date:20060221

Marc,
I have a box with two TDM400P's. All of the ports are
FXO's. System is working fine on CentOS.

Regards,
Dovid

 Can someone give me a definite answer as to wether
 or not you can
 reliably run multiple TDM400P's in the same machine?
 
 I need 4 x FXO and 4 x FXS to connect to both the
 PSTN and existing key
 system, but I have seen several threads suggesting
 that this is not a
 supported configuration



__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Recommended rack-mountable server anyone?

2006-02-21 Thread mitcheloc

Hey everyone,

I've been doing a lot of research into a decent server for Asterisk
but I seem to be running and circles and now I am turning to you. The
issue I have is it needs to be rack mountable (so a Dell SC430 isn't
going to work) and preferably have 3 pci ports. The problem that I
seem to be running into is that when I look at servers from Dell or
IBM or the like they only seem to support PCI-X which (from what I
understand) does not support the Digium cards that we already have and
that they still make. So if anyone has a suggestion or has a server
they rather prefer for it's reliability, expandability, etc, please
recommend it!

Thank you in advance,
Mitchel
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's

I know one of the guys there that runs the place. They
know a lot about asterisk. I cant say all that I know
but I will just say that soon they will be very
asterisk friendly. As far as getting a plan without an
adapter they do have a plan. It is called a
myDevicePlan. I am not sure if its on thier site or
not. If you email them they will send you a form. I
believe the address is [EMAIL PROTECTED]

Regards,
Dovid

--- John covici [EMAIL PROTECTED] wrote:

 As to myphonecompany.com, they seem to have never
 heard of asterisk --
 do they support not buying their adaptor, or how do
 they work things?
 
 Thanks.
 
 on Monday 02/20/2006 Dovid
 Bender([EMAIL PROTECTED]) wrote
   I personaly use VoipJet, Teliax and
 myPhoneCompany.
   They are all great. Dont remember if teliax
 supported
   IAX. I know that myPhoneCompany for sure dosent.
 They
   use SIP. I did however ind that thier voice
 quality is
   very good.
   
Can anyone give me some good recommendations
 for
VoIP providrs that
support Asterisk PBX's?  We're based in Georgia
 and
   
   
  
 __
   Do You Yahoo!?
   Tired of spam?  Yahoo! Mail has the best spam
 protection around 
   http://mail.yahoo.com 
   ___
   --Bandwidth and Colocation provided by
 Easynews.com --
   
   Asterisk-Users mailing list
   To UNSUBSCRIBE or update options visit:
 

http://lists.digium.com/mailman/listinfo/asterisk-users
 
 -- 
 Your life is like a penny.  You're going to lose it.
  The question is:
 How do
 you spend it?
 
  John Covici
  [EMAIL PROTECTED]
 ___
 --Bandwidth and Colocation provided by Easynews.com
 --
 
 Asterisk-Users mailing list
 To UNSUBSCRIBE or update options visit:
   

http://lists.digium.com/mailman/listinfo/asterisk-users
 


__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] sniffing sip password/uri/host info

2006-02-21 Thread Andreas Sikkema

 Ethereal would probably be a batter analyzer. Not sure how well it
 seppurts sip, though. Unlike tcpdump it won't work on-the-fly. But you
 can also get tcpdump to dump raw data and analyze it off-line with
 ethereal.

Ethereal can also show SIP traffic on-the-fly! 

update list of packets in real time and 
automatic scrolling in live capture

A sip display filter is needed so you only see SIP traffic, 
a sip capture filter might be needed for very busy networks

-- 
Andreas Sikkema   BBned NV
Software EngineerPlaneetbaan 4
+31 (0)23 70743422132 HZ Hoofddorp 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] good voip

Again. What do you need ? Incoming and outgoing,
trunking etc. ?

I personaly use.
Voipjet.com
myPhonecompany.com
Teliax.com
I have heard others talk about:
JunctionNetworks

There others that are just not coming to mine. If I
remember them I will try to email them as well.

Dovid

 Everything. I really don't know where to begin. We
 make and distribute 
 custom Linux boxes and to include a VOIP solution
 using Asterisk would 
 be great. Ultimately to usurp the phone co. entirely
 I suppose would be 
 the ultimate.
 ___



__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk behind Centrex

I do not know a lot about centrex but I know that most
PBX's support POTS lines (usually for faxing). You can
have them switch over the lines that they send you to
pots and then you can plug the lines in to a TDM400P.

Regards,
Dovid

--- Devin Heckman [EMAIL PROTECTED] wrote:

 Hi,
 
 I'm looking at setting up an Asterisk PBX in our
 office, which gets its
 phone lines (digital signaling, analog voice) from
 the main campus,
 which uses Centrex.
 
 Does anyone know if this falls under analog or
 digital for hardware
 buying? I was looking at getting a Digium
 TDM-series, but apparently our
 lines aren't pots (due to the digital signaling).
 
 Could someone enlighten me a bit?
 
 Thanks a bunch.
 
 
 Devin Heckman
 University of California, Berkeley
 RSSP-IT Residential Computing
 ___
 --Bandwidth and Colocation provided by Easynews.com
 --
 
 Asterisk-Users mailing list
 To UNSUBSCRIBE or update options visit:
   

http://lists.digium.com/mailman/listinfo/asterisk-users
 


__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning

2006-02-21 Thread Peter Fern

I had exactly the same experience running IAX2, but also experienced 
half-duplex calls on top of that (though I think that's a different but 
with IAX handoff), and in the end dropped it completely for SIP.


We run g729 over dedicated fibre, and the resyncs were occurring all 
over the place with quite ludicrous values logged for delay.  I tried 
tweaking the jitterbuf, turning it off completely, and reverting to the 
old jitterbuffer implementation. none of which made any difference.  I 
also tried with and without trunking enabled.


SIP is running much more acceptably now.

Adam Robins wrote:



After many days of playing with the new jitterbuffer and trunking options for IAX2, I 
have finally received almost acceptable quality.  I am receiving 5-8 complaints a day of 
calls breaking up from both the customer and agent sides.  What I have 
discovered is that in most of these cases, the new jitterbuffer performed a resync during 
the call.  Currently, I have the resyncthreshold, and all other jb parameters at their 
default levels  The traffic is running over a fairly high latency WAN connection between 
Canada and Atlanta (IAX2, ILBC).  Idle ping times run about 85ms.

Below are the resync messages for this past Friday.  Knowing that I have a slow 
connection, should I set the resync at a much higher level?  I appreciate any 
assistance you may provide.

Thanks,
Adam

Feb 17 09:07:41 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -34, 
this delay 1651, threshold 1488, new offset -1651
Feb 17 09:07:42 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, 
this delay -1684, threshold 1000, new offset 33
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 176, 
this delay 1835, threshold 1126, new offset -1835
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 32, 
this delay 1673, threshold 1062, new offset -1673
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, 
this delay -1663, threshold 1300, new offset -172
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, 
this delay -1635, threshold 1300, new offset -38
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -22, 
this delay 2335, threshold 1054, new offset -2373
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 11, 
this delay 2363, threshold 1082, new offset -2535
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -71, 
this delay 2249, threshold 1054, new offset -2249
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -180, 
this delay -2359, threshold 1360, new offset -14
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, 
this delay -2354, threshold 1300, new offset -181
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, 
this delay -2297, threshold 1240, new offset 48
Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 109, 
this delay 1556, threshold 1136, new offset -1556
Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -30, 
this delay -1439, threshold 1000, new offset -117
Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, 
this delay 1608, threshold 1048, new offset -1725
Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -29, 
this delay -1616, threshold 1058, new offset -109
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 21, 
this delay 1751, threshold 1620, new offset -1751
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, 
this delay 1724, threshold 1686, new offset -1724
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -60, 
this delay -1716, threshold 1000, new offset -8
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -119, 
this delay -1757, threshold 1000, new offset 6
Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 75, 
this delay 1421, threshold 1326, new offset -1421
Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 274, 
this delay 1595, threshold 1282, new offset -1595
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1311, 
this delay 820, threshold 1824, new offset -2415
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1349, 
this delay 761, threshold 1752, new offset -2182
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -299, 
this delay -2127, threshold 1598, new offset -288
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -270, 
this delay -2106, threshold 1540, new offset -76
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 98, 
this delay 1878, threshold 1206, new offset -1878
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 44, 
this delay 1799, threshold 1150, new offset -1799
Feb 17 11:46:15

Re: [Asterisk-Users] Tormenta CAS signaling

2006-02-21 Thread Steve Underwood


Viktor Tatianin wrote:


Hi Steve

I attempt change in  zapata.conf

cas=1-15:1101   but use zttool view ABCD bits 1010

Regards,
Viktor
 


Have you put the E1 in CAS mode with something like:

span=1,1,0,cas,hdb3

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood
Sent: Friday, February 10, 2006 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Tormenta CAS signaling


Viktor Tatianin wrote:

 


Hello

Can anyone know how may change(inverting) cas signaling ABCD bits at the
Tormenta 2 (four E1 ports) cards
My cards send idle code ABCD 0101 but my mux which use as channel bank wait
ABCD 1001


   


The idle code is set in zapata.conf. For example:

cas=1-15:1101

Sets CAS mode for channels 1 to 15, with the idle pattern 1101.

Regards,
Steve
 



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Multiple TDM400P's in a single machine


 I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing 
 key system, but I have seen several threads suggesting that this is 
 not a supported configuration
 
 This bad boy might be what you need:
 
http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM2400Ptab=details
 
 If not, consider an external channel bank:
 http://www.voipsupply.com/product_info.php?products_id=868
 http://www.voipsupply.com/product_info.php?products_id=781
 
 It would be great if you could let the list know which route you 
 take, and the success (or lack thereof) that you have with it!

Or, take a close look at the Sangoma A200D. Takes one pci slot but
can be expanded from a 4-port single card to 24 ports (fxs/fxo, mix
or match). The card has hardware echo cancellation with 128 tail
support.

The downside to the A200D is even though only one pci slot is used
to interface to the motherboard, the add-on daughter cards needed
to expand beyond four ports cover up other pci slots, leaving those
unusable in most cases.


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Recommended rack-mountable server anyone?

2006-02-21 Thread Alexander Burke


Hello, Mitchel!

At 07:41 AM 02/21/2006, you wrote:

I've been doing a lot of research into a decent server for Asterisk
but I seem to be running and circles and now I am turning to you. The
issue I have is it needs to be rack mountable (so a Dell SC430 isn't
going to work) and preferably have 3 pci ports. The problem that I
seem to be running into is that when I look at servers from Dell or
IBM or the like they only seem to support PCI-X which (from what I
understand) does not support the Digium cards that we already have and
that they still make. So if anyone has a suggestion or has a server
they rather prefer for it's reliability, expandability, etc, please
recommend it!


