Re: [Asterisk-Users] CID Number problem

[Peter Svensson]

On Sat, 30 Apr 2005, Ma Zhiyong wrote:

> I use TE405P as gateway and Eicon PRI card as fax card.
> 
> When I receive the caller number from PSTN, I found it was 51863500. While I 
> dial the FAX trunk, FaxGetty get the caller number 051863500.
> 
> -- Executing NoOp("Zap/124-1", "51863500") in new stack
> -- Executing Dial("Zap/1-1", "ZAP/g1/51863507") in new stack
> 
> Apr 30 13:30:50 faxserver FaxGetty[28254]: --> [33:RING CID: 051863500 DAD: 
> 51863507]

Gather a "pri intense debug span X" log. One possible cause is in the Type 
Of Number (TON) handling in Asterisk. What is the prilocaldialplan set to 
for the link to the Eicon PRI? Are you using any of the "nationalprefix" 
or similar options?

Peter


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[Asterisk-Users] Re: Can't get incoming calls with IAX trunks (FWD & Teliax)

[Iassen Hristov]

This question will be better addressed on the aah forums.

I would suggest:
1) have you setup a  DID?
2) take a look in the log file 
tail -f /var/log/asterisk/full
3) see the numerous threads on the aah forums about how to configure FWD
and Teliax (and other providers)

I personally have both FWD and Teliax working perfectly fine with aah.

Here are the relevant parts of my iax_additional.conf. Of course you should
replace USERID, FWDNUMBER and PASSWORD with yours. Of course you should not
change iax_additional.conf directly, but paste the different parts in AMP

-- excerpt from iax_additional.conf --
register=USERID:[EMAIL PROTECTED]
register=FWDNUMBER:[EMAIL PROTECTED]

[fwd]
username=FWDNUMBER
type=peer
secret=PASSWORD
qualify=yes
host=iax2.fwdnet.net
disallow=all
context=from-pstn
auth=md5
allow=ulaw

[fwd-in]
type=user
inkeys=freeworlddialup
disallow=all
context=from-pstn
auth=rsa
allow=ulaw

[teliax]
username=USERID
type=friend  ; one should think it has to be peer, but that does not work
secret=PASSWORD
host=voip.teliax.com
context=from-pstn
auth=md5

[teliax-in]
username=USERID
type=user
secret=PASSWORD
host=voip.teliax.com
context=from-pstn
auth=md5
-- end excerpt from iax_additional.conf --


> 
> Message: 28
> Date: Fri, 29 Apr 2005 21:03:39 -0700 (PDT)
> From: "Patrick Gray, Jr." <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Can't get incoming calls with IAX trunks
>   (FWD &  Teliax)
> To: asterisk-users@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=us-ascii
> 
> I have [EMAIL PROTECTED] 0.9 running, and everything seems
> to work well EXCEPT incoming calls.
> 
> I have an FWD and Teliax trunk (both using IAX), and a
> Cisco 7960 SIP phone connected to Asterisk. 
> Everything tests fine:
> 
> - Can call from softphone to Cisco and vice versa
> - Asterisk inbound simulation works like a champ
> - Voicemail works fine
> - Outbound calls to both trunks works fine
> 
> However, when I call into my system on the FWD or
> Teliax trunks, nothing happens.  Nothing appears on
> the asterisk console so I’m not even sure where to
> start.  I’m suspecting network problems, but don’t
> know what to look for.  My asterisk box sits on my
> LAN, behind an IPCop-based NAT router.  I’ve forwarded
> port 4569 UDP and TCP to the asterisk box, but still
> no joy.  I’ve googled and checked voip-info, but
> everything that mentions NAT as a potential problem
> points to IAX as the solution.  Trunk-wise, I’m pure
> IAX (only SIP is the 7960, and it’s on the same
> network as the asterisk box).
> 
> I’m pretty new to asterisk, so if you can dumb down
> any debugging advice I’d appreciate it.
> 
> Thanks a ton!
> 
> Pat
> 



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Re: [Asterisk-Users] Caller-ID Block

[lenz]

Hello,
from what I see, I guess they're only ways to insert a piece of speech  
without recording it; you could easily record the phrases yourself and add  
Playback()s instead.

