ever commented on the noise level and it  
does not interfere with conversations.  Server room - now thats  
another story completely.








 Gordon

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Jon Pounder

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Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

2007-09-12 Thread shadowym

Maybe his comments were taken out of context as they don't have the whole
interview posted.  Why is he talking about queue games,  Biologicall and
other extremely niche crap when there are huge holes in the basic offering
(SLA and SCA)?

From: Al lists [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, September 11, 2007 8:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

I liked the queue game concept!
although it could be cruel!

On 9/11/07, Steve Totaro [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]  wrote:

http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up

Seems the Adtran relationship goes way back...

Thanks,
Steve Totaro

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Re: [asterisk-users] Astribank 32 and Far End Disconnection

2007-09-12 Thread Tzafrir Cohen

On Wed, Sep 12, 2007 at 11:59:41AM -0300, Gleidson Antonio Henriques wrote:
 Hi all,
 
I'm interested in buy a Astribank-32 but i never heard about detection of 
 far-end disconnection.
Does anyone have some experience about that functionality in this 
 hardware ?
Thanks in Advance,

Far end disconnection supervision. big words for a functionality you
would actually expect of an FXO adapter, I believe - to report to the
software driving it when nothing is connected to the port and hence no
use trying to make calls through that specific port.

Battery for the FXO means that it connected to an FXS on the oter side
(and hence gets power through the wire). The Astribank FXO module will
report the channel as in alarm to Asterisk and hence you will not be
able to make calls to it when nothing is connected there to take your
calls.

This means that you can configure asterisk to dial through Zap/g0 and
it would only call through FXO ports in group 0 which are actually
plugged. No need to reconfigure your PBX (and restarting Asterisk, or
changing your dialplan) just because you added / slightly-hanged an
analog trunk.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Direct dialing to correct extension from analog lines

2007-09-12 Thread Tzafrir Cohen

On Wed, Sep 12, 2007 at 04:55:39PM +0200, Lars Bensmann wrote:
 Hi,
 
 I have a problem with people that are calling from analog lines.
 
 We have a block of numbers 12345 - 0 to -99. 

00 to 99, right?

 Most calls are transmitting
 the whole number including the extension. There's no problem with that.
 
 But people calling from analog lines are connected to our asterisk box
 as soon as they finish dialing 12345. They don't get a chance to dial an
 extension.

That is what overlapdial is for, right?

But you need a unidirectional overlap dial?

 
 Just inserting a WaitExten() does not help, as it does not get any
 additional digits. I guess I would have to Answer() the call first, but I
 really would like to avoid this as this might generate unnecessary costs
 for the caller.
 
 I used 'bri debug span', but I don't see any information coming in after
 initiating the call to the base number.
 
 I can set up an extension with the base number to dial a certain
 extension, but then the extension the caller wants to dial is lost. If I
 don't set up this extension the caller does not get through at all as
 asterisk rejects the call. I tried setting autofallthrought=no, but it
 didn't make a difference.
 
 How do I tell asterisk to wait for the extension from the PSTN?
 
 I'm using a Xorcom Astribank BRI 8-port with Asterisk
 1.2.19-BRIstuffed-0.3.0-PRE-1y-e (from the Xorcom Debian repository).
 
 Thanks in advance,
 Lars

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Problems with Asterisk behind a firewall

2007-09-12 Thread Christian




On 2007-09-12 at 13:41 lemmel lemmel wrote:

I don't have enough experiment to help you, but I can suggest you this :
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
Many thanks, will have a look.
Christian

_
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http://eragon-heroic-fantasy.spaces.live.com/


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[asterisk-users] Solution: Sysmaster and Asterisk

2007-09-12 Thread Mani Nair

Hello Guys,

 

After adding money into my sysmaster phone account I am able to make calls
outside.thnx

 

  _  

From: Mani Nair [mailto:[EMAIL PROTECTED] 
Sent: Friday, September 07, 2007 9:16 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Sysmaster and Asterisk

 

Hello Guys,

 

I am unable to make calls to outside number from some of my extensions.
Internally I am able to make and receive calls between extensions and also I
am able to receive call from outside number. Any suggestions?

Then in am thinking of getting rid of Sysmaster and configure Trixbox to do
the entire job that currently my Sysmaster is doing. Any suggestions..?

 

Mani

 

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[asterisk-users] res_snmp

2007-09-12 Thread yonoko molomo

Hi,

I have problems compiling asterisk 1.4.11 with res_snmp.
I do 'make menuselect', and I see that this resource module depends on netsnmp.
I am using centOS 4.5.
I do:
 yum install net-snmp net-snmp-devel net-snmp-utils net-snmp-libs
I don't know if i am missing something.

I go to the source directory and I do:
./configure

but still does not work:
 ...
 checking for curses.h... (cached) yes
 checking for net-snmp-config... /usr/bin/net-snmp-config
 checking for snmp_register_callback in -lnetsnmp... no
 ...

When i run 'make menuselect' I cannot get rid of the XXX, and i can't
select res_snmp.

I also tried to install net-snmp-perl package but it does not help. i
have no clue how to continue.

any ideas?
thanks

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[asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

2007-09-12 Thread Euler Pereira

Hey all!

I'm newbie in the Asterisk World but old in other telephony systems like
Lucent/Avaya, Sopho, Siemens and Linux/Unix system.

I'm in doubt, as based system, should I install Fedora, Debian,
Slackware, FreeBSD our Sun Solaris? Which is more robust for a small
Asterisk system, about 8 extensions, 4 hardphone and 4 softphone?

Thanks in advance!
Euler
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Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

2007-09-12 Thread Atis

On 9/12/07, Al lists [EMAIL PROTECTED] wrote:
 I liked the queue game concept!
 although it could be cruel!

Hmm, can't find anything about queue games in google. Anybody have
some more details? I would like to have some functionality of that..

Regards,
Atis

-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org

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Re: [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

2007-09-12 Thread Alex Balashov


   As far as I know, Asterisk was developed on Linux, and although it will 
work on other systems, my best experiences have been with Linux.  This is
especially true once you start talking about integration with Zaptel, 
since it requires kernel module hooks, etc.  And I think your situation 
does, since you mention a hardphone.

   As far as what distribution to install, Linux is Linux is Linux.  Use 
whatever decisionmaking criteria you'd use in any other situation for
this type of system, Asterisk aside.  Personally, I run Debian and find
it the easiest to quickly install, turn up and manage if I'm deploying
an embedded system, but that's a claim that's likely to be disputed and
can lead to a religious war as much as any other question-by-proxy
concerning distributions.

   In the end, it's a matter of personal preference, mixed in with a few 
subtle business-level differentiators (Fedora's official, corporate 
support vs. Debian's organic, ad hoc, open-source nature, for example).

-- Alex

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

2007-09-12 Thread Gordon Henderson

On Wed, 12 Sep 2007, Euler Pereira wrote:

 Hey all!