As I understand it, PCI-X is fully backwards-compatible with PCI (as 
in the presence of a PCI card on a PCI-X bus will cause that bus to 
drop back to regular PCI mode). If you want something super-reliable 
which can run Linux, Solaris, or Windows, and you require three PCI 
slots, this may interest you:

http://www.sun.com/servers/entry/x4200/

(Click on the Gallery link for pretty pictures.)

I'm seriously considering two X2100s (because I don't need four disks 
or any PCI cards):

http://www.sun.com/servers/entry/x2100/

These boxes will run Solaris, Linux, or (ack) Windows, and their 
remote monitoring/management support is second to none.


--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada 



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Recommended rack-mountable server anyone?

2006-02-21 Thread Cory Andrews


Supermicro!

Cory J Andrews

VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY
- Original Message - 
From: [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Tuesday, February 21, 2006 7:41 AM
Subject: [Asterisk-Users] Recommended rack-mountable server anyone?


Hey everyone,

I've been doing a lot of research into a decent server for Asterisk
but I seem to be running and circles and now I am turning to you. The
issue I have is it needs to be rack mountable (so a Dell SC430 isn't
going to work) and preferably have 3 pci ports. The problem that I
seem to be running into is that when I look at servers from Dell or
IBM or the like they only seem to support PCI-X which (from what I
understand) does not support the Digium cards that we already have and
that they still make. So if anyone has a suggestion or has a server
they rather prefer for it's reliability, expandability, etc, please
recommend it!

Thank you in advance,
Mitchel
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning

2006-02-21 Thread Adam Robins

Thank you for validating that I am not going mad!

I made some additional tweaks for today.  We'll see how it goes.  If not
well, then I'll try SIP for tomorrow.

Thanks,
Adam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Fern
Sent: Tuesday, February 21, 2006 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

I had exactly the same experience running IAX2, but also experienced
half-duplex calls on top of that (though I think that's a different but
with IAX handoff), and in the end dropped it completely for SIP.

We run g729 over dedicated fibre, and the resyncs were occurring all
over the place with quite ludicrous values logged for delay.  I tried
tweaking the jitterbuf, turning it off completely, and reverting to the
old jitterbuffer implementation. none of which made any difference.  I
also tried with and without trunking enabled.

SIP is running much more acceptably now.

Adam Robins wrote:

 
After many days of playing with the new jitterbuffer and trunking
options for IAX2, I have finally received almost acceptable quality.  I
am receiving 5-8 complaints a day of calls breaking up from both the
customer and agent sides.  What I have discovered is that in most of
these cases, the new jitterbuffer performed a resync during the call.
Currently, I have the resyncthreshold, and all other jb parameters at
their default levels  The traffic is running over a fairly high latency
WAN connection between Canada and Atlanta (IAX2, ILBC).  Idle ping times
run about 85ms.
 
Below are the resync messages for this past Friday.  Knowing that I
have a slow connection, should I set the resync at a much higher level?
I appreciate any assistance you may provide.
 
Thanks,
Adam
 
Feb 17 09:07:41 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay

-34, this delay 1651, threshold 1488, new offset -1651 Feb 17 09:07:42 
WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, this 
delay -1684, threshold 1000, new offset 33 Feb 17 10:21:04 
WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 176, this delay

1835, threshold 1126, new offset -1835 Feb 17 10:21:04 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay 32, this delay 1673, 
threshold 1062, new offset -1673 Feb 17 10:21:04 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1663, 
threshold 1300, new offset -172 Feb 17 10:21:04 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1635, 
threshold 1300, new offset -38 Feb 17 10:21:48 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -22, this delay 2335, 
threshold 1054, new offset -2373 Feb 17 10:21:48 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay 11, this delay 2363, 
threshold 1082, new offset -2535 Feb 17 10:21:48 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -71, this delay 2249, 
threshold 1054, new offset -2249 Feb 17 10:21:48 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -180, this delay -2359, 
threshold 1360, new offset -14 Feb 17 10:21:48 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -150, this delay -2354, 
threshold 1300, new offset -181 Feb 17 10:21:48 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -120, this delay -2297, 
threshold 1240, new offset 48 Feb 17 10:34:28 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay 109, this delay 1556, 
threshold 1136, new offset -1556 Feb 17 10:34:28 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -30, this delay -1439, 
threshold 1000, new offset -117 Feb 17 10:34:32 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1608, 
threshold 1048, new offset -1725 Feb 17 10:34:32 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -29, this delay -1616, 
threshold 1058, new offset -109 Feb 17 10:45:08 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay 21, this delay 1751, 
threshold 1620, new offset -1751 Feb 17 10:45:08 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1724, 
threshold 1686, new offset -1724 Feb 17 10:45:08 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -60, this delay -1716, 
threshold 1000, new offset -8 Feb 17 10:45:08 WARNING[1078] 
chan_iax2.c: Resyncing the jb. last_delay -119, this delay -1757, 
threshold 1000, new offset 6 Feb 17 11:28:45 WARNING[1078] chan_iax2.c:

Resyncing the jb. last_delay 75, this delay 1421, threshold 1326, new 
offset -1421 Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the 
jb. last_delay 274, this delay 1595, threshold 1282, new offset -1595 
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay

-1311, this delay 820, threshold 1824, new offset -2415 Feb 17 11:29:03

WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1349, this 
delay 761, threshold 1752, new offset -2182 Feb 17 11:29:03 
WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay

[Asterisk-Users] Fromstring when sending e-mail on recieved voicemail

2006-02-21 Thread Arne Morten Johansen


Hi. I'm having trouble controlling the user info when sending e-mails
from asterisk via sendmail to a Microsoft exchange server.

When I receive the email the sender is always
[EMAIL PROTECTED] and the name of the sender is always
Added by portage for asterisk. I want to change both sender-address
and the name of the sender.

I'm using Gentoo for my asterisk box.

Can anyone help me on this one? 

Regards
Arne Morten Johansen

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] pickup problem on Asterisk 1.2.4

2006-02-21 Thread Francesco Angi

Hi everybody,
I'm facing a strange problem after upgrading Asterisk from 1.0.9 to
1.2.4.
Sometimes, when receiving an incoming call from pstn, although my sip
phones ring correctly (I've got both softphones and hardware phones),
noone can pick up the call. Asterisk CLI shows me that the phones are
ringing, then nothing happens, so there's no problem _after_ someone
picked up, simply Asterisk doesn't notice a phone picked up!
Upgrading Asterisk I only did some changes to my dialplan, according to
the upgrade page.
My card is a TE110P, this is my zapata file:

[channels]
language=it

context=default

signalling=pri_cpe
switchtype=euroisdn

overlapdial=yes

pridialplan = unknown
prilocaldialplan = unknown  
nationalprefix = 0
internationalprefix = 00

echocancel=yes
echotraining = 100
echocancelwhenbridged=yes

immediate=no
group=1
language=it
musiconhold=default
channel = 1-15,17-31



Thanks for help,
_fangi_
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Set CallerIDNum for outgoing calls on a PRI+DDI line

2006-02-21 Thread Mimmus

Hi,
I'd like to know if and how can I set CallerIDNum for outgoing calls on a
PRI line with DDI.
Does anyone know if italian Telecom permit this?

Thanks
-- 
Domenico Viggiani

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] good voip

2006-02-21 Thread CyberSource


Dovid Bender wrote:

Again. What do you need ? Incoming and outgoing,
trunking etc. ?

I personaly use.
Voipjet.com
myPhonecompany.com
Teliax.com
I have heard others talk about:
JunctionNetworks

There others that are just not coming to mine. If I
remember them I will try to email them as well.

Dovid

  

Everything. I really don't know where to begin. We
make and distribute 
custom Linux boxes and to include a VOIP solution
using Asterisk would 
be great. Ultimately to usurp the phone co. entirely
I suppose would be 
the ultimate.

___





__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___

--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

  
I want to do everything, incoming, outgoing, real phone number (DID ?) 
Like I said, I am really new to the whole thing and need some very basic 
help. I know Linux very well but I am not a telephony guy, alot of the 
terminology to me is foreign.  I would like to be able to take my real 
phone number that I use with Ma Bell and have it come to my asterisk box 
and take a message or forward if needed and I would like to be able to 
call from my asterisk box using that phone number (at least for caller 
id purpose). If you can lead me off with some examples and/or providers 
so I can accomplish this, I would be very greatful. Thanks, Peter

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Setting up an EICON CARD with CAPI

2006-02-21 Thread cédric Buzay

Ok I update from sourceforge the chan-capi-cm 0.6.4 with the good 
capi.conf  and now it's working fine.
I still not know why the old one (package from debian) are not working 
with the old capi.conf ?


Thanks for your help ,

Cédric

Armin Schindler a écrit :

What version of chan_capi do you use? Your capi.conf is for an old 
chan_capi. If you use an old version, please update to chan_capi

from sourceforge.net and adapt your capi.conf.

Armin


On Tue, 21 Feb 2006, cédric Buzay wrote:

 


Hi everybody.

I'm trying to setting up a V4 BRI EICON card on ASTERISK 1.0.7
My linux is a debian.
It was working during a few days an suddenly (after a lot of reboot)
I've got this error message that seems to be very popular but I couldn't
find any
answer on the net :

==.
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
[chan_capi.so] = (Common ISDN API for Asterisk)
== Parsing '/etc/asterisk/capi.conf': Found
Feb 21 03:08:34 NOTICE[10319]: chan_capi.c:2645 load_module: unable to
listen!
Feb 21 03:08:34 WARNING[10319]: loader.c:345 ast_load_resource:
chan_capi.so: load_module failed, returning -1
== Unregistered channel type 'CAPI'
Feb 21 03:08:34 WARNING[10319]: loader.c:391 load_modules: Loading
module chan_capi.so failed!
===

I've got all the persmissions on the .conf files and on the /dev/capi20

my drivers are those :
dmesg | grep -i capi
Eicon DIVA - CAPI Interface driver (http://www.melware.net)
divacapi: Rel:2.0  Rev:1.24  Build: 105-75(local)
divacapi: module unloaded.
Eicon DIVA - CAPI Interface driver (http://www.melware.net)
divacapi: Rel:2.0  Rev:1.24  Build: 105-75(local)

and they work at a CAPI  level :

Update CFGLib information ... succeeded
Start adapter Nr:1 - 'Diva Server V-4BRI-8', SN: 23208 ... OK (already
active)
Successfully updated configuration of Diva Server V-4BRI-8 PORT: 0 SN:
Successfully updated configuration of Diva Server V-4BRI-8 PORT: 1 SN:
Successfully updated configuration of Diva Server V-4BRI-8 PORT: 2 SN:
Successfully updated configuration of Diva Server V-4BRI-8 PORT: 3 SN:
Successfully updated configuration of Diva TTY driver
Successfully updated configuration of Diva MTPX driver
Successfully updated configuration of Diva CAPI driver

My modules.conf seems correct:
=
[]
noload = app_intercom.so
; 
;  Explicitly load the chan_modem.so early on to be sure

;  it loads before any of the chan_modem_* 's afte rit
; 
; load = chan_modem.so

; load = res_musiconhold.so
load = chan_capi.so
; 
;  Load either OSS or ALSA, not both

;  By default, load OSS only (automatically) and do not load ALSA
; 
noload = chan_alsa.so

noload = chan_oss.so

; 
;  Module names listed in global section will have symbols globally

;  exported to modules loaded after them.
; 
[global]

;chan_modem.so=yes
chan_capi.so=yes

And my capi.conf also :

; 
;  CAPI config
; 
; 
[general]

nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]
msn=0MYNUMBER0
incomingmsn=*
controller=1
; softdtmf=1
; accountcode=
context=demo
;echosquelch=1
echocancel=yes
; echotail=64
; callgroup=1
; deflect=12345678
devices=2
=

Any ideas ???