BTW, I'd like to thank Tim for sharing his recipe with us. Anybody else's  
got a recipe to share? :-)
l.

In data Sat, 30 Apr 2005 01:11:45 -0400, Daniel Salama <[EMAIL PROTECTED]>  
ha scritto:

Tim,
This certainly looks interesting. I just have a question about the  
recipe: it makes reference to some AGI perl scripts. Is the source  
available? Or may be it's irrelevant.

Thanks,
Daniel
On Apr 29, 2005, at 9:10 PM, Tim Litwiller wrote:
Daniel Salama wrote:
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a  
list of callers to be blocked. When they call, they should hear busy  
and then we hang up. We have about 100 DIDs routed to different  
contexts and I wouldn't want to have to manually edit all contexts. Is  
there a way to do something global to create something like a black  
list of caller IDs to block?
Thanks,
Daniel
I used bits and pieces that I got from this list and from the wiki and  
made this up - I'm using it on aah - but it should be usable with  
slight modification on any asterisk install.

If you just want busy you'll have to edit the blacklisted1 macro  
currently it plays the SIT sound and then the this number is no longer  
in service message

It usually gets me taken off their call lists pretty quickly  :)
http://www.oinko.net/astrecipes/index.php?from=0&q=astrecipes/ 
how+to+blacklist+unwanted+callerid

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[Asterisk-Users] CID Number problem

[Ma Zhiyong]

Hi, all. I'm glad I put asterisk and hylafax 
together just like PSTN->Asterisk->Hylafax->Email. And the 
fax2email function works well.
But I also find some bugs about CID number.
 
I use TE405P as gateway and Eicon PRI card as fax 
card.
 
When I receive the caller number from PSTN, I found it was 
51863500. While I dial the FAX trunk, FaxGetty get the caller number 
051863500.
 
-- Executing NoOp("Zap/124-1", "51863500") in new stack-- 
Executing Dial("Zap/1-1", "ZAP/g1/51863507") in new stack
 
Apr 30 13:30:50 faxserver FaxGetty[28254]: --> 
[33:RING CID: 051863500 DAD: 51863507]
 
Any idea?
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Re: [Asterisk-Users] Asterisk Hardware Architecture Group

[Dinesh Nair]

On 04/30/05 02:42 Matt Roth said the following:
Does anyone have an interest in forming a hardware architecture group?
absolutely !
It seems that Asterisk is so tightly linked to specialized hardware and 
its corresponding architecture that developing the software alone is 
insufficient for its adoption to large scale applications.
yes, plus with the industry perception that PBXes are supposed to be up 
100% of the time (note, i said perception), having discussions on hardware 
vendors and architectures which allows us to achieve this is an excellent 
repository of knowledge.

--
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
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| done; done  |
+=+
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Re: [Asterisk-Users] Asterisk and sendmail

[Roger Hanson]

Bill Ford wrote:
Since all the asterisk program needs to do is send mail through smtp,
and since using sendmail for this purpose is a bit like using Jeff
Gordon's racing engine on a bicycle we opted to scrap sendmail and use
msmtp. This is basically just an smtp engine. To our mail server, it
looks just like any other smtp client. It's very easy to set up. After
you have it working from the command line, simply remove the sendmail
startup from init.d, rename the sendmail executable (I renamed mine
sendmail.orig, and it's in /usr/sbin). msmtp installs itself in
/usr/local/bin, so I created a link in the sbin directory to link
msmtp to the filename sendmail in sbin.
(ln -s /usr/local/bin/msmtp /usr/sbin/sendmail). That way whenever a
program calls sendmail, it is in reality calling msmtp. It works like
a charm.
Hope this helps
Bill
On 4/29/05, Chuck Keeter <[EMAIL PROTECTED]> wrote:
 

Hi all,
Can someone point me in the right direction to configuring sendmail to work
with Asterisk voicemail and faxes?
I did  a bit of research on the web but came up more confused that when I
started.
It's the basic setup I'm having trouble with, where to add the SMTP and
login and user name to sendmail to use a smart host in getting v-mail and
faxes to the people they are going to.
Thanks in advance.
Chuck Keeter
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I had the same problem and was advised to can sendmail and switch to 
postfix, which I've never had trouble with.