I'm newbie in the Asterisk World but old in other telephony systems like
 Lucent/Avaya, Sopho, Siemens and Linux/Unix system.

I'm in doubt, as based system, should I install Fedora, Debian,
 Slackware, FreeBSD our Sun Solaris? Which is more robust for a small
 Asterisk system, about 8 extensions, 4 hardphone and 4 softphone?

Which of Fedora, Debian or Slackware do you know best?

I use Debian, but that's because it's the one I know best.

Gordon

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Re: [asterisk-users] Flash IDE

2007-09-12 Thread Mojo with Horan Company, LLC

Diego Iastrubni wrote:
  Disable DMA on that drive. Thee HD/DOM/CF-card does not support DMA 
and linux tries to DMA it.
This is with ide=nodma kernel option.  It's just loading the kernel and 
initrd at the beginning, before the kernel's actually booted.

  (gmail quoting stinks)
huh?  what's that mean?  putting the response after the quote? 

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Re: [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

2007-09-12 Thread Tzafrir Cohen

On Wed, Sep 12, 2007 at 05:25:16PM +0100, Gordon Henderson wrote:
 On Wed, 12 Sep 2007, Euler Pereira wrote:
 
  Hey all!
 
 I'm newbie in the Asterisk World but old in other telephony systems like
  Lucent/Avaya, Sopho, Siemens and Linux/Unix system.
 
 I'm in doubt, as based system, should I install Fedora, Debian,
  Slackware, FreeBSD our Sun Solaris? Which is more robust for a small
  Asterisk system, about 8 extensions, 4 hardphone and 4 softphone?
 
 Which of Fedora, Debian or Slackware do you know best?

I don't know Asterisk on FreeBSD well enough so I can't give a good
opinion on it.

I think Asterisk on Solaris is not mature enough. So unless you have a
different reason to use Solaris (too many simple compilation issues), or 
you're otherwise more familar with Solaris, you better avoid it at this 
stage.

Of the three you mentioned, I have some reservations with Fedora. Fedora
is a fast-moving Distribution. Somewhat a keen to Debian's Testing. For
about a year the distribution is maintained, but still mutates. And
hence has more chances of breaking. Then it stops recieving secirity
updates at all.

The one you have not mentioned is CentOS, which is probably worth
looking at instead of Fedora.

Debian is a very solid choice, IMHO. My personal preference.

Slackware has a different mentality: you decide what happens. You have
to apply more configurations and do more work. Maybe this also fits well
with the mentality of installing Asterisk from source. I personally have
no experince with it.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

2007-09-12 Thread Jaswinder Singh

I prefer centos , debian/ubuntu are also a good option . It just depends on
which distribution you are comfortable with . We also have asterisk running
very stable on slackware .

On 12/09/2007, Gordon Henderson [EMAIL PROTECTED] wrote:

 On Wed, 12 Sep 2007, Euler Pereira wrote:

  Hey all!
 
 I'm newbie in the Asterisk World but old in other telephony systems
 like
  Lucent/Avaya, Sopho, Siemens and Linux/Unix system.
 
 I'm in doubt, as based system, should I install Fedora, Debian,
  Slackware, FreeBSD our Sun Solaris? Which is more robust for a small
  Asterisk system, about 8 extensions, 4 hardphone and 4 softphone?

 Which of Fedora, Debian or Slackware do you know best?

 I use Debian, but that's because it's the one I know best.

 Gordon

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[asterisk-users] Digium Appliance

2007-09-12 Thread Matt

Hi,
Has anyone actually gotten their hands on an appliance yet?   If so, how
robust and working are they?  Any issues?
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Re: [asterisk-users] Installing Asterisk on to CentOS 4



Ove Aursand wrote:
 Anthony Francis wrote:
 Ove Aursand wrote:
   
 Abdul wrote:
 
 Hi expets,

 I have installed Asterisk 1.4.11 on CentOS4 successfully without any 
 error.
 But when i am trying to start asterisk with following cmd i am 
 getting unknown command.

 [EMAIL PROTECTED] ~]$ asterisk -vvc
 -bash: asterisk: command not found
 [EMAIL PROTECTED] ~]$

 I checked modules and other configuration files which are installed 
 correctly.

 Please help me to locate this problem.

 Thank You

   
 Try the command as root:
 [EMAIL PROTECTED] ~]$ su -
 *enter password*
 [EMAIL PROTECTED] ~]# asterisk -cvv

 Rgds,
 Ove
   
   
 
 
 Ove,

 You should seriously reconsider showing your IP in posts.

 You have open 22, 25, 53, 110, 111, 80, 143, 443, 3306

 MySQL open to the world? Seriously? Yikes

 Anthony


   
 Thanks for the heads up, but I just copied from Abdul's post. But I 
 guess he is reading this message too. Does everyone on this list 
 automatically nmap IP addresses from posts btw? :P

 Ove

I cannot speak for anyone else, but if I see them, and I am in the mood, 
hell yeah I port scan them.
Although not automatically, I can assure you that is what is happening 
daily to every IP address in the world anyway, LOL

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] online active call watching

2007-09-12 Thread Mojo with Horan Company, LLC

Dinesh Nair wrote:
 On Mon, 10 Sep 2007 13:43:46 -0800, Mojo with Horan  Company, LLC wrote:

   
 Though still in the proof-of-concept stage, my project AstSee from 
 http://www.astsee.com/ might be fun to play with if you're using 
 linux/XWindows.  There are screenshots there.
 

 that may be so, but without source, there's no way we can test it on
 freebsd. i'll stick with fop for the timebeing, thank you. 

   
Ok, so source is available now.  Do your worst:  Innovate!


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Re: [asterisk-users] Generating an old-fashioned dialtone

2007-09-12 Thread Stephen Bosch

Phil Reynolds wrote:
 Quoting Clayton Milos [EMAIL PROTECTED]:
 Is there a way to generate an old-fashioned dial tone with Asterisk?

 I'm thinking of one that sounds like:

 http://www.seg.co.uk/telecomm/dial tone.wav
 
 As far as I know dialtone with SIP can only be generated on the handsets.
 We're using Cisco 7960's with SIP firmware on them and they generate a
 dialtone.
 
 As far as I know I didn't mention generating it as a dialtone on a SIP  
 phone, merely generating the tone.
 
 I can probably put it on Zap phones easily enough if I wish, but I'd  
 need to know how to generate it first, and all I am after right now is  
 the sound.

It's been years since I was in the UK. I can't remember what the modern
dial tone sounds like. When did it change?

-Stephen-

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Re: [asterisk-users] TE405P intermittent yellow alarm

2007-09-12 Thread Steve Totaro

Richard van der Hoff wrote:
 Folks,

 I really hope you can help me here - I'm beginning to tear my hair out!