Thanks

Cédric




___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

   




___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
   



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] sniffing sip password/uri/host info

2006-02-21 Thread Andrea Cristofanini - Gedam Europe Srl

 
  
  I want to sniff all these info to test a sip ip phone talking to a asterisk
  server.  I have used tcpdump, but It just shows the 
  
 
 Ethereal would probably be a batter analyzer. Not sure how well it
 seppurts sip, though. Unlike tcpdump it won't work on-the-fly. But you
 can also get tcpdump to dump raw data and analyze it off-line with
 ethereal.

Ethereal does a pretty good job at decoding both sip and iax packets.
I use it a lot (on a separate laptop).


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Fromstring when sending e-mail on recieved voicemail

2006-02-21 Thread Barry Flanagan




Arne Morten Johansen wrote:

Hi. I'm having trouble controlling the user info when sending e-mails
from asterisk via sendmail to a Microsoft exchange server.

When I receive the email the sender is always
[EMAIL PROTECTED] and the name of the sender is always
Added by portage for asterisk. I want to change both sender-address
and the name of the sender.



This is actually picked up out of /etc/passwd by default AFAIK. In 
voicemail.conf you can change the serveremail and fromstring 
settings, although I think this is only in 1.2.x.


Hope this helps.


-Barry Flanagan
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] asterisk related job offer in Florida

2006-02-21 Thread jarnaud

Hello,

I hope it's ok to post here for a job offer.

A dynamic IVR company has a current opportunity for a RD Jr developer.
The right candidate will have a background in developing and managing Linux 
based software systems, some experience in the IVR industry is a huge plus.

Expertise in the following areas is a must:
* VoIP: Asterisk
* Languages: Ruby, C, PHP, MySQL, VXML
* OS: GNU/Linux (Debian preferred)

Position Requirements:
* Analyze, design, develop, test, deploy and maintain key IVR components
* Complete development/programming as described in design specifications
* Create and document technical designs to achieve project requirements
* Maintain integrity of applications, follow standards
* Modify, enhance and maintain applications as required
* Develop strong relationships with partners 
* Investigate and resolve system problems in a timely manner
* Update system documentation as needed

Position is based in Plantation, FL (Broward county)

Submit your resume with salary requirements to [EMAIL PROTECTED]

Thank you,
---

JA
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] asterisk 1.2.4 doesn't detect the PSTN hang up

2006-02-21 Thread makevuy


Hy,

I'm writing from Spain.

I have the 1.2.4 asterisk version and 1.2.3 zaptel version. I've heart 
that this asterisk's version detects correctly de hang up of PSTN, but 
in my case this thing doesn't happen.


Moreover, my asterisk sends the next messages in the CLI:

Feb 21 15:03:13 WARNING[10363]: chan_zap.c:10876 setup_zap: Ignoring 
signalling
Feb 21 15:03:13 WARNING[10363]: chan_zap.c:10876 setup_zap: Ignoring 
answeronpolarityswitch
Feb 21 15:03:13 WARNING[10363]: chan_zap.c:10876 setup_zap: Ignoring 
hanguponpolarityswitch
Feb 21 15:03:13 WARNING[10363]: chan_zap.c:10876 setup_zap: Ignoring 
signalling

   -- Reconfigured channel 3, FXS Kewlstart signalling

What could be happened?

Thanks for all.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Multiple TDM400P's in a single machine

2006-02-21 Thread Sean Cook

Same setup with two TDM400 (8FXO) running for over a year.

On Tue, 2006-02-21 at 01:37 +0100, Thomas Artner wrote:
 Am Tuesday 21 February 2006 00:24 schrieb Marc Archer:
  Hi All,
 
 
 
  Can someone give me a definite answer as to wether or not you can
  reliably run multiple TDM400P's in the same machine?
 
  I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key
  system, but I have seen several threads suggesting that this is not a
  supported configuration
 
 
 
 i have two tdm400p's  (2xFXO, 6xFXS) in one desktop machine used as asterisk 
 server for a small office (so the pc hardware is nothing special).
 This configuration is running since two weeks without any problems!
 
 
 
 
  Thanks,
 
 
 
  Marc.
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 Asterisk-Users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk behind Centrex

2006-02-21 Thread Sean Cook

I believe that Centrex is ISDN correct?

Sean

On Tue, 2006-02-21 at 04:55 -0800, Dovid Bender wrote:
 I do not know a lot about centrex but I know that most
 PBX's support POTS lines (usually for faxing). You can
 have them switch over the lines that they send you to
 pots and then you can plug the lines in to a TDM400P.
 
 Regards,
 Dovid
 
 --- Devin Heckman [EMAIL PROTECTED] wrote:
 
  Hi,
  
  I'm looking at setting up an Asterisk PBX in our
  office, which gets its
  phone lines (digital signaling, analog voice) from
  the main campus,
  which uses Centrex.
  
  Does anyone know if this falls under analog or
  digital for hardware
  buying? I was looking at getting a Digium
  TDM-series, but apparently our
  lines aren't pots (due to the digital signaling).
  
  Could someone enlighten me a bit?
  
  Thanks a bunch.
  
  
  Devin Heckman
  University of California, Berkeley
  RSSP-IT Residential Computing
  ___
  --Bandwidth and Colocation provided by Easynews.com
  --
  
  Asterisk-Users mailing list
  To UNSUBSCRIBE or update options visit:

 
 http://lists.digium.com/mailman/listinfo/asterisk-users
  
 
 
 __
 Do You Yahoo!?
 Tired of spam?  Yahoo! Mail has the best spam protection around 
 http://mail.yahoo.com 
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 Asterisk-Users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

SV: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail

2006-02-21 Thread Arne Morten Johansen

Yeah I did change those. I'm using 1.0.8 (Or was it 9?).
It seams that the system overrides these settings? 

-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry Flanagan
Sendt: 21. februar 2006 14:54
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail



Arne Morten Johansen wrote:
 Hi. I'm having trouble controlling the user info when sending e-mails
 from asterisk via sendmail to a Microsoft exchange server.
 
 When I receive the email the sender is always
 [EMAIL PROTECTED] and the name of the sender is always
 Added by portage for asterisk. I want to change both sender-address
 and the name of the sender.
 

This is actually picked up out of /etc/passwd by default AFAIK. In 
voicemail.conf you can change the serveremail and fromstring 
settings, although I think this is only in 1.2.x.

Hope this helps.


-Barry Flanagan
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail on recievedvoicemail

2006-02-21 Thread Barry Flanagan




Arne Morten Johansen wrote:

Yeah I did change those. I'm using 1.0.8 (Or was it 9?).
It seams that the system overrides these settings? 


You may need to put the asterisk user into the trusted user list of 
sendmail - by default sendmail will not allow users apart from trusted 
one to change the From setting. Your server is running as the user 
asterisk and if you add that to the trusted list it might do the trick.


Sorry, but I don't know what sendmail Gentoo uses so can't tell you 
exactly how to do this.


Hope this helps.

-Barry Flanagan




-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry Flanagan
Sendt: 21. februar 2006 14:54
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail



Arne Morten Johansen wrote:


Hi. I'm having trouble controlling the user info when sending e-mails
from asterisk via sendmail to a Microsoft exchange server.

When I receive the email the sender is always
[EMAIL PROTECTED] and the name of the sender is always
Added by portage for asterisk. I want to change both sender-address
and the name of the sender.




This is actually picked up out of /etc/passwd by default AFAIK. In 
voicemail.conf you can change the serveremail and fromstring 
settings, although I think this is only in 1.2.x.


Hope this helps.


-Barry Flanagan
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



--

-Barry Flanagan
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

SV: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail

2006-02-21 Thread Arne Morten Johansen

Just one more question. In /etc/passwd there's a line with asterisk and 
added by portage in it. Can I just change this without screwing up 
everything? Or is there a command to change user info or something? As you can 
see, I'm not so good in Linux.

-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry Flanagan
Sendt: 21. februar 2006 14:54
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail



Arne Morten Johansen wrote:
 Hi. I'm having trouble controlling the user info when sending e-mails
 from asterisk via sendmail to a Microsoft exchange server.
 
 When I receive the email the sender is always
 [EMAIL PROTECTED] and the name of the sender is always
 Added by portage for asterisk. I want to change both sender-address
 and the name of the sender.
 

This is actually picked up out of /etc/passwd by default AFAIK. In 
voicemail.conf you can change the serveremail and fromstring 
settings, although I think this is only in 1.2.x.

Hope this helps.


-Barry Flanagan
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Multiple TDM400P's in a single machine

2006-02-21 Thread Gerard Saraber

3 TDM cards here, I had artifacts if any of the cards were sharing
interrupts, the trick was to add the cards 1 at the time to get them
each on their own irq. The system isn't in production yet, so I don't
know how well it'll hold up under load, so far so good in testing
though.
9xFXO 1xFXS 2xUnused

   CPU0   
  0:  592857335IO-APIC-edge  timer
  8:  0IO-APIC-edge  rtc
  9:  0   IO-APIC-level  acpi
177:   15559638   IO-APIC-level  eth0
185:3677271   IO-APIC-level  libata, NVidia CK8S
193:  0   IO-APIC-level  ehci_hcd:usb1
201:  0   IO-APIC-level  ohci_hcd:usb2
209:  0   IO-APIC-level  ohci_hcd:usb3
217:  592717484   IO-APIC-level  wctdm
225:  592712578   IO-APIC-level  wctdm
233:  592725907   IO-APIC-level  wctdm
NMI:  40812 
LOC:  592769967 
ERR:  0
MIS:  0

On Tue, 2006-02-21 at 09:23 +, Sean Cook wrote:
 Same setup with two TDM400 (8FXO) running for over a year.
 
 On Tue, 2006-02-21 at 01:37 +0100, Thomas Artner wrote:
  Am Tuesday 21 February 2006 00:24 schrieb Marc Archer:
   Hi All,
  
  
  
   Can someone give me a definite answer as to wether or not you can
   reliably run multiple TDM400P's in the same machine?
  