Here's how I did it::
Here is how you can switch MTAs and configure postfix (assumes WBEL is 
the distro):

0. Install redhat-switch-mail (as root, if you don't have it):
yum install redhat-switch-mail
1. Run (as root):
redhat-switch-mail
Select "Postfix"
2. Edit (with vi, as root) /etc/postfix/main.cf
such that:
mydomain = yourdomain.com
and:
myorigin = $mydomain
3. As root:
service postfix restart
Hope this helps.
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Re: [Asterisk-Users] Caller-ID Block

[Daniel Salama]

Tim,
This certainly looks interesting. I just have a question about the  
recipe: it makes reference to some AGI perl scripts. Is the source  
available? Or may be it's irrelevant.

Thanks,
Daniel
On Apr 29, 2005, at 9:10 PM, Tim Litwiller wrote:
Daniel Salama wrote:
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a  
list of callers to be blocked. When they call, they should hear busy  
and then we hang up. We have about 100 DIDs routed to different  
contexts and I wouldn't want to have to manually edit all contexts.  
Is there a way to do something global to create something like a  
black list of caller IDs to block?
Thanks,
Daniel
I used bits and pieces that I got from this list and from the wiki and  
made this up - I'm using it on aah - but it should be usable with  
slight modification on any asterisk install.

If you just want busy you'll have to edit the blacklisted1 macro  
currently it plays the SIT sound and then the this number is no longer  
in service message

It usually gets me taken off their call lists pretty quickly  :)
http://www.oinko.net/astrecipes/index.php?from=0&q=astrecipes/ 
how+to+blacklist+unwanted+callerid

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[Asterisk-Users] SER + Asterisk

[Deon]

I'm working with SER + Asterisk. I was told that to have SER push calls to
multiple Asterisk servers, I can use the LCR Module, I'll just give all
the Asterisk servers the same weight/price, and SER will randomly send
outbound requests to each Asterisk server. It's not truly equally
balanced, so one server could get more calls while the other has spare
resources. So although it does increase the number of simul. outbound
calls that can be made, it still doesn't make me feel good knowing it's
not perfectly load balanced. Can somebody help elaborate? Is there a
better way to get SER to evenly balance between the Asterisk servers? 

I wonder if I use Asterisk's ability to limit the number of simult. calls,
if Asterisk gets more than, lets say 300, calls, then it would reject
calls, I wonder if SER would then try sending it to the other Asterisk
server, which may have available channels. 

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[Asterisk-Users] Call routing

[Daniel Salama]

I have two asterisk boxes connected using IAX. There are two T1s on 
each box. I have all my dialing rules in one of the asterisk boxes and 
all of my agents register on the same box where I have all the dialing 
rules. See diagram below:

Asterisk_1 <--2xT1--> PSTN
||
||
Asterisk_2 <--2xT1--> PSTN
||
||
SIP_Agents
I'm wondering how can I configure extensions.conf in Asterisk_1 so that 
EVERY incoming call (regardless of DID or CallerID or whatever) 
received from PSTN in Asterisk_1 is routed via IAX to Asterisk_2? 
Basically, anything that arrives from Zap/g1 or Zap/g2 in Asterisk_1 
should be automatically routed to Asterisk_2 preserving all call 
features, such as DID, CallerID, etc.

Any ideas?
Thanks,
Daniel
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[Asterisk-Users] CallerID on cell phone

[Lee Lee]

Hi All
 
I am using asterisk to redirect some extension calls to few cell phones.
 
I was wondering if it is possible to have * display on the cell phone as 'PRIVATE NUMEBR' or 'CALLS' instead of the calling number.
 
 
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Re: [Asterisk-Users] Caller-ID Block

[Daniel Salama]

What's what I'm trying to avoid. To answer your question: I have TE4XXP 
with T1s (not PRIs). What I want to do is block it based on the 
caller-id and not the DID Number. That way, I don't have to write 100+ 
lines.