 About 10 days ago my company moved to a new office. As a result of this,
 we've plugged our PBX box, which has happily been running for the last
 three years, into our new E1 line. Since then, I've been seeing
 intermittent yellow alarms. Obviously, since this was working fine in
 the old office, the thing to suspect is the new line - but the telco
 (British Telecom) aren't really helping much.

 The box is a 2.4GHz Intel box, with a TE405P installed in it (we're only
 using one of the spans). I'm using the zaptel drivers version 1.4.4
 (I've also tried 1.0.2 with similar results). zaptel.conf has:

 span=1,1,0,ccs,hdb3,crc4
 bchan=1-15,17-31
 dchan=16

 Essentially it runs fine for a few hours; then zttool reports yellow
 alarm and calls can be made neither in nor out. After a while this
 clears itself again and all is well for another few hours.

 At this point, I'd really like to know what a yellow alarm actually
 means. I've read that it indicates that that the other end of the E1 is
 in an alarm condition: however BT's terminating unit seems quite happy
 with no alarm conditions at all.

 So, really hoping that someone can shed some light on what this might
 all mean.

 Cheers,

 Richard


   
Check your cabling.  Replace it with new stuff.  Re-punch everything. 

It is obviously somewhere in the line.  If the above does not fix it, 
maybe you can get a lucky and get a good tech out that will stick around 
to see the issue.

Thanks,
Steve

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[asterisk-users] Callback for unanswered transfers...

2007-09-12 Thread Luis Antonio Prata Barbosa

Hi,

Does anybody know if there is a way for a call goes back to transferer if
unanswered ?

Thanks

Luis A P Barbosa
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Re: [asterisk-users] Different Networks

2007-09-12 Thread Erik Anderson

On 9/7/07, Mike Hammett [EMAIL PROTECTED] wrote:
 If it has nothing to do with Asterisk, then why does every other device work
 as its supposed to?

You never answered as to whether or not you're able to get out past
your gateway with any other network applications on your asterisk
server.  Fire up [links/lynx] and pull up www.google.com.  Does it
work?

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Re: [asterisk-users] Callback for unanswered transfers...

2007-09-12 Thread Atis

On 9/12/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote:
 Hi,

 Does anybody know if there is a way for a call goes back to transferer if
 unanswered ?

Yes, before Dial to transferer set some variable that have he's
extension, and in your defined TRANSFER_CONTEXT, use Dial with g
option. After that, check DIALSTATUS!=ANSWERED and Dial back.

I have it working, if those details aren't enough, or something
doesn't go well, just ask.

Regards,
Atis
-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org

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Re: [asterisk-users] Generating an old-fashioned dialtone

2007-09-12 Thread Anthony Messina

On Wednesday 12 September 2007 02:57:18 am Phil Reynolds wrote:
 Is there a way to generate an old-fashioned dial tone with Asterisk?

 I'm thinking of one that sounds like:

 http://www.seg.co.uk/telecomm/dialtone.wav

see if you can find it here http://www.3amsystems.com/wireline/tone-search.htm

then try to put it in indications.conf.

-- 
Anthony -  http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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[asterisk-users] (no subject)

2007-09-12 Thread Niki Selken


Hello,

I am looking for an Asterisk consultant for occasional support on an  
asterisk phone system located in San Francisco. It would probably be  
primary remote support, but we may need some on site support  
occasionally. Please let me know if you are interested and available.


Thanks,

Niki Selken
Junior Systems Administrator
Colorful Expressions

[EMAIL PROTECTED]



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[asterisk-users] Looking for Asterisk Consultant in San Franicsco

2007-09-12 Thread Niki Selken


Hello,

I am looking for an Asterisk consultant for occasional support on an  
asterisk phone system located in San Francisco. It would probably be  
primary remote support, but we may need some on site support  
occasionally. Please let me know if you are interested and available.


Thanks,

Niki Selken
Junior Systems Administrator
Colorful Expressions

[EMAIL PROTECTED]



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[asterisk-users] Conference bridge.

2007-09-12 Thread Alex Balashov


Any recommendations for an affordable SIP conference bridge unit?  I mean 
one that isn't crappy;  something where the duplex and cancellation 
functions that are traditionally built into such devices actually work.

I am referring to something that looks like this . . .

http://www.hardware.com/products/cnet/I212272.jpg

But not necessarily that.

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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[asterisk-users] Agent Callback Login in 1.4

Awhile back I had heard some talk, in this list I believe that Agent 
callback login was going to be deprecated in 1.4, I see it is still 
there. Does anyone know what is happening with this?

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

I read the article, and it seems he was talking about cleaning the code 
and making it the best it could be, while the author was talking about 
the other things.

Anthony

shadowym wrote:

 Maybe his comments were taken out of context as they don’t have the 
 whole interview posted. Why is he talking about queue games, 
 Biologicall and other extremely niche crap when there are huge holes 
 in the basic offering (SLA and SCA)?

 *From:* Al lists [mailto:[EMAIL PROTECTED]
 *Sent:* Tuesday, September 11, 2007 8:28 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Mark Spencer: Digium is Growing Up 
 (VONMAG)

 I liked the queue game concept!
 although it could be cruel!

 On 9/11/07, *Steve Totaro* [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:

 http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up

 Seems the Adtran relationship goes way back...

 Thanks,
 Steve Totaro



-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] Digium Appliance

2007-09-12 Thread David Boyd

Hi Mat,
i have been working with the aa50 for a couple of weeks now.  They are
slick looking devices that still have a few bugs. I tried to use the
device like an end user without previous knowledge of Asterisk or the
asteriskGUI, and can say right off that a typical person will not be
able to use the device by gui only. The interface does not create all
entries required to configure either outbound routing or DID, outbound
caller id for either sip or IAX looks to the fullname field in the
users.conf file rather than CID entry They are working to correct
the issues, however as of yet no known release date for firmware fixes.
Having said that if you want to edit files via the gui by hand and make
appropriate changes then the device seems to work ok.  Did have an issue
where after reboot the system would register an IAX trunk with the
provider but outbound calls would fail until you kicked the system to
force a new registration.  A couple of times changes that were saved at
the home page failed to commit to the flash card, replaced the flash and
have not seen that issue again, but Little things that make me oogee
about putting into a customer location right now.

db

On Wed, 2007-09-12 at 12:52 -0400, Matt wrote:
 Hi,
 Has anyone actually gotten their hands on an appliance yet?   If so,
 how robust and working are they?  Any issues?
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Re: [asterisk-users] Generating an old-fashioned dialtone

On Wed, Sep 12, 2007 at 11:23:51AM -0600, Stephen Bosch wrote:
 It's been years since I was in the UK. I can't remember what the modern
 dial tone sounds like. When did it change?