   I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key
   system, but I have seen several threads suggesting that this is not a
   supported configuration
  
  
  
  i have two tdm400p's  (2xFXO, 6xFXS) in one desktop machine used as 
  asterisk 
  server for a small office (so the pc hardware is nothing special).
  This configuration is running since two weeks without any problems!
  
  
  
  
   Thanks,
  
  
  
   Marc.
  ___
  --Bandwidth and Colocation provided by Easynews.com --
  
  Asterisk-Users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 Asterisk-Users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
-- 
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Application pppd

2006-02-21 Thread Giordano Grandis




Hi 
guys,
just a question: can 
i use the pppd application with a HFC PCI card using 
bristuff.

Thanks for 
all






Giordano
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, SunFire X2100)

2006-02-21 Thread Roberto Pereyra

Hi

Take a look this site:

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

roberto2006/2/20, Steve Kennedy [EMAIL PROTECTED]:
On Tue, Feb 21, 2006 at 12:17:43AM +1100, Mark Edwards wrote: At 06:33 AM 02/20/2006, you wrote:Please forgive the question, but what is the rationale behind using Solarisover Linux as an asterisk hosting platform?
Solaris is also a supported OS (well if you pay for it). It's also 64bit and any program written for earlier versions will just work. It's32 bit layer also works out the box (trying to use 32 bit apps on 64 bit
Linux can be a PITA).It's also very fast and debugging stuff can be much easier.Steve--NetTek LtdUK mob +44-(0)7775 755503UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]Euro Tech News Blog http://eurotechnews.blogspot.com___
--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: 
http://lists.digium.com/mailman/listinfo/asterisk-users-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID: 
[EMAIL PROTECTED]For reliable and professional DNS, use DNS Made Easy!http://www.dnsmadeeasy.com/u/14989

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Send flash through zap channel

2006-02-21 Thread Stefan Märkle

Hi everyone,

our setup includes a NEC PBX connected to our asterisk via bri lines.
The NEC has a doorphone feature, which is just an extension that calls you when 
someone rings. When connected to this extensions, a 'flash' signalling opens 
the door.

So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't 
able to do so.

Setup: asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1k, Quad-Bri Junghanns Card, Bris  
set on p2pte.

What I tried and didn't work:
* Using Flash() in dialplan - doesn't work since channel is Dial()-ed and 
doesn't allow applications at that very moment
* Typing *0 on phone = zap channel doc says this should send flash, but 
doesn't seem to work in bridged scenarios (ZAP=*=ZAP or SIP=*=ZAP)
* Typing # on phone = as of documentation, this sometimes emulates flash = 
not in my setup
* Tried the above from snom sip phone, sip ata with analogue phone and 
flash-key, mobile phone called in via another zap channel = no difference 
between the incomings

Has somebody any hints for me?

Stefan


-- 
Stefan Märkle   Netpioneer GmbH
Leitender Systemarchitekt   Beiertheimer Allee 18
[EMAIL PROTECTED]  76137 Karlsruhe 

*** Besuchen Sie uns vom 09.03.- 15.03.2006 auf der Cebit 2006 in Hannover. Sie 
finden uns in Halle 3 auf Stand D31 als Mitaussteller der Imperia AG ***

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk behind Centrex

2006-02-21 Thread Leo Ann Boon

Usually analog but can be IP as well. In Singapore, Singtel offers both 
analog and IP centrex services.



Sean Cook wrote:


I believe that Centrex is ISDN correct?

Sean

On Tue, 2006-02-21 at 04:55 -0800, Dovid Bender wrote:
 


I do not know a lot about centrex but I know that most
PBX's support POTS lines (usually for faxing). You can
have them switch over the lines that they send you to
pots and then you can plug the lines in to a TDM400P.

Regards,
Dovid

--- Devin Heckman [EMAIL PROTECTED] wrote:

   


Hi,

I'm looking at setting up an Asterisk PBX in our
office, which gets its
phone lines (digital signaling, analog voice) from
the main campus,
which uses Centrex.

Does anyone know if this falls under analog or
digital for hardware
buying? I was looking at getting a Digium
TDM-series, but apparently our
lines aren't pots (due to the digital signaling).

Could someone enlighten me a bit?

Thanks a bunch.


Devin Heckman
University of California, Berkeley
RSSP-IT Residential Computing
___
--Bandwidth and Colocation provided by Easynews.com
--

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
 

 


http://lists.digium.com/mailman/listinfo/asterisk-users
   


__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___

--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
   



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



 



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] calling from SIP to a h.323 device with oh323

2006-02-21 Thread Guillermo Salas M

On Mon, 2006-02-20 at 17:04 +0100, Marc Patino Gómez wrote:
 Hi,
 
 Can you post your working config, I'm wasting my time to config h323-sip
 



Is working now :)

I'm using asterisk-oh323 0.7.3 on my asterisk 1.2.4 box.

I've to configure in oh323.conf with gatekeeper=DISABLED and the context
of my sip clients. The H.323 device is configured to use the asterisk ip
address as gateway. With this config I can use SIP/IAX2 trunks to call
outside from the h.323 device and can call from SIP/IAX2 to H.323 and
from H.323 to my SIP/IAX2 devices :)

sip*CLI oh323 show conf
sip*CLI
 Configuration of OpenH323 channel driver
--
Version: 0.7.3
Listening on address: 0.0.0.0:1720
Gatekeeper used:  No gatekeeper
FastStart/H245Tunnelling/H245inSetup: ON/ON/ON
Supported formats in pref. order: alaw0 ulaw1 gsm2 g7233 g7294
Jitter buffer limits (min/max): 20-100 ms
TCP port range: 1 - 2
UDP (RAS) port range: 1 - 2
UDP (RTP) port range: 1 - 2
IP Type-of-Service value: 0
User input mode: tone
Max number of inbound H.323 calls: 100
Max number of outbound H.323 calls: 100
Max number of simultaneous H.323 calls: 100
Max call rate (ingress direction): 1.00/30
Default language: es
Default music class: default
Default context: from-internal

sip*CLI


I've to create the h.323 extentions for the two ports of my H.323 device
(ext 103 and 104 for port 1 and port 2) :

[ext-local]
include = ext-local-custom
exten = 101,1,Macro(exten-vm,novm,101)
exten = 101,hint,SIP/101
exten = 102,1,Macro(exten-vm,novm,102)
exten = 102,hint,SIP/102
exten = 103,1,Macro(exten-vm,novm,103)
exten = 103,hint,OH323/[EMAIL PROTECTED]
exten = 104,1,Macro(exten-vm,novm,104)
exten = 104,hint,OH323/[EMAIL PROTECTED]
exten = 555,1,Macro(exten-vm,novm,555)
exten = 555,hint,SIP/555




 
 Thanks
 
 Guillermo Salas M wrote:
 
 Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can
 make calls from one h.323 device to the world using sip trunks :)
 
 I can call to sip devices from the h.323 one. Now I want to make calls
 from sip to h.323 but it does not work. Maybe one of us have a
 configuration example to do this?
 
 I'm using the latest svn version (compiled yesterday).
 
 =
 Connected to Asterisk SVN-branch-1.2-r10487 currently running on sip
 (pid = 29977)
 nip*CLI
 
 
 
 Best regards,
 
 
   
 
 
 
-- 
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
SIP   : [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
http://www.telcocarrier.net

Linux User: 255902
Soporte en Linea en http://www.manta.telconet.net

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Send flash through zap channel

2006-02-21 Thread C F

I had to add this same feature recently for a client that has centrix
lines and wanted to use the conference feature of the centrix lines
which requires a flash, here is the setup:
PSTEN  CENTRIX LINES  ADIT 600 FXO CARD  ASTERISK  ADIT 600
FXS CARD  AVAYA PARTNER ACS R6.
When someone is on the phone from the Avaya system they wanted to be
able to flash the centrix line. Here is my
/etc/asterisk/extensions.conf:

[pstn-in]
exten = s,1,Noop(${CALLERID(all)})
exten = s,2,Noop()
exten = s,3,Noop()
exten = s,4,Set(DYNAMIC_FEATURES=inflash)
exten = s,5,Dial(Zap/g2,,t) ;Zap/g2 is the FXS card on the Adit600

[avayaout]
exten = _1NX,1,Set(DYNAMIC_FEATURES=outflash)
exten = _1NX,2,Macro(dialoutbound,${EXTEN:1},,${LTRUNK})

[macro-dialoutbound]
exten = s,1,Noop()
exten = s,2,Dial(${ARG3}/${ARG1},,T)
exten = s,3,Goto(${DIALSTATUS},1)
exten = s,103,Goto(3)

Here is my features.conf:
[applicationmap]
inflash = *4,caller,Flash,()
outflash = *3,callee,Flash,()

When a call comes in and they want to flash the line then they press
*4, if they call out and they want to flash the line then they press
*3.

I had to put in the t or T above so that asterisk stays in the media
path and listens for the *3/4.
Keep in mind:
1. It doesn't matter if you put in T or t, both the caller and the
callee can press *3/4 to activate this features, in your case this is
a huge security problem. As someone that needs access to the door
could just press the door call button, and then use *3/4 above to
flash the line and get in.
2. The flash app only plays on FXO ports, which means that you might
have to play around with the inflash and outlash callee/caller
options.

I'm not sure if you have any FXO ports in your config, but if you
don't it wont work.

Hope this helps.

On 2/21/06, Stefan Märkle [EMAIL PROTECTED] wrote:
 Hi everyone,

 our setup includes a NEC PBX connected to our asterisk via bri lines.
 The NEC has a doorphone feature, which is just an extension that calls you 
 when someone rings. When connected to this extensions, a 'flash' signalling 
 opens the door.

 So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't 
 able to do so.

 Setup: asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1k, Quad-Bri Junghanns Card, Bris  
 set on p2pte.

 What I tried and didn't work:
 * Using Flash() in dialplan - doesn't work since channel is Dial()-ed and 
 doesn't allow applications at that very moment
 * Typing *0 on phone = zap channel doc says this should send flash, but 
 doesn't seem to work in bridged scenarios (ZAP=*=ZAP or SIP=*=ZAP)
 * Typing # on phone = as of documentation, this sometimes emulates flash = 
 not in my setup
 * Tried the above from snom sip phone, sip ata with analogue phone and 
 flash-key, mobile phone called in via another zap channel = no difference 
 between the incomings

 Has somebody any hints for me?

 Stefan


 --
 Stefan Märkle   Netpioneer GmbH
 Leitender Systemarchitekt   Beiertheimer Allee 18
 [EMAIL PROTECTED]  76137 Karlsruhe

 *** Besuchen Sie uns vom 09.03.- 15.03.2006 auf der Cebit 2006 in Hannover. 
 Sie finden uns in Halle 3 auf Stand D31 als Mitaussteller der Imperia AG ***

 ___
 --Bandwidth and Colocation provided by Easynews.com --

 Asterisk-Users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problem win Unicall

2006-02-21 Thread acriollo

Hi Carlos , how do you did this part ?  I also included a bit timeout of 120 seconds in the dial command.