Thanks,
Daniel
On Apr 29, 2005, at 6:23 PM, Stefan Gofferje wrote:
Daniel Salama schrieb:
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a 
list of callers to be blocked. When they call, they should hear busy 
and then we hang up. We have about 100 DIDs routed to different 
contexts and I wouldn't want to have to manually edit all contexts. 
Is there a way to do something global to create something like a 
black list of caller IDs to block?
Depends on what channel / hardware you are using...
try exten => /,1,Busy
regards,
Stefan
--
(o_   Stefan Gofferje  | Linux Systems Specialist
//\   Reg'd Linux User #247167 | Network Security Specialist
V_/_  Heckler & Koch - the original point and click interface
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asterisk-users@lists.digium.com

[Dan Levine]

Are you sure it's registering?

- 
Dan Levine 
CYTEXONE | Your Technology Specialists 
t: 877.CYTEXONE x 810 
l: 212.477.0990 x 810 
e: [EMAIL PROTECTED] 
http://www.cytexone.com 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Gray, Jr.
Sent: Saturday, April 30, 2005 12:04 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD
&Teliax)

I have [EMAIL PROTECTED] 0.9 running, and everything seems
to work well EXCEPT incoming calls.

I have an FWD and Teliax trunk (both using IAX), and a
Cisco 7960 SIP phone connected to Asterisk. 
Everything tests fine:

-   Can call from softphone to Cisco and vice versa
-   Asterisk inbound simulation works like a champ
-   Voicemail works fine
-   Outbound calls to both trunks works fine

However, when I call into my system on the FWD or
Teliax trunks, nothing happens.  Nothing appears on
the asterisk console so I'm not even sure where to
start.  I'm suspecting network problems, but don't
know what to look for.  My asterisk box sits on my
LAN, behind an IPCop-based NAT router.  I've forwarded
port 4569 UDP and TCP to the asterisk box, but still
no joy.  I've googled and checked voip-info, but
everything that mentions NAT as a potential problem
points to IAX as the solution.  Trunk-wise, I'm pure
IAX (only SIP is the 7960, and it's on the same
network as the asterisk box).

I'm pretty new to asterisk, so if you can dumb down
any debugging advice I'd appreciate it.

Thanks a ton!

Pat

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Re: [Asterisk-Users] Asterisk and sendmail

[Bill Ford]

Since all the asterisk program needs to do is send mail through smtp,
and since using sendmail for this purpose is a bit like using Jeff
Gordon's racing engine on a bicycle we opted to scrap sendmail and use
msmtp. This is basically just an smtp engine. To our mail server, it
looks just like any other smtp client. It's very easy to set up. After
you have it working from the command line, simply remove the sendmail
startup from init.d, rename the sendmail executable (I renamed mine
sendmail.orig, and it's in /usr/sbin). msmtp installs itself in
/usr/local/bin, so I created a link in the sbin directory to link
msmtp to the filename sendmail in sbin.
(ln -s /usr/local/bin/msmtp /usr/sbin/sendmail). That way whenever a
program calls sendmail, it is in reality calling msmtp. It works like
a charm.

Hope this helps
Bill

On 4/29/05, Chuck Keeter <[EMAIL PROTECTED]> wrote:
> Hi all,
> 
> Can someone point me in the right direction to configuring sendmail to work
> with Asterisk voicemail and faxes?
> 
> I did  a bit of research on the web but came up more confused that when I
> started.
> 
> It's the basic setup I'm having trouble with, where to add the SMTP and
> login and user name to sendmail to use a smart host in getting v-mail and
> faxes to the people they are going to.
> 
> Thanks in advance.
> 
> Chuck Keeter
> 
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asterisk-users@lists.digium.com

[Henry Devito]

From the CLI if you do a iax2 show registry, does it show you registered? 
Maybe you can post the parts of your config that pertains to your question?
- Original Message - 
From: "Patrick Gray, Jr." <[EMAIL PROTECTED]>
To: 
Sent: Friday, April 29, 2005 11:03 PM
Subject: [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD 
&Teliax)