The first version of it appeared in parts of Sutton Coldfield in 1976,
but some places still had the old tone into the 1990s. The modern one is
of a slightly higher pitch than the 1976 version. Much of Europe uses a
similar tone. The secondary dial tone in France (that followed use of
19 when that was the International prefix) was quite similar too.

The 1976 version of the tone came to Hednesford in 1985 - we had only
lived there for a few months and my mother thought something had gone
wrong.

-- 
Phil Reynolds
 o   mail: [EMAIL PROTECTED]
|L_ \  / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/  Waltham 66, Emley Moor 69, Droitwich 79, Windows 95

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Re: [asterisk-users] Digium Appliance

2007-09-12 Thread Steve Totaro

Sounds robust to me.

David Boyd wrote:
 Hi Mat,
 i have been working with the aa50 for a couple of weeks now.  They are
 slick looking devices that still have a few bugs. I tried to use the
 device like an end user without previous knowledge of Asterisk or the
 asteriskGUI, and can say right off that a typical person will not be
 able to use the device by gui only. The interface does not create all
 entries required to configure either outbound routing or DID, outbound
 caller id for either sip or IAX looks to the fullname field in the
 users.conf file rather than CID entry They are working to correct
 the issues, however as of yet no known release date for firmware fixes.
 Having said that if you want to edit files via the gui by hand and make
 appropriate changes then the device seems to work ok.  Did have an issue
 where after reboot the system would register an IAX trunk with the
 provider but outbound calls would fail until you kicked the system to
 force a new registration.  A couple of times changes that were saved at
 the home page failed to commit to the flash card, replaced the flash and
 have not seen that issue again, but Little things that make me oogee
 about putting into a customer location right now.

 db

 On Wed, 2007-09-12 at 12:52 -0400, Matt wrote:
   
 Hi,
 Has anyone actually gotten their hands on an appliance yet?   If so,
 how robust and working are they?  Any issues?
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Re: [asterisk-users] Generating an old-fashioned dialtone

On Wed, Sep 12, 2007 at 02:29:35PM -0500, Anthony Messina wrote:
 On Wednesday 12 September 2007 02:57:18 am Phil Reynolds wrote:
  Is there a way to generate an old-fashioned dial tone with Asterisk?
 
  I'm thinking of one that sounds like:
 
  http://www.seg.co.uk/telecomm/dialtone.wav
 
 see if you can find it here http://www.3amsystems.com/wireline/tone-search.htm
 
 then try to put it in indications.conf.

Hmmm... Dominican Republic's 33/16,0/16 might be a starting point for
further tuning

-- 
Phil Reynolds
 o   mail: [EMAIL PROTECTED]
|L_ \  / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/  Waltham 66, Emley Moor 69, Droitwich 79, Windows 95

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Re: [asterisk-users] Flash IDE

2007-09-12 Thread Hans Witvliet

On Wed, 2007-09-12 at 09:19 -0400, Jon Pounder wrote:
 there is tons of information about linux and flash drives on the  
 nslu2-linux.org and the openwrt sites.
 
 main points :
 
 - disable swap
 - disable atime
 - disable most logging
 
 once the drive is not being written to then it will last a long time.

I would also recommend to do all the (/tmp, /var/tmp, ...) writings to
ramdisk,
and send all the logging to a syslog server instead of local-log-files.

-- 
pgp-id: 926EBB12
pgp-fingerprint: BE97 1CBF FAC4 236C 4A73  F76E EDFC D032 926E BB12
Registered linux user: 75761 (http://counter.li.org)

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[asterisk-users] AsteriskNOW

2007-09-12 Thread Seysan

Hello,

is there any User Interface available in Asterisk NOW?

in Trixbox, As far as I know there is ARI, but does Asterisk Now has
anything for the Extension owners, I mean User portal not Admin portal?


Regards,


Seysan
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[asterisk-users] Assistance needed.

2007-09-12 Thread Tim King

I am looking for help from someone familiar with using asterisk and openser
to build a rather large VOIP network. I have 6 servers in place each with
their own purpose. I will give a brief summary and hopefully someone out
there is able to help be finalize this dialplan. I have six servers in
place. One has 4 port T1 to handle trunks to PSTN. One has 2 port T1 and is
to be used for SS7 links. Third is to be used as primary sip machine running
openser. 4th is voicemail server using asterisk. 5th is asterisk machine to
be used for VPBX customers. 6th is to hold all mysql databases with
configuration from all servers to providing a single point of provisioning.
If anyone out here thinks they can help please advise.


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Re: [asterisk-users] Generating an old-fashioned dialtone

2007-09-12 Thread Jay R. Ashworth

On Wed, Sep 12, 2007 at 10:13:03AM +0100, Phil Reynolds wrote:
 I can probably put it on Zap phones easily enough if I wish, but I'd  
 need to know how to generate it first, and all I am after right now is  
 the sound.

I believe that's roughly 250hz beating with 10hz.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

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[asterisk-users] Problems with two trunks

2007-09-12 Thread Joshua Small

Hi,

 

I am attempting to setup an asterisk server, current specs:

CentOS release 5 (Final)

Asterisk 1.4.11

Asterisk-gui checked out from SVN last week

 

I started with a fairly basic setup involving one VOIP provider who
provided one dial in number, and a couple of handsets. Config files are
below. It was pretty much totally built by Asterisk-gui, except for the
fact I had to add insecure=very into users.conf in order to stop the
dialin from our provider presenting an authentication error. Advice on
any more correct approach would be appreciated, but is not the focus of
this post:

 

Users.conf

;several handsets setup like this...

[6001]

callwaiting = yes

context = numberplan-custom-1

email = [EMAIL PROTECTED]

fullname = Joshua Small

hasagent = yes

hasdirectory = yes

hasiax = no

hasmanager = no

hassip = yes

hasvoicemail = no

host = dynamic

mailbox = 6001

secret = X

threewaycalling = yes

registeriax = no

registersip = yes

canreinvite = no

nat = no

dtmfmode = rfc2833

vmsecret = 1234

 

;some PSTNS

[trunk_2]

callerid = asreceived

context = DID_trunk_2

group = 2

hasexten = no

hasiax = no

hassip = no

trunkname = Ports 1,2,3,4

trunkstyle = analog

zapchan = 1,2,3,4

 

;my IP trunk

[trunk_3]

allow = all

context = DID_trunk_3

dialformat = ${EXTEN:1}

hasexten = no

hasiax = no

hassip = yes

host = gw02.mytel.net.au

port = 5060

registeriax = no

registersip = yes

secret = 

trunkname = Custom - MyTel2

trunkstyle = customvoip

username = 

type = friend

nat = yes

 

;extensions.conf

[numberplan-custom-1]

plancomment = DialPlan1

include = default

include = parkedcalls

exten = _0X!,1,Macro(trunkdial,${trunk_3}/${EXTEN:1})

comment = _0X!,1,First,standard

;a failover to PSTN, not yet enabled

;exten = _0X!,2,Macro(trunkdial,${trunk_2}/0${EXTEN:1})

;comment = _0X!,1,First,standard

 

At this point, everything appears to work fine. We can make calls from
our several handsets using our voip link no problems.