Thanks in advanced.

Regards
Athiel2006/2/10, Carlos Chavez [EMAIL PROTECTED]:



  
  


On Fri, 2006-02-10 at 08:38 -0200, Darlon wrote:

Try to change the value of protocolvariant in the unicall.conf. Please, send us here the result.







 I am using mx,10,4 in the protocol variant of unicall.conf.
What seemed to solve the problem is a very old tip that said I should
change the DEFAULT_T1 value of mfcr2.c fomr 5000 to something like
2. I also included a bit timeout of 120 seconds in the dial
command. For the moment every call is going through although I
still have some testing to do.








-- Carlos ChavezDirector de TecnologíaTelecomunicaciones Abiertas de México S.A. de C.V.Tel: +52-55-91169161 Ext 2001






-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBD7OR0Vhw7eWImqUMRAg3HAJ0bzfK7twgfRueZuhp984FQO91EoQCcCcfwauI71ZV1Cu3ZI+sMNkT52Wk==3e1I-END PGP SIGNATURE-
___--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Dell PowerEdge 2850

2006-02-21 Thread Richard OSS

Thank you very much Darren.I did look at Dell's website for the info but was not able to find the PCI voltage info. Perhaps I looked at the wrong place or missed it. Googling also did not give me answers.I called Dell myself and the tech support person was very helpful. He confirmed that Dell PE 2850 indeed has 3.3V for PCI X.Digium's support also confrimed this. They suggested I exchange my TE205P card with the TE210P card which works with PCI-X 3.3V. I am a newbie at Asterisk and am learning a lot thanks to the responses of the members of this list.richardDarren Reilly [EMAIL PROTECTED] wrote: Dell website Useguide has the info and its:Expansion Bus
Bus type PCI-X, PCI Express Expansion slots via riser card cage:PCI-Xone 3.3-V, 64-bit, 100-MHz or three 3.3-V, 64 bit, 133MHz PCI Expressone x4 lane width one x8 lane widthI would have thought the dell website would have been the first place to look.Took less than 2 mins to get the relevant info I cannot believe tech support couldn't give you that information.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus DarilionSent: 20 February 2006 23:23To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Dell PowerEdge 2850Ryan Amos wrote: I use a PE2850 with CentOS 4.2 on it (as parent says, it is essentially RHEL 4 without the support contract.) Extremely stable; no problems with asterisk at all. Dell makes 2 PCI riser cards for this serve
r, I
believe one of them has 5v slots. I have a 3.3v card so I can't tell you on that.Der PE2850 bietet eine Auswahl aus zwei E/A-Riser-Karten:o E/A-PCI-Riser-Karte (3 PCI-X-Steckplätze: 3 x 64 Bit/133 Mhz) odero E/A-PCI-Riser-Karte (2 PCI Express-Steckplätze: 1 x8-Lane und 1 x4-Lane, beide mit x8-Anschlüssen, und 1 PCI-X-Steckplatz: 1x 64 Bit/100 MHz)Both riser cards only have 64 Bit PCI slots. I think 64 bit is always 3.3 Volt - isn't it?regardsklaus -Ryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Monday, February 20, 2006 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell PowerEdge 2850 Don't know about the Dell. I personaly use Cent OS
(www.centos.org) which is RHEL ES without paying for it. I have it on my server and it seems to be holding up just fine. --- Richard OSS <[EMAIL PROTECTED]>wrote: Hello, Digium uses the Dell PE 2850 for their testing. This site says that 3.3V PCI slot. http://www.voip-info.org/wiki/view/Asterisk+hardware We are planning on purchasing a Dell PE 2850 and putting a TE205P card on it. However, the needs a 5V PCI slot. Does Dell PE 2850 has a 5V PCI slot? A person in our group tried to call Dell's customer support but they do not seem to know. We will also be using RHEL ES 4 as the OS. Anybody have experience (good/bad) for this type of configuration? We are going to use it primarily as a conferencing server
serving
30-50 simultaneous users. Can anybody recommend an alternative server that works well with TE205P and RHEL ES 4? This is our first time using Asterisk so we would like to have it pain free as much as possible. Thank you very much. richard ___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Voicemail 0 for operator call routing

2006-02-21 Thread Paul Tinsley

Does anyone know of a way to specify what extension is dialed when 0 is 
pressed in the voicemail system.  I have a situation where there is more 
than one secretary and they want the 0 to redirect to the appropriate 
secretary for the two groups of people. 


So an example would be:
555-1234 - voicemail - Secretary 1
555-1235 - voicemail - Secretary 2

Any help would be greatly appreciated.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)

2006-02-21 Thread Mike Clark


Steve Kennedy wrote:


On Tue, Feb 21, 2006 at 06:16:06AM -0500, Alexander Burke wrote:

 


Hello, Steve!
At 03:55 AM 02/21/2006, you wrote:
   


ztdummy was only used for timing. Linux 2.6 provides this function in
the kernel and I assume Solaris already has timing functions there.
 

Page 36 of Asterisk: The Future Of Telephony 
(O'Reilly Press) states that you either require a 
Digium PCI card to provide clocking, or ztdummy 
if you lack the PCI hardware required to provide 
timing. It goes on to mention that a UHCI USB 
controller was required pre-2.6 but now that 
there's a 1kHz clocking source in the kernel, 
ztdummy will attach to that instead, thus 
eliminating the requirement for the UHCI USB controller.
While it doesn't explicity say so, it seems to 
very strongly imply that either a PCI card or 
ztdummy are *required* for some Asterisk 
functionality (namely music-on-hold and 
conferencing, apparently). Is this actually not the case?
   



OK, that's not what I inferred - but you could be right?

Is there a definative answer on this, or I'll have to go and re-install
a test system ;)


Steve

 

When I try to start a meetme conference on an Asterisk system without 
TDM hardware or ztdummy loaded, Allison pleasantly tells me  That is 
not a valid conference.


Mike Clark
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail on recievedvoicemail

2006-02-21 Thread Barry Flanagan




Arne Morten Johansen wrote:

Just one more question. In /etc/passwd there's a line with asterisk and added by 
portage in it. Can I just change this without screwing up everything? Or is there a command 
to change user info or something? As you can see, I'm not so good in Linux.




Yes,  'usermod -c New Asterisk Description asterisk' should do the trick!

-Barry Flanagan
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] realtime sip_buddies does not store ip address


Hi list
i use SVN branch , i have real time working good with IAX2
The problem i have is for sip_buddies , any SIP  acount register does 
not store ip addres inside the table.

This only for SIP iax2 works great.

i also have in sip.conf
rtupdate=yes

any ideas ?



--
Cheers Andrea

Andrea Cristofanini
Gedam Europe S.r.l.
Gedam Advanced Communication LTD
mobile : +39 3291871756
office : +39 011 5694900
freevoip : 6838602
MSN : [EMAIL PROTECTED]
http://www.gedameurope.com
http://www.asterisknews.it
http://freevoip.gedameurope.com

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] good voip

Peter,
Diffrent companys offer diffrent services. For example
myPhoneCompany offers DID's for both inbound and
outbound. Thier basid DID plan is $5.00 with unlimited
incoming and 60 outgoing minutes. Each additional is
$0.029. Or $10.00 a month with 500 outgoing and the
same rates as above. Voipjet for instance only offers
outbound termination at I believe $0.013. So if you
were setting up a customer You would use
myPhoneCompany for incoming and VoipJet for outgoing.
Teliax I believe offers inbound (origination),
outbound (termination) or both. You also may want more
than one provider in case your primary one fails.
There are many options that you can use. The best way
is to mix and match. For instance one provider may be
cheaper to the UK while another is cheaper to China.
The best thing to start with is to order basic
accounts from the diffrent providers. Look at the
results that you get and go from there as to how you
want to develop your system.

Regards,
Dovid
 do do everything, incoming, outgoing, real
 phone number (DID ?) 
 Like I said, I am really new to the whole thing and
 need some very basic 
 help. I know Linux very well but I am not a
 telephony guy, alot of the 
 terminology to me is foreign.  I would like to be
 able to take my real 
 phone number that I use with Ma Bell and have it
 come to my asterisk box 
 and take a message or forward if needed and I would
 like to be able to 
 call from my asterisk box using that phone number
 (at least for caller 
 id purpose). If you can lead me off with some
 examples and/or providers 
 so I can accomplish this, I would be very greatful.
 Thanks, Peter


__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] G723 error

2006-02-21 Thread Matt

Ok,
Right now I have
disallow=all
allow=ulaw
allow=g723

Does it read it bottom up maybe?

On 2/16/06, yusuf [EMAIL PROTECTED] wrote:
 Matt,

 I you dont define a sip user/peer and just use a dial, asterisk will
 automatically use the codec that it prefers,  in my experince whenever i
 dial SIP without defining a sip user/peer it always dials g711alaw/ulaw.

 So in sip.conf in [general] (which would set codec choice for ALL sip
 calls) or in the defined section for sip device/user/peer have
 disallow=all
 allow=g723


 Matt wrote:
  Well... correct except that there is no [sipdevice].. it is all done
  through IP registration on the other person's end.So.. all I have
  is the dial statement.  Is there a way to set a variable or something
  right before the dial? (To my knowledge there isn't).
 
  On 2/15/06, yusuf [EMAIL PROTECTED] wrote:
 
 I am assuming you made a profile in sip.conf like so
 
 [sipdevice]
 type=peer
 host=x.x.x.x
 ...
 .
 .
 disallow=all
 allow=ulaw
 
 and in extensions.conf
 
 exten = _X.,1,Dial(SIP/sipdevice/${EXTEN})
 
 then this MUST work.  :)
 
 you can do a sip debug or set debug 10
 
 yusuf
 
 Matt wrote:
 
 Hi,
 How do I specify a codec to use for a SIP call?
 
 IE.. If I'm doing Dial(SIP/blah) for some reason the call is
 connecting using the codec at the bottom of my allow list rather then
 top (G711u)... and I'd like to force it to G711u if possible.
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 Asterisk-Users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 Asterisk-Users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
  ___
  --Bandwidth and Colocation provided by Easynews.com --
 
  Asterisk-Users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 

 ___
 --Bandwidth and Colocation provided by Easynews.com --

 Asterisk-Users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] how to add stun functionality in asterisk

2006-02-21 Thread Matt

JP,
There isn't much to show :)
Yes.. I am running the STUN server on the asterisk box so that VoIP
ATA's and phones behind firewall's can connect to the asterisk server
with no ports needing to be opened.

Setup is...
download stund.
unzip.. compile... run
WALA!  Stun server :)

Then just put the address for the stun server in your ATA and it also
just works.  You may need to tinker with the VIA settings in your
ATA.