We have two different accounts with our provider, the goal being certain
handsets will connect to this account and therefore be billed
separately. I haven't gotten as far as to add the extra handsets and set
an appropriate dialplan, all I did was add this to users.conf:

 

[trunk_extra]

allow = all

context = DID_trunk_3

dialformat = ${EXTEN:1}

hasexten = no

hasiax = no

hassip = yes

host = gw02.mytel.net.au

port = 5060

registeriax = no

registersip = yes

secret = 

trunkname = Custom - MyTel Two

trunkstyle = customvoip

username = XX

type = friend

nat = yes

 

From this point on, my existing handsets don't appear to be able to get
a line out. My console looks like this (from the first call out):

Connected to Asterisk 1.4.11 currently running on asterisk (pid = 31999)

-- Remote UNIX connection

Verbosity is at least 8

-- Executing [EMAIL PROTECTED]:1]
Macro(SIP/8001-b7d0bb20, trunkdial|SIP/trunk_3/0425298582) in new
stack

-- Executing [EMAIL PROTECTED]:1] Dial(SIP/8001-b7d0bb20,
SIP/trunk_3/0425298582) in new stack

-- Called trunk_3/0425298582

[Sep 13 20:42:16] WARNING[32011]: chan_sip.c:12016
handle_response_invite: Received response: Forbidden from 'Joshua
Small sip:[EMAIL PROTECTED];tag=as29bb274d'

-- SIP/trunk_3-097ac708 is circuit-busy

  == Everyone is busy/congested at this time (1:0/1/0)

-- Executing [EMAIL PROTECTED]:2] Goto(SIP/8001-b7d0bb20,
s-CONGESTION|1) in new stack

-- Goto (macro-trunkdial,s-CONGESTION,1)

-- Executing [EMAIL PROTECTED]:1]
NoOp(SIP/8001-b7d0bb20, ) in new stack

  == Auto fallthrough, channel 'SIP/8001-b7d0bb20' status is
'CONGESTION'

 

 

Any advice on why our trunk_3 becomes congested, just because
trunk_extra is set to exist, is appreciated.

Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 959 |
www.visinet.com.au http://www.visinet.com.au/  

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[asterisk-users] Zap channels: no sound with certain call paths

2007-09-12 Thread Christian Weeks

Hi,
A most peculiar and vexing problem for you all. I hope I have been
verbose enough without being a firehose ;)

The set up:
I have a channel bank, using the r1t1 rhino driver with a rhino T1 card
(the channel bank itself is a very legacy piece of equipment)- this
supplies FXS for all the house phones. Also, a Wildcard TDM400P, using
the wctdm module with 1 FXO module, this supplies FXO to the upstream
telco (a single line).

The problem:
Lately, and without any configuration changes, incoming calls that route
through the Wildcard (from the telco) to the channel bank (well, a phone
connected to the channel bank) have no voice in either direction.
Obviously, this is rather frustrating. The same configuration has worked
quite reliably for the past year or so, so I am reasonably confident
that the problem isn't directly configuration related (though I have,
since this started occuring tried various configs).

The version where this started to occur (intermittently) was
asterisk/zaptel in debian etch (the 1.2 branch). I have since upgraded
to zaptel/asterisk from debian sid (the 1.4 branch) and the problems
have gotten marginally worse.

Stuff I have tried:
1. Zap-Zap (calling one channel bank extn from another) works fine.
2. Zap-anywhere (calling out from CB to telco through wildcard, or to
SIP provider, or IAX provider) works fine.
3. telco-Zap (calling in from telco to CB line) fails: no voice.
4. SIP/IAX-Zap (calling in from a SIP client to CB line) works.

Diagnostics examined:
1. ztmonitor any line -v shows expected signals, from the asterisk
perspective. But e.g. in scenario 3 above, there is no received voice
from the zap line. Which is consistent with the dialled CB line not
being properly connected somehow.

Oddities noticed:
1. Sometimes, when picking up a CB line, there is no dialtone. Only
resolution has been to reset the computer.
2. There are several odd messages in the log files:
(/var/log/syslog)
[..snip..]
Sep 12 17:52:04 phone kernel: Got pulse digit 36 on R1T1/0/3??
(note: lots of these, at least one per CB line, whenever we restart or
reprobe the module)
[..snip..]
Sep 12 17:53:29 phone asterisk[2638]: rc_avpair_new: unknown attribute
1490026597
(lots of these too, there seems to be a correlation between these
messages and no voice routings)
(/var/log/asterisk/messages (I have verbosity up nice and high))
[Sep 12 20:35:20] WARNING[3174] chan_zap.c: Ring/Off-hook in strange
state 6 on channel 25
(I've had this since I set the environment up. No one seems to be able
to give a sane answer as to why).

Finally, here's an interesting oddity. I can get the voice to come up,
in certain circumstances, by doing the following:
1. Dial in from telco using cellphone.
2. Answer with CB Zap line. No voice.
3. Hang up the CB Zap line.
4. Re-open any Zap CB line, execute a dial that uses telco line.
5. The telco line picks up (to execute the dial); voice is now connected
to the still waiting original call.

Here's the log file:
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Starting simple switch
on 'Zap/25-1'
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing
[EMAIL PROTECTED]:1] Goto(Zap/25-1, incoming-home|s|1) in new stack
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Goto
(incoming-home,s,1)
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing
[EMAIL PROTECTED]:1] NoOp(Zap/25-1,  Number) in new stack
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing
[EMAIL PROTECTED]:2] Set(Zap/25-1, TRANSFER_CONTEXT=transfer) in new
stack
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing
[EMAIL PROTECTED]:3] GotoIfTime(Zap/25-1, 9:00-20:00|*|*|*?s-DAY|1)
in new stack
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Goto
(incoming-home,s-DAY,1)
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing
[EMAIL PROTECTED]:1] Dial(Zap/25-1,
Zap/1Zap/3Zap/2Zap/10Zap/5Zap/6SIP/cpw...)
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 1
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 3
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 2
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 10
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 5
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 6
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called me
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/1-1 is ringing
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/3-1 is ringing
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/2-1 is ringing
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/10-1 is ringing
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/5-1 is ringing
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/6-1 is ringing
[Sep 12 18:23:51] VERBOSE[3051] logger.c: -- SIP/me-081f3db0 is
ringing
[Sep 12 18:23:53] VERBOSE[3051] logger.c: -- Zap/10-1 answered
Zap/25-1
[Sep 12 18:23:53] VERBOSE[3051] logger.c: -- Hungup 'Zap/6-1'
[Sep 12 18:23:53] VERBOSE[3051] logger.c: -- Hungup 'Zap/5-1'
[Sep 12 18:23:53]

Re: [asterisk-users] Problems with two trunks


I would have usually used sip.conf or iax.conf - users.conf is not
something I know well

PaulH


On Thu, 2007-09-13 at 10:44 +1000, Joshua Small wrote:
 Hi,
 
  
 
 I am attempting to setup an asterisk server, current specs:
 
 CentOS release 5 (Final)
 
 Asterisk 1.4.11
 
 Asterisk-gui checked out from SVN last week
 
  
 
 I started with a fairly basic setup involving one VOIP provider who
 provided one dial in number, and a couple of handsets. Config files
 are below. It was pretty much totally built by Asterisk-gui, except
 for the fact I had to add “insecure=very” into users.conf in order to
 stop the dialin from our provider presenting an authentication error.
 Advice on any more correct approach would be appreciated, but is not
 the focus of this post:
 
  
 
 Users.conf
 
 ;several handsets setup like this...
 
 [6001]
 
 callwaiting = yes
 
 context = numberplan-custom-1
 
 email = [EMAIL PROTECTED]
 
 fullname = Joshua Small
 
 hasagent = yes
 
 hasdirectory = yes
 
 hasiax = no
 
 hasmanager = no
 
 hassip = yes
 
 hasvoicemail = no
 
 host = dynamic
 
 mailbox = 6001
 
 secret = X
 
 threewaycalling = yes
 
 registeriax = no
 
 registersip = yes
 
 canreinvite = no
 
 nat = no
 
 dtmfmode = rfc2833
 
 vmsecret = 1234
 
  
 
 ;some PSTNS
 
 [trunk_2]
 
 callerid = asreceived
 
 context = DID_trunk_2
 
 group = 2
 
 hasexten = no
 
 hasiax = no
 
 hassip = no
 
 trunkname = Ports 1,2,3,4
 
 trunkstyle = analog
 
 zapchan = 1,2,3,4
 
  
 
 ;my IP trunk
 
 [trunk_3]
 
 allow = all
 
 context = DID_trunk_3
 
 dialformat = ${EXTEN:1}
 
 hasexten = no
 
 hasiax = no
 
 hassip = yes
 
 host = gw02.mytel.net.au
 
 port = 5060
 
 registeriax = no
 
 registersip = yes
 
 secret = 
 
 trunkname = Custom - MyTel2
 
 trunkstyle = customvoip
 
 username = 
 
 type = friend
 
 nat = yes
 
  
 
 ;extensions.conf
 
 [numberplan-custom-1]
 
 plancomment = DialPlan1
 
 include = default
 
 include = parkedcalls
 
 exten = _0X!,1,Macro(trunkdial,${trunk_3}/${EXTEN:1})
 
 comment = _0X!,1,First,standard
 
 ;a failover to PSTN, not yet enabled
 
 ;exten = _0X!,2,Macro(trunkdial,${trunk_2}/0${EXTEN:1})
 
 ;comment = _0X!,1,First,standard
 
  
 
 At this point, everything appears to work fine. We can make calls from
 our several handsets using our voip link no problems.
 
 We have two different accounts with our provider, the goal being
 certain handsets will connect to this account and therefore be billed
 separately. I haven’t gotten as far as to add the extra handsets and
 set an appropriate dialplan, all I did was add this to users.conf:
 
  
 
 [trunk_extra]
 
 allow = all
 
 context = DID_trunk_3
 
 dialformat = ${EXTEN:1}
 
 hasexten = no
 
 hasiax = no
 
 hassip = yes
 
 host = gw02.mytel.net.au
 
 port = 5060
 
 registeriax = no
 
 registersip = yes
 
 secret = 
 
 trunkname = Custom - MyTel Two
 
 trunkstyle = customvoip
 
 username = XX
 
 type = friend
 
 nat = yes
 
  
 
 From this point on, my existing handsets don’t appear to be able to
 get a line out. My console looks like this (from the first call out):
 
 Connected to Asterisk 1.4.11 currently running on asterisk (pid =
 31999)
 
 -- Remote UNIX connection
 
 Verbosity is at least 8
 
 -- Executing [EMAIL PROTECTED]:1]
 Macro(SIP/8001-b7d0bb20, trunkdial|SIP/trunk_3/0425298582) in new
 stack
 
 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/8001-b7d0bb20,
 SIP/trunk_3/0425298582) in new stack
 
 -- Called trunk_3/0425298582
 
 [Sep 13 20:42:16] WARNING[32011]: chan_sip.c:12016
 handle_response_invite: Received response: Forbidden from 'Joshua
 Small sip:[EMAIL PROTECTED];tag=as29bb274d'
 
 -- SIP/trunk_3-097ac708 is circuit-busy
 
   == Everyone is busy/congested at this time (1:0/1/0)
 
 -- Executing [EMAIL PROTECTED]:2] Goto(SIP/8001-b7d0bb20,
 s-CONGESTION|1) in new stack
 
 -- Goto (macro-trunkdial,s-CONGESTION,1)
 
 -- Executing [EMAIL PROTECTED]:1]
 NoOp(SIP/8001-b7d0bb20, ) in new stack
 
   == Auto fallthrough, channel 'SIP/8001-b7d0bb20' status is
 'CONGESTION'
 
  
 
  
 
 Any advice on why our trunk_3 becomes congested, just because
 trunk_extra is set to exist, is appreciated.
 
 Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887
 959 | www.visinet.com.au 
 
 This e-mail is intended for use by the named recipients only and
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 this message that pertain to the sender's employer and its products
 and services represent the opinion of the sender and not
 necessarily those of the employer. 
 
  
 
 
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Re: [asterisk-users] Conference bridge.

2007-09-12 Thread Matt Riddell

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Alex Balashov wrote:
 Any recommendations for an affordable SIP conference bridge unit?  I mean 
 one that isn't crappy;  something where the duplex and cancellation 
 functions that are traditionally built into such devices actually work.

Most people tend to go for the polycom kit.

- --
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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uch487tBDa1dA+hPKIXbqcM=
=5os6
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Re: [asterisk-users] Agent Callback Login in 1.4

2007-09-12 Thread Matt Riddell

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Anthony Francis wrote:
 Awhile back I had heard some talk, in this list I believe that Agent 
 callback login was going to be deprecated in 1.4, I see it is still 
 there. Does anyone know what is happening with this?

It has been deprecated and use of it is not recommended.

- --
Kind Regards,

Matt Riddell
Director
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iD8DBQFG6JVHDQNt8rg0Kp4RAq30AKC3ELS6DodauFNnmcu9zcJYsmTCpQCfSCye
gZEbl6yWKi+EjUxUS4J2NuU=
=A2Wo
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Re: [asterisk-users] Agent Callback Login in 1.4

2007-09-12 Thread Ryan Stark

I tried to use it back in 1.4.6 or so and it is horribly broken, I ended up
rewriting the functionality with dynamic queue members in the dial plan.  I
really liked the call back agent feature set. I found it to be far superior
to dynamic queue member alternative.