On 2/18/06, Koopmann, Jan-Peter [EMAIL PROTECTED] wrote:
 On Friday, February 17, 2006 7:34 PM Matt wrote:

  Yes Sir!   This is what I use:
  http://www.vovida.org/applications/downloads/stun/
 
  Works like a charm!  Been running it in production for about a year.

 Good hint. Can you possibly provide a bit more insight on this? Are you 
 running STUN so that your phones behind NAT can easily connect to your server 
 or the other way around? I would really like to see the relevant parts of 
 your setup.

 Kind regards,
   JP
 ___
 --Bandwidth and Colocation provided by Easynews.com --

 Asterisk-Users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Send flash through zap channel

2006-02-21 Thread Ira


At 07:24 AM 02/21/2006, you wrote:
* Using Flash() in dialplan - doesn't work since channel is 
Dial()-ed and doesn't allow applications at that very moment
* Typing *0 on phone = zap channel doc says this should send flash, 
but doesn't seem to work in bridged scenarios (ZAP=*=ZAP or SIP=*=ZAP)
* Typing # on phone = as of documentation, this sometimes emulates 
flash = not in my setup
* Tried the above from snom sip phone, sip ata with analogue phone 
and flash-key, mobile phone called in via another zap channel = no 
difference between the incomings


Transfer to this extension works. This example flashes and then calls 
back 1XX if you send it to 6XX.

exten = _6[0-2][0-4],1,Flash()
exten = _6[0-2][0-4],2,Dial(SIP/1${EXTEN:1},,rtT)

OR you can try this:

in features.conf:

[applicationmap]
zapflash=*3,callee,flash

if you put any spaces in the above line, it will not work!!!

in extensions.conf add this line right before the dial commands where 
you want this to work:


exten = s,12, set(DYNAMIC_FEATURES=zapflash)

Then *3 should flash the line.

Ira 



--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.1.375 / Virus Database: 267.15.12/265 - Release Date: 02/20/2006


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Incoming ISDN DATA calls answered by asterisk IVR! - How to stop that?

2006-02-21 Thread Pisac

When incoming DATA calls arrive on ISDN, Asterisk recognise that this is
DATA call, but behaving like it is voice call: Answering call, playing
IVR messages, etc...

How to stop that? I want that only VOICE calls are answered by Asterisk,
and DATA/FAX to be ignored.

(I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f with ZapHFC ISDN BRI
lines)

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problem win Unicall

2006-02-21 Thread Carlos Chavez





On Tue, 2006-02-21 at 09:52 -0600, acriollo wrote:

Hi Carlos , how do you did this part ?  I also included a bit timeout of 120 seconds in the dial command.

Thanks in advanced.

Regards
Athiel



 It should say a BIG timeout, not a BIT, sorry. Just do a Dial(Unicall/g1/${EXTEN},120,${OPTIONS})







-- 
Carlos Chavez
Director de Tecnologa
Telecomunicaciones Abiertas de Mxico S.A. de C.V.
Tel: +52-55-91169161 Ext 2001








signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] What business IP phone to use

2006-02-21 Thread mustardman29

 

I have been struggling with this issue for about a year now.  There were
just too many IP phones to choose from at all sorts of price points and not
enough information about any of them.  Now I am looking at the situation
again and if anything it has gotten worse.  There are even more phones and
all sorts of opinions.  For every person that says phone x is great there is
someone else complaining about it.

I ended up buying a Grandstream GXP2000 and an Aastra 9133i to test so I
pretty much know what those two phones are about.  Lot's of people talking
about Polycom phones but they still seem to have their problems and since
they don't officially support Asterisk I have my concerns.  I really don't
want to have to keep buying phones to find out for myself as it get's
expensive real fast.

Is there any unbiased comparison of various phones and features anywhere.
If someone wrote a book I'd buy it but it would probably be obsolete before
it was published with the rate of new IP phone introductions and firmware
revisons.  I hear some people praising the GXP2000 phones and I gotta wonder
what they are smokin (regardless of firmware revison) so I just don't know
who to believe anymore.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Sangoma A200D analog card with fxo's

FYI...

Just installed one of the new Sangoma A200D analog pstn cards with the
hardware echo canceller on a trial basis. The card has four fxo interfaces.

Excellent audio quality, excellent echo cancelling, and excellent audio 
levels.

The four pstn lines at this location are rather long analog loops that
have rather long echo trails. I started with a pair of x100p's a couple
of years ago, swapped those out for one of the first TDM04b cards, had
the TDM04b replaced with a later revision (H), and have always had at
least some echo on pstn calls.

Our pstn lines have a -7.1 measured loss from the CO's milliwatt generator.
I've configured this new card with gains of 7.0 db to compensate for that
loss, and audio level is now extemely good.

Presumably the Sangoma hardware canceller handles much longer echo tails,
and those tails have been completely eliminated.

The card's setup was not exactly clear has the documentation for this
new card is somewhat fragmented across multiple readme's, etc.

Based on about one hour's worth of use, I'd recommend this card over
everything that I've tested. (Testing has included several ata type
devices plus the x100p and tdm card.)

Will be testing analog fax and many other items over the next several
days/weeks.


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] good voip

2006-02-21 Thread Martin Joseph



This is also very dependent on where you are and who your ISP is...

I used Teliax and there setup instructions and support are excellent.  
Unfortunately for me,  my ISP (frickin comcast) has a very poor route 
to Teliax's servers.  This seems to be somewhat changeable,  but is 
consistently poor enough that I had to explore other options, which I 
am still doing.


I now prepaid $10 to Nufone.net, who have a really poor website, but by 
all accounts here on the list, provide top shelf performance.  This 
seems to be true as my outgoing calls are no sounding much better 
(according to the called parties).


In any case, the prepaid thing is a boon for testing these options.  
Prepaying $10 gets you lots of opportunity to make calls and see how 
they sound and work for you.


Good Luck,
Marty

PS A central resource of various Voip terminators and the quality of 
routes to/from various ISP's would be a great boon.  Is there such a 
thing?


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Outbound Routing does not use Multiple Trunks

2006-02-21 Thread Nate List


Hello,
   I have a TDM400  and currently have 2 of the ZAP Trunks configured 
on it.  Zap/1-1 and Zap/2-1.  I am Running [EMAIL PROTECTED] Version 2.4 
with AMP version 1.10.010


In my Outbound Routing I have the Trunk Sequence set up so that 0 is 
Zap/1-1 and 1 is ZAP/2-1  What I see is that when Trunk Sequence 0 is 
full, it does not open Trunk Sequence 1.  I have found that this is true 
even if I have Trunk Sequence 0 set to a VoIP Line  and Max channels is 
reached, it will not open Trunk Sequence 1. 

If i have Trunk Sequence 0 set to Zap/g0 then it will open the other Zap 
channels in order, but i need to be able to order the ZAP channels 
because they are charged at different rates. 

Have others experienced this issue?  What should I be looking at to 
debug this?  I have included the output below.  SIP/700 initiated a call 
and Zap/1-1 answered, but when SIP/731 attempted a call, it just sat 
there and eventually hangs up.


Thanks,
Nate

-- Executing Macro(SIP/700-8d41, dialout-trunk|2|9**|) in new stack
   -- Executing GotoIf(SIP/700-8d41, 1?3:2)) in new stack
   -- Goto (macro-dialout-trunk,s,3)
   -- Executing Macro(SIP/700-8d41, user-callerid) in new stack
   -- Executing DBget(SIP/700-8d41, AMPUSER=DEVICE/700/user) in new 
stack

   -- DBget: varname=AMPUSER, family=DEVICE, key=700/user
   -- DBget: set variable AMPUSER to 700
   -- Executing DBget(SIP/700-8d41, 
AMPUSERCIDNAME=AMPUSER/700/cidname) in new stack

   -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=700/cidname
   -- DBget: set variable AMPUSERCIDNAME to 2002-ATA
   -- Executing GotoIf(SIP/700-8d41, 0?5) in new stack
   -- Executing SetCallerID(SIP/700-8d41, 2002-ATA 700) in new 
stack
   -- Executing NoOp(SIP/700-8d41, Using CallerID 2002-ATA 700) 
in new stack

   -- Executing Macro(SIP/700-8d41, record-enable|700|OUT) in new stack
   -- Executing GotoIf(SIP/700-8d41, 0  0?2:4) in new stack
   -- Goto (macro-record-enable,s,4)
   -- Executing AGI(SIP/700-8d41, 
recordingcheck|20060221-113809|1140539889.439) in new stack

   -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 recordingcheck|20060221-113809|1140539889.439: Outbound recording not 
enabled

   -- AGI Script recordingcheck completed, returning 0
   -- Executing NoOp(SIP/700-8d41, No recording needed) in new stack
   -- Executing Macro(SIP/700-8d41, outbound-callerid|2) in new stack
   -- Executing DBget(SIP/700-8d41, 
USEROUTCID=AMPUSER/700/outboundcid) in new stack

   -- DBget: varname=USEROUTCID, family=AMPUSER, key=700/outboundcid
   -- DBget: set variable USEROUTCID to
   -- Executing GotoIf(SIP/700-8d41, 1?4) in new stack
   -- Goto (macro-outbound-callerid,s,4)
   -- Executing GotoIf(SIP/700-8d41, 1?6) in new stack
   -- Goto (macro-outbound-callerid,s,6)
   -- Executing NoOp(SIP/700-8d41, CallerID set to 2002-ATA 
700) in new stack

   -- Executing SetGroup(SIP/700-8d41, OUT_2) in new stack
   -- Executing CheckGroup(SIP/700-8d41, 1) in new stack
   -- Executing SetVar(SIP/700-8d41, DIAL_NUMBER=9**) in new stack
   -- Executing SetVar(SIP/700-8d41, DIAL_TRUNK=2) in new stack
   -- Executing AGI(SIP/700-8d41, fixlocalprefix) in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
 fixlocalprefix: Added prefix. New number: 16**9**
   -- AGI Script fixlocalprefix completed, returning 0
   -- Executing SetVar(SIP/700-8d41, OUTNUM=16**9**) in new stack
   -- Executing Cut(SIP/700-8d41, custom=OUT_2|:|1) in new stack
   -- Executing GotoIf(SIP/700-8d41, 0?16) in new stack
   -- Executing Dial(SIP/700-8d41, ZAP/1-1/16**9**) in new stack
   -- Called 1-1/16**9**
   -- Zap/1-1 answered SIP/700-8d41
   -- Executing Macro(SIP/731-d09e, dialout-trunk|2|3**|) in 
new stack

   -- Executing GotoIf(SIP/731-d09e, 1?3:2)) in new stack
   -- Goto (macro-dialout-trunk,s,3)
   -- Executing Macro(SIP/731-d09e, user-callerid) in new stack
   -- Executing DBget(SIP/731-d09e, AMPUSER=DEVICE/731/user) in new 
stack

   -- DBget: varname=AMPUSER, family=DEVICE, key=731/user
   -- DBget: set variable AMPUSER to 731
   -- Executing DBget(SIP/731-d09e, 
AMPUSERCIDNAME=AMPUSER/731/cidname) in new stack

   -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=731/cidname
   -- DBget: set variable AMPUSERCIDNAME to Nates Home
   -- Executing GotoIf(SIP/731-d09e, 0?5) in new stack
   -- Executing SetCallerID(SIP/731-d09e, Nates Home 731) in 
new stack
   -- Executing NoOp(SIP/731-d09e, Using CallerID Nates Home 
731) in new stack

   -- Executing Macro(SIP/731-d09e, record-enable|731|OUT) in new stack
   -- Executing GotoIf(SIP/731-d09e, 0  0?2:4) in new stack
   -- Goto (macro-record-enable,s,4)
   -- Executing AGI(SIP/731-d09e, 
recordingcheck|20060221-113834|1140539914.441) in new stack

   -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 recordingcheck|20060221-113834|1140539914.441: Outbound recording not 
enabled

   -- AGI Script recordingcheck completed, returning 0
   -- Executing NoOp(SIP

[Asterisk-Users] Looking for programer...