-Ryan

On 9/12/07, Anthony Francis [EMAIL PROTECTED] wrote:

 Awhile back I had heard some talk, in this list I believe that Agent
 callback login was going to be deprecated in 1.4, I see it is still
 there. Does anyone know what is happening with this?

 --
 Thank you and have a wonderful day,

 Anthony Francis
 Rockynet VOIP
 (303) 444-7052 opt 2
 [EMAIL PROTECTED]


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Re: [asterisk-users] Conference bridge.

On Wed, 2007-09-12 at 16:44 -0400, Alex Balashov wrote:
 Any recommendations for an affordable SIP conference bridge unit?  I mean 
 one that isn't crappy;  something where the duplex and cancellation 
 functions that are traditionally built into such devices actually work.

Do you want something cheap or something that works?

You can't have both.

PaulH


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Re: [asterisk-users] Agent Callback Login in 1.4


It's a great feature, and one hopes it will return one day.

PaulH


On Wed, 2007-09-12 at 19:01 -0700, Ryan Stark wrote:
 I tried to use it back in 1.4.6 or so and it is horribly broken, I
 ended up rewriting the functionality with dynamic queue members in the
 dial plan.  I really liked the call back agent feature set. I found it
 to be far superior to dynamic queue member alternative. 
 
 -Ryan
 
 On 9/12/07, Anthony Francis [EMAIL PROTECTED] wrote:
 Awhile back I had heard some talk, in this list I believe that
 Agent
 callback login was going to be deprecated in 1.4, I see it is
 still
 there. Does anyone know what is happening with this?
 
 --
 Thank you and have a wonderful day, 
 
 Anthony Francis
 Rockynet VOIP
 (303) 444-7052 opt 2
 [EMAIL PROTECTED]
 
 
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Re: [asterisk-users] Callback for unanswered transfers...

2007-09-12 Thread Luis Antonio Prata Barbosa

Thank you.

2007/9/12, Atis [EMAIL PROTECTED]:

 On 9/12/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote:
  Hi,
 
  Does anybody know if there is a way for a call goes back to transferer
 if
  unanswered ?

 Yes, before Dial to transferer set some variable that have he's
 extension, and in your defined TRANSFER_CONTEXT, use Dial with g
 option. After that, check DIALSTATUS!=ANSWERED and Dial back.

 I have it working, if those details aren't enough, or something
 doesn't go well, just ask.

 Regards,
 Atis
 --
 Atis Lezdins,
 IT Responsible of BEST Riga,
 [EMAIL PROTECTED]
 ICQ: 142239285
 Skype: atis.lezdins
 Cell Phone: +371 28806004 [Tele2, Latvia]
 Work phone: +1 800 7502835 [Toll free, USA]
 ?BEST? - www.BEST.eu.org

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[asterisk-users] Trunk Outbound Route for a Cisco VOIP router?

2007-09-12 Thread Doug

Hi,

I am trying to set up TrixBox/FreePBX with
a trunk and outbound route to a Cisco VOIP
router.  Has anyone on the list done this
successfully?  Willing to share config file
snippets?

Mucho thanks if you can help!


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Re: [asterisk-users] Agent Callback Login in 1.4

2007-09-12 Thread Matt Riddell

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Paul Hales wrote:
 It's a great feature, and one hopes it will return one day.
 
 PaulH
 
 
 On Wed, 2007-09-12 at 19:01 -0700, Ryan Stark wrote:
 I tried to use it back in 1.4.6 or so and it is horribly broken, I
 ended up rewriting the functionality with dynamic queue members in the
 dial plan.  I really liked the call back agent feature set. I found it
 to be far superior to dynamic queue member alternative. 

As the previous mail noted, it can be recreated with dynamic queue
members and so is unlikely to make a return.

The only problem is that the example for how to do this is written in
AEL, and that may be more than first time users can get their head around.

I tried to find the link, but can't - maybe someone else can help with that.

- --
Kind Regards,

Matt Riddell
Director
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4LXESRhUBvMjTrIw2GsgbOg=
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[asterisk-users] No Sound on Zap Channels

2007-09-12 Thread Jon Weisman

All,

I've got a strange issue here. When I make a SIP call to say my voicemail app, 
I hear audio just fine. However when I dial from PSTN into my Asterisk box, I 
see that its playing the voice files, but I hear nothing, then the call drops. 
I'm running Fedora Core 6, and Asterisk 1.2.24. CLI output below. T-1 is PRI, 
showing normal, dchannel is up as well. Any help is greatly appreciated.


Thanks,
Jon


 -- Accepting call from '2125551212' to '6465551212' on channel 0/23, span 4
-- Executing VoiceMail(Zap/95-1, u100) in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Channel 0/23, span 4 got hangup request, cause 34
  == Spawn extension (default, 6465551212, 1) exited non-zero on 'Zap/95-1'
-- Hungup 'Zap/95-1'
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Re: [asterisk-users] Agent Callback Login in 1.4



Matt Riddell wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Paul Hales wrote:
   
 It's a great feature, and one hopes it will return one day.

 PaulH


 On Wed, 2007-09-12 at 19:01 -0700, Ryan Stark wrote:
 
 I tried to use it back in 1.4.6 or so and it is horribly broken, I
 ended up rewriting the functionality with dynamic queue members in the
 dial plan.  I really liked the call back agent feature set. I found it
 to be far superior to dynamic queue member alternative. 
   

 As the previous mail noted, it can be recreated with dynamic queue
 members and so is unlikely to make a return.

 The only problem is that the example for how to do this is written in
 AEL, and that may be more than first time users can get their head around.

 I tried to find the link, but can't - maybe someone else can help with that.

 - --
 Kind Regards,

 Matt Riddell
 Director
   
Not only that but AEL doesnt mesh with realtime.

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] Trunk Outbound Route for a Cisco VOIP router?

2007-09-12 Thread Jon Weisman

Doug,

Not sure on the trixbox side but for asterisk:

Asterisk server: 10.0.0.1
Cisco Gateway: 10.0.0.2

In sip.conf

[cisco]
context=cisco
type=friend
host=10.0.0.2
dtmf=rfc2833

extension.conf
exten=_011.,1,Dial(SIP/[EMAIL PROTECTED])

In the Cisco:

dial-peer voice 100 voip
application session
destination-pattern .T
session protocol sipv2
session target ipv4:10.0.0.1
session transport udp

dial-peer voice 1 pots
application session
destination-pattern 011T
no digit-strip
direct-inward-dial
port 0:D
forward-digits-all
!
sip-ua
retry invite 4
retry response 3
retry bye 2
retry cancel 2
sip-server ipv4:10.0.0.1

hope this helps

-Jon



- Original Message - 
From: Doug [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 11:18 PM
Subject: [asterisk-users] Trunk  Outbound Route for a Cisco VOIP router?


 Hi,

 I am trying to set up TrixBox/FreePBX with
 a trunk and outbound route to a Cisco VOIP
 router.  Has anyone on the list done this
 successfully?  Willing to share config file
 snippets?

 Mucho thanks if you can help!


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[asterisk-users] FW: Problems with two trunks

2007-09-12 Thread Joshua Small

Update on this:

 

I found that by changing insecure = very to insecure = invite, adding
the second trunk no longer stopped calls working.

I've read the documentation on this switch and still don't see how it
applies/is meant to get used.

 

Anyway, with this change in place, the following may help:

 

asterisk*CLI sip show registry

HostUsername   Refresh State
Reg.Time

gw02.mytel.net.au:5060  1 120 Request Sent


gw02.mytel.net.au:5060  2 105 Registered
Thu, 13 Sep 2007 23:33:47

 

I have set a dial plan so that some handsets use the  (not the real
number)  extension (which work) and now I only need to determine why
1 never seems to register.

 

If I remove all traces of the  connection from my config, 1
registers fine.

Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 959 |
www.visinet.com.au http://www.visinet.com.au/  

This e-mail is intended for use by the named recipients only and
contains confidential information. Opinions and other information in
this message that pertain to the sender's employer and its products and
services represent the opinion of the sender and not necessarily those
of the employer. 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
Small
Sent: Thursday, 13 September 2007 10:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problems with two trunks

 

Hi,

 

I am attempting to setup an asterisk server, current specs:

CentOS release 5 (Final)

Asterisk 1.4.11

Asterisk-gui checked out from SVN last week

 

I started with a fairly basic setup involving one VOIP provider who
provided one dial in number, and a couple of handsets. Config files are
below. It was pretty much totally built by Asterisk-gui, except for the
fact I had to add insecure=very into users.conf in order to stop the
dialin from our provider presenting an authentication error. Advice on
any more correct approach would be appreciated, but is not the focus of
this post:

 

Users.conf

;several handsets setup like this...

[6001]

callwaiting = yes

context = numberplan-custom-1

email = [EMAIL PROTECTED]

fullname = Joshua Small

hasagent = yes

hasdirectory = yes

hasiax = no

hasmanager = no

hassip = yes

hasvoicemail = no

host = dynamic

mailbox = 6001

secret = X

threewaycalling = yes

registeriax = no

registersip = yes

canreinvite = no

nat = no

dtmfmode = rfc2833

vmsecret = 1234

 

;some PSTNS

[trunk_2]

callerid = asreceived

context = DID_trunk_2

group = 2

hasexten = no

hasiax = no

hassip = no

trunkname = Ports 1,2,3,4

trunkstyle = analog

zapchan = 1,2,3,4

 

;my IP trunk

[trunk_3]

allow = all

context = DID_trunk_3

dialformat = ${EXTEN:1}

hasexten = no

hasiax = no

hassip = yes

host = gw02.mytel.net.au

port = 5060

registeriax = no

registersip = yes

secret = 

trunkname = Custom - MyTel2

trunkstyle = customvoip

username = 

type = friend

nat = yes

 

;extensions.conf

[numberplan-custom-1]

plancomment = DialPlan1

include = default

include = parkedcalls

exten = _0X!,1,Macro(trunkdial,${trunk_3}/${EXTEN:1})

comment = _0X!,1,First,standard

;a failover to PSTN, not yet enabled

;exten = _0X!,2,Macro(trunkdial,${trunk_2}/0${EXTEN:1})

;comment = _0X!,1,First,standard

 

At this point, everything appears to work fine. We can make calls from
our several handsets using our voip link no problems.

We have two different accounts with our provider, the goal being certain
handsets will connect to this account and therefore be billed
separately. I haven't gotten as far as to add the extra handsets and set
an appropriate dialplan, all I did was add this to users.conf:

 

[trunk_extra]

allow = all

context = DID_trunk_3

dialformat = ${EXTEN:1}

hasexten = no

hasiax = no

hassip = yes

host = gw02.mytel.net.au

port = 5060

registeriax = no

registersip = yes

secret = 

trunkname = Custom - MyTel Two

trunkstyle = customvoip

username = XX

type = friend

nat = yes

 

From this point on, my existing handsets don't appear to be able to get
a line out. My console looks like this (from the first call out):

Connected to Asterisk 1.4.11 currently running on asterisk (pid = 31999)

-- Remote UNIX connection

Verbosity is at least 8

-- Executing [EMAIL PROTECTED]:1]
Macro(SIP/8001-b7d0bb20, trunkdial|SIP/trunk_3/0425298582) in new
stack

-- Executing [EMAIL PROTECTED]:1] Dial(SIP/8001-b7d0bb20,
SIP/trunk_3/0425298582) in new stack

-- Called trunk_3/0425298582

[Sep 13 20:42:16] WARNING[32011]: chan_sip.c:12016
handle_response_invite: Received response: Forbidden from 'Joshua
Small sip:[EMAIL PROTECTED];tag=as29bb274d'

-- SIP/trunk_3-097ac708 is circuit-busy

  == Everyone is busy/congested at this time (1:0/1/0)

-- Executing [EMAIL PROTECTED]:2] Goto(SIP/8001-b7d0bb20,
s-CONGESTION|1) in new stack

-- Goto

Re: [asterisk-users] bug in 1.2.24

2007-09-12 Thread Anton Krall

Thank Isaac, Ill try it this way.. Im currently using this before entering
the queue so calls from the queue are recorded:

exten =
s,n,SetVar(MONITOR_FILENAME=/var/spool/asterisk/${TIMESTAMP}-${UNIQUEID}-${C
ALLERIDNUM}-Queue-Ventas)
exten = s,n,SetVar(TRANSFER_CONTEXT=internalphones)

So I could just run mixmonitor instead of those lines and thats it? Queue
call will be recorded and everything that happens afterwards if it is
transferred?
 
Saludos
 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Isaac Xiao
Sent: martes, 11 de septiembre de 2007 06:24 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] bug in 1.2.24

It is not a bug. attended Transfer is using Local channel, if you have a
look the debug log from CLI, you will see why it fails. To solve this
problem, enable recording before the calls go into the queue. 

Exten = ,1,MixMonitor(...)
Exten = ,2,Goto(ext-queue, , 1)

This will ensure you to record the customer/caller's channel instead of
exten's channel. So no matter where you transfer the call and as long as
the caller not hangup the call, it will be always recorded.

By the way, 1.2.24 stable, we got problem with 1.2.21. 1.2.17 seems
stable.

Good luck,
Isaac Xiao
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Re: [asterisk-users] Agent Callback Login in 1.4