2006-02-21 Thread Doug G

ITSP seeking C programmer to work on Asterisk and SER. 

[EMAIL PROTECTED]

Located in Northern NJ


Sorry if I should not post this here


Doug 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Call queue design issues and suggestions

2006-02-21 Thread Joe

Greetings to all.

I am currently implementing call queues for a customer and have come across
several problems.

The customer is an airline representative, and will be using call queues for
different airline reservations. The customer requires that any agent be able
to login to any number of queues. This means that queue members have to be
dynamic, not using member = agent/101 for example.

I am not sure of the best way to accomplish this.

I initially just setup agentcallback, and hard coded the agents in each
queue, but this means that when an agent logs in he/she will be in all
queues where member = agent/xxx.

My next thought was to use a combination of agentcallback and addQueueMember
to add SIP extensions to particular queues. I currently have a mechanism by
which the user can dial a number, enter the two letter airline code, mysql
translates this airline code into a real queue name, and the user is then
added to this queue. Of course the two letter airline code could be used for
the queue name to avoid the mysql lookup, something like queue-xx. Along
these lines, does anyone know if it is possible to use AddQueueMember with
Agent/xxx, or just with real extensions? The main problem with this is that
there would be no way to globally logoff agents (if real extensions had to
be used) from all the queues they are logged in to.

My current idea is to use agentcallback in combination with a php/mysl
interface. This of course would require realtime queue configuration. The
user would use agentcallback to login, and the web interface to choose the
queues he/she wanted to join.

The customer also wants a way of seeing which queues the agents are logged
into. This could also be run from mysql backend. I would also like to some
how integrate this into the Cisco 7940 xml capabilities.

Would love to hear form anyone regarding these issues.

Regards,
Joe



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] how to add stun functionality in asterisk

2006-02-21 Thread Bill Gibbs

What's the benefit of using stund vs nat=yes in your sip.conf for that
device?  I haven't had any issues behind firewalls when I enable that
option, and no ports are needed to be opened.

Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, February 21, 2006 11:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] how to add stun functionality in asterisk

JP,
There isn't much to show :)
Yes.. I am running the STUN server on the asterisk box so that VoIP
ATA's and phones behind firewall's can connect to the asterisk server
with no ports needing to be opened.

Setup is...
download stund.
unzip.. compile... run
WALA!  Stun server :)

Then just put the address for the stun server in your ATA and it also
just works.  You may need to tinker with the VIA settings in your
ATA.

On 2/18/06, Koopmann, Jan-Peter [EMAIL PROTECTED] wrote:
 On Friday, February 17, 2006 7:34 PM Matt wrote:

  Yes Sir!   This is what I use:
  http://www.vovida.org/applications/downloads/stun/
 
  Works like a charm!  Been running it in production for about a year.

 Good hint. Can you possibly provide a bit more insight on this? Are
you running STUN so that your phones behind NAT can easily connect to
your server or the other way around? I would really like to see the
relevant parts of your setup.

 Kind regards,
   JP
 ___
 --Bandwidth and Colocation provided by Easynews.com --

 Asterisk-Users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] BAD/GOOD Echo Cancel

2006-02-21 Thread Doug Lytle


Doug Lytle wrote:

Doug Lytle wrote:

[EMAIL PROTECTED] wrote:

I put a Tellabs 64ms echo canceller into my facility this weekend and 
am praying that it removes are echo problem.  If it does, I plan on 
making it a standard on my Asterisk installs that have a channel bank 
or T1.




Well, the day is almost over here and not one echo reported today.  
Very impressive!  I had 5 more cards delivered today.




Just as a follow up to this. 

I purchased 5 Tellabs 64ms cards on ebaY.  The very 1st card that I put 
into production worked quite nicely (Little residual echo, but very much 
an improvement).  I was on site this weekend and figured I would test 
out the remaining cards, one at a time to confirm they were in working 
condition.  This upcoming Monday, I had many complaints of echo.  Went 
back Tuesday morning and went over the settings of the previous card 
compared to this replacement.  I turns out that only 1 of the 5 cards 
had option 38 available (Send Side Echo Cencellation).


So, if you are in the market for one of these cards because of local 
echo, you'll want to confirm with the vendor of 'Send side Echo 
Cancellation', or at least ask him to check for option 38 after powering 
up. Out of the 5 purchased, only 1 had it.


Doug

--
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] What business IP phone to use

2006-02-21 Thread Ross C

[Mr.] Mustard,

There's no one-stop IP phone review site that I know of (that has one
person/company comparing all of the IP phones side by side).  
You're right, the gxp-2000 is a little on the low end as IP phones go.
However, you're also getting a lot of features for your buck with the GXP.
I used the GXP2000's in a bakery installation; the users of the phone always
have stuff all over their hands, thus I didn't see much sense in putting a
really nice phone there.  Two of the phones have already needed to be
replaced because of people spilling liquids all over them; it was only $100
to replace a GXP2000 vs. 200+ to replace a nice polycom with many call
appearances.
Regarding the polycoms--
I wouldn't worry about the polycoms not 'officially' supporting asterisk.
LOTS of people use them with Asterisk (including myself).  For me, the
biggest pain was getting them configured correctly (the xml config files are
a horrendous PITA--if someone were to write a book, I'd prefer it be on this
;) ).  BUT once they're configured, I LOVE them. And so do the users of the
phones. They have great build quality and a great speakerphone (one of the
best).  In short, I would give the Polycoms a solid recommendation for an
all-around good business phone to use with Asterisk.

I know lots of people also love the Snoms.  I can't really vouch for them
too much; I have one, I just haven't used it really.

Someone should make an epinions.com of sorts for IP phones and IP phone
equipment.  I think it would get used...


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mustardman29
Sent: Tuesday, February 21, 2006 11:58 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] What business IP phone to use

 

I have been struggling with this issue for about a year now.  There were
just too many IP phones to choose from at all sorts of price points and not
enough information about any of them.  Now I am looking at the situation
again and if anything it has gotten worse.  There are even more phones and
all sorts of opinions.  For every person that says phone x is great there is
someone else complaining about it.

I ended up buying a Grandstream GXP2000 and an Aastra 9133i to test so I
pretty much know what those two phones are about.  Lot's of people talking
about Polycom phones but they still seem to have their problems and since
they don't officially support Asterisk I have my concerns.  I really don't
want to have to keep buying phones to find out for myself as it get's
expensive real fast.

Is there any unbiased comparison of various phones and features anywhere.
If someone wrote a book I'd buy it but it would probably be obsolete before
it was published with the rate of new IP phone introductions and firmware
revisons.  I hear some people praising the GXP2000 phones and I gotta wonder
what they are smokin (regardless of firmware revison) so I just don't know
who to believe anymore.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Download Asterisk: The Future Of Telephony [More Info]

2006-02-21 Thread Bob McDowell


Speaking of this book, where can I get it?  Obviously I can read the
pdf, but I lack the facility to print it in any usable fashion.  The
labor and materials that I have spent on trying to print it thus far
probably outweighs the cost of the silly thing.  Is it only available
online, or do you think Barnes and Noble, Borders, etc might have it?


Bob McDowell

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Burke
Sent: Monday, February 20, 2006 6:22 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Download Asterisk: The Future Of Telephony
[More Info]

One thing I forgot to mention: I also cropped the registration and cut
marks off the sides of the pages. If you want the uncropped version,
get:
http://www.alexburke.ca/asterisk-tfot-uncropped.pdf

Sorry about the excessive noise, but I figured I should mention this.


Date: Mon, 20 Feb 2006 18:55:50 -0500
To: asterisk-users@lists.digium.com
From: Alexander Burke [EMAIL PROTECTED]
Subject: Download Asterisk: The Future Of Telephony

Hello, list!

I'm hosting a mirror of the book Asterisk: The Future Of Telephony
by O'Reilly Press, published under the Creative Commons license; I
believe this license allows me to do this, but if I'm mistaken, please
let me know.

I've taken the liberty of fixing the page numbers so Acrobat is now
aware of the correct number of each page, and shrinking the filesize
with Acrobat's Reduce File Size tool (while still maintaining
compatibility with Acrobat 4.0, apparently).

I bought a paper copy before I knew the book was available online, but
it's good enough that even had I known it was available online, I still

would have bought it on paper.

You're welcome to download it and keep it on hand -- it makes for
EXCELLENT reading:
http://www.alexburke.ca/asterisk-tfot.pdf

--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] how to tape letters in xlite

2006-02-21 Thread Bayrouni


Hello all,
How to tape letters in xlite softphone, when using the 
Directory application (or generally when is needed).


Thank you.
--
Bayrouni
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] commercial package for vertical services

2006-02-21 Thread Patrick Fortin


Hi

Are there any packages to implement vertical services in asterisk

commercial (or free)

Thanks

Patrick

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] What business IP phone to use

 I hear some 
 people praising the GXP2000 phones and I gotta wonder what 
 they are smokin (regardless of firmware revison) so I just 
 don't know who to believe anymore.

As one of those who's praised the GXP2000, I feel I should just add that
it's all relative *to the price point*. The GXP2000 is probably the best
phone I can get hold of at that price point (£70 or so) here in the UK. The
9133i is £80 + PoE injector (£14), which is quite a big increase in budget
on 20 or 30 phones.

 Is there any unbiased comparison of various phones and 
 features anywhere.

As the discussion about the GXP2000 showed, it's not really features that's
important - it's more a question of reliable firmware, build quality, etc.

If you're after one or two nice office phones, I don't think you can beat
getting 2nd hand Cisco 7960s off ebay, putting the latest SCCP firmware on
them and using them with chan_sccp. I've done that at 3 locations where I
spend lots of time, and I really like the feel of the 7960. I can't justify
the price of them new, but from auction, the prices are far more reasonable
(going rate seems to be about £110 in the UK).

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Outbound Routing does not use Multiple Trunks

2006-02-21 Thread Mimmus

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Nate List
 Sent: Tuesday, February 21, 2006 7:17 PM

 ...
 In my Outbound Routing I have the Trunk Sequence set up so that 0 is
 Zap/1-1 and 1 is ZAP/2-1  What I see is that when Trunk 
 Sequence 0 is full, it does not open Trunk Sequence 1.  

Peraphs this bug in AMP:

###
Max Channels Bug Remains. A bug has been reported because of a deprecated
command that makes [EMAIL PROTECTED]'s calculation of maximum channels invalid.
To fix it, goto AMP-Maintenance-Config
Edit-extensions.conf-macro-dialout-trunk and comment out line s,7 so that
it looks like this:

;exten = s,7,CheckGroup(${OUTMAXCHANS_${ARG1}})

Then insert a new line s,7 just below it which looks like this:

exten = s,7,GotoIf($[ ${GROUP_COUNT()}  ${OUTMAXCHANS_${ARG1}} ]?108)

Then click the Update button and reload Asterisk to activate the change.
###

[from http://mundy.org/blog/index.php?p=112]

Keep me informed if this solves your problem.

Bye
Mimmus

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] BAD/GOOD Echo Cancel


  I put a Tellabs 64ms echo canceller into my facility this weekend and 
  am praying that it removes are echo problem.  If it does, I plan on 
  making it a standard on my Asterisk installs that have a channel bank 
  or T1.
 
 
  Well, the day is almost over here and not one echo reported today.  
  Very impressive!  I had 5 more cards delivered today.
 
 
 Just as a follow up to this. 
 
 I purchased 5 Tellabs 64ms cards on ebaY.  The very 1st card that I put 
 into production worked quite nicely (Little residual echo, but very much 
 an improvement).  I was on site this weekend and figured I would test 
 out the remaining cards, one at a time to confirm they were in working 
 condition.  This upcoming Monday, I had many complaints of echo.  Went 
 back Tuesday morning and went over the settings of the previous card 
 compared to this replacement.  I turns out that only 1 of the 5 cards 
 had option 38 available (Send Side Echo Cencellation).
 
 So, if you are in the market for one of these cards because of local 
 echo, you'll want to confirm with the vendor of 'Send side Echo 
 Cancellation', or at least ask him to check for option 38 after powering 
 up. Out of the 5 purchased, only 1 had it.

I've never used a Tellabs before, but might try changing the input and
output analog lines around (eg, reverse it). Don't have a clue if that
would really work.


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] NEED COMMENT ON USING FEDORA CORE 3

2006-02-21 Thread ADEGOKE ARUNA


Dear all,

Can somebody share his experience with me in using fedora core 3 as asterisk
server using quad port card (e1/pri) at full capacity.

goksie


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] What business IP phone to use

2006-02-21 Thread Raymond McKay


For every person that says phone x is great there is
someone else complaining about it.


Its very simple why there are soo many answers to the what phone to use 
question.  The answer really comes down to a matter of personal preferance 
and end-users needs.  Mind you, some phones are better than others but the 
answer really comes down to what you plan on doing with the phones and the 
types of end-users using the phones.  With that said, here are my personal 
recommendations and why


1) SNOM 360/320:  If you are transintioning a small business from something 
similar to an Avaya partner system, these are the phones to use.  They are 
moderatly stable and support features that many end-users are used to such 
as Intercom, Line Indicators, MWI etc.  In the newest firmwares, you get the 
highest flexibility of soft button configuration of any phone in the market. 
Be sure to due some testing before implementing any new firmwares on thiese 
phones though.  SNOM has been less than stellar when it comes to testing new 
release versions.  Currently 5.3 seems to crash the phones regularly.  Other 
than that, they are a good solid phone, they look and feel like business 
telephones (something you can't say about many phones like the Grandstream 
and the like)  Team these up with some of the new low cost PoE options from 
Linksys and Netgear and you have yourself a great solution. The web based 
configuration file ability on these phones makes for interesting things you 
can do with PHP and dynamic config files. As the phones also support GSM, 
you can get arround having to buy G729 licenses when bandwidth is a concern. 
The best part is that the price is somewhat moderate on these phones.  Don't 
expect to beat out pricing on rock bottom systems with these phones, but as 
they say, you get what you pay for.


2) Polycom 301/501/601: Also a solid performer.  The 601 makes for a great 
attendant phone with the option of an expansion pack with LCD programmable 
labels for the soft buttons.  (great if you have a fluid office situation). 
I find the configuration files a bit more confusing and you'll have to use 
TFTP instead of HTTP with these precluding the use of dynamic PHP driven 
config files.  On the upside, Polycom support is much better than SNOM.  I 
get responses from them in a day wheras from SNOM it sometimes takes up to a 
week to get a question answered.  The prices on these cannot be beat for the 
functionality that they offer.  They also support many of the features like 
Line indication and Intercom.  Phone stability is quite high and there is a 
lesser problem with buggy firmware being released


3) Cisco 79XX:  A great phone and solid performer but it comes at a steep 
price.  I use these only in enviroments where end-users have worked with 
them before lowering training costs overall.  In those situations, the 
phones nearly sell themselves so long as people are willing to pay for the 
Cisco premium.  Other than their rock solid reliability, they really don't 
offer anything special unless you are in an enviroement that might use phone 
based XML applications


Now all of this is not to say that a sub $100 phone might not be the right 
choice for your situation.  For business phones though, I tend to follow 
this set of guidelines.


1) If it doesn't support PoE I won't implement it.  Support phones with 
wall-warts or bricks is just a added hassle and adds TCO as most end up 
being replaced once or twice during the lifetime of the phone when someone 
trips over them etc.  With PoE switches from linksys starting at $500, there 
is absolutely no reason not to consider them.


2) Autoconfiguration should be simple yet powerful and VERY well 
documented..  If you can't get the phone manufacturer to give you a manual 
on TFTP configuration or HTTP configuration that is clear and concise, it 
just isn't worth the effort of trying to figure it out yourself.


3) Stability, Stability, Stability.  People have gotten used to the fact 
that phone networks and systems rarely go down.  Telling someone their phone 
crashed usually gets you a funny look.  If a phone you are selecting crashes 
twice while you are testing, that is far too many time.  Heck, once it too 
many times.


4) Is the company going to be around tomorrow:  A lot of VoIP manufactures 
have come and gone, many more will come and go.  Stick to the bigger names. 
You'll end up paying more up front, but they will be around to support you 
in the future and at least you will be able to give your end-users an 
upgrade path that minimalizes the learning curve.  I.e. older SNOM phones 
work very similarly to the newer ones so when you upgrade say a Snom 190 to 
a 320/360, the user just needs to figure out where the buttons are now but 
otherwise feels they are on a same or similar phone.


These are my recommendations.  As with all such things, your mileage may 
vary.  I have sold and installed pretty much every kind of phone there is 
out

RE: [Asterisk-Users] Download Asterisk: The Future Of Telephony [More Info]

2006-02-21 Thread Alexander Burke


Hello, Bob!

At 01:32 PM 02/21/2006, you wrote:

Speaking of this book, where can I get it?  Obviously I can read the
pdf, but I lack the facility to print it in any usable fashion.  The
labor and materials that I have spent on trying to print it thus far
probably outweighs the cost of the silly thing.  Is it only available
online, or do you think Barnes and Noble, Borders, etc might have it?


Oh, I wouldn't print the whole thing; the price of the paper copy 
doesn't make it cost-effective to run one off... unless you happen to 
work at a place with a nice laser printer and a spiral-binding 
machine, I guess!


Any reputable book seller should be able to order it by its ISBN 
(0596009623). I bought my paper copy from Amazon, and had it in a 
week. It *is* a real book -- the PDF that was released is (most of) 
exactly what went to the book printing company -- the markings in the 
corners are alignment marks, and the vertical and horizontal lines in 
the margins are the cut marks for binding. The table of contents and 
index are missing, probably because they're fairly useless in a file 
you can do full-text searches on, and also probably to make 
counterfeiters actually have to do some work.


--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada 



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] NEED COMMENT ON USING FEDORA CORE 3


 Can somebody share his experience with me in using fedora core 3 as asterisk
 server using quad port card (e1/pri) at full capacity.

Runs fine and is very stable.  Full capacity is 100% dependent on exactly
what asterisk is doing (eg, transcoding), the PC hardware, etc, and has
nothing to do with fc3.



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] What business IP phone to use

I agree with most of Raymond's other points, but I have to take issue with
this one:

 1) If it doesn't support PoE I won't implement it.  Support 
 phones with wall-warts or bricks is just a added hassle and 
 adds TCO as most end up being replaced once or twice during 
 the lifetime of the phone when someone trips over them etc.  
 With PoE switches from linksys starting at $500, there is 
 absolutely no reason not to consider them.

That's one *bloody* expensive switch, considering a decent quality 24-port
10/100 switch can be had for £40 (say $70). It's very difficult to justify a
recommendation that a small business should pay over 7 times the price for a
PoE capable switch.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Uninstall Asterisk

2006-02-21 Thread Tom


I have a server in my lab running asterisk (v1.2.1)
and ztdummy. (No zaptel hardware is present in the
server).

I have to free up this server to be used for a
completely different application.

What is the best step-by-step procedure to permanently
remove/uninstall asterisk, asterisk-addons,
asterisk-sounds, and zaptel/ztdummy?
(I did not see much on the web regarding this. 
Google: uninstall asterisk site:lists.digium.com)

So far, I've done this...
rmmod ztdummy
rmmod zaptel
/etc/rc.d/init.d/asterisk stop
(and verified this with lsmod and ps -ef)

Thanks.

Tom

~~~

[EMAIL PROTECTED] ~]# df -k
Filesystem   1K-blocks  Used Available
Use% Mounted on
/dev/mapper/VolGroup00-LogVol00
507748111653369881 
24% /
/dev/sda1   101086  9186 86681 
10% /boot
none517900 0517900  
0% /dev/shm
/dev/sda5   25 10308233106  
5% /tmp
/dev/sda3  5052060   2223912   2571512 
47% /usr
/dev/sda2  9068648219620   8388368  
3% /var
[EMAIL PROTECTED] ~]#






__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] how to add stun functionality in asterisk

 What's the benefit of using stund vs nat=yes in your sip.conf 
 for that device?  I haven't had any issues behind firewalls 
 when I enable that option, and no ports are needed to be opened.

For some strange reason, even with nat=yes sometimes when a user's IP
changes, the phone doesn't realise it and sends another SIP refresh.
Asterisk promptly ignores it since there's no registration from that IP.

With stun, the phone realises it's IP has changed and sends an invite to
asterisk rather than a refresh.

I could be wrong, but using stun seems to have improved the nat-related
issues some of our customers have had with their home-based phones no end.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] good voip