I have [EMAIL PROTECTED] 0.9 running, and everything seems
to work well EXCEPT incoming calls.
I have an FWD and Teliax trunk (both using IAX), and a
Cisco 7960 SIP phone connected to Asterisk.
Everything tests fine:
- Can call from softphone to Cisco and vice versa
- Asterisk inbound simulation works like a champ
- Voicemail works fine
- Outbound calls to both trunks works fine
However, when I call into my system on the FWD or
Teliax trunks, nothing happens.  Nothing appears on
the asterisk console so I'm not even sure where to
start.  I'm suspecting network problems, but don't
know what to look for.  My asterisk box sits on my
LAN, behind an IPCop-based NAT router.  I've forwarded
port 4569 UDP and TCP to the asterisk box, but still
no joy.  I've googled and checked voip-info, but
everything that mentions NAT as a potential problem
points to IAX as the solution.  Trunk-wise, I'm pure
IAX (only SIP is the 7960, and it's on the same
network as the asterisk box).
I'm pretty new to asterisk, so if you can dumb down
any debugging advice I'd appreciate it.
Thanks a ton!
Pat
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[Asterisk-Users] Can't get incoming calls with IAX trunks (FWD & Teliax)

[Patrick Gray, Jr.]

I have [EMAIL PROTECTED] 0.9 running, and everything seems
to work well EXCEPT incoming calls.

I have an FWD and Teliax trunk (both using IAX), and a
Cisco 7960 SIP phone connected to Asterisk. 
Everything tests fine:

-   Can call from softphone to Cisco and vice versa
-   Asterisk inbound simulation works like a champ
-   Voicemail works fine
-   Outbound calls to both trunks works fine

However, when I call into my system on the FWD or
Teliax trunks, nothing happens.  Nothing appears on
the asterisk console so I’m not even sure where to
start.  I’m suspecting network problems, but don’t
know what to look for.  My asterisk box sits on my
LAN, behind an IPCop-based NAT router.  I’ve forwarded
port 4569 UDP and TCP to the asterisk box, but still
no joy.  I’ve googled and checked voip-info, but
everything that mentions NAT as a potential problem
points to IAX as the solution.  Trunk-wise, I’m pure
IAX (only SIP is the 7960, and it’s on the same
network as the asterisk box).

I’m pretty new to asterisk, so if you can dumb down
any debugging advice I’d appreciate it.

Thanks a ton!

Pat

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[Asterisk-Users] Asterisk and sendmail

[Chuck Keeter]

Hi all,
Can someone point me in the right direction to configuring sendmail to work 
with Asterisk voicemail and faxes?

I did  a bit of research on the web but came up more confused that when I 
started.

It's the basic setup I'm having trouble with, where to add the SMTP and 
login and user name to sendmail to use a smart host in getting v-mail and 
faxes to the people they are going to.

Thanks in advance.
Chuck Keeter
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Re: [Asterisk-Users] Pattern Matching

[Michael D Schelin]

Hey Mojo, I'm thinking you might try using priorty 's to set some kind 
routing. just a thought..


Mojo Jojo wrote:
We recently had our PRI installed, we currently have 100 toll-free's 
pointing to it.

I have almost everything working great but..
I have setup the first few numbers we want to use coming in from the 
PRI and they work great, but..

What I want to do is setup an extension with pattern matching to 
answer for any numbers called that are pointed to our system and PRI 
but not yet in use/configured.

I have been successful at setting up pattern matching as a catch all 
for 98 or so numbers not in use yet and I have been successful setting 
up the 2 numbers I want to make use of for now.

Problem is, I can't use both at the same time!
If I turn on the pattern matching then my greeting plays for the 
configured number, then the message plays for the invalid number 
(basically executing the extension with the pattern matching).

I have read about sorting with pattern matching by using an include, I 
did this but it's not really helping.

I have set a response timeout after the first extension plays it's 
greeting, I would think it should wait until it times out but it 
doesn't, it just immediately moves to the pattern matched extension.

I must be missing something big here..
Any help is appreciated..
--
Private Label Wholesale Internet Access!
http://www.YourOwnISP.com
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Re: [Asterisk-Users] Need info : lspci

[Dan Perik]

Clone here as well.

:00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface

- DAn



Marco Supino wrote:

> Hi,
>
> I need some info from people with the x100p card (digium or clone),
> please send me the output of "lspci" and "lspci -n" from your linux
> machine, i am tring to find out something on my * server.
>
> Thanks.
>
> Marco.
>
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Re: [Asterisk-Users] Caller-ID Block

[Tim Litwiller]

Daniel Salama wrote:
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a list 
of callers to be blocked. When they call, they should hear busy and then 
we hang up. We have about 100 DIDs routed to different contexts and I 
wouldn't want to have to manually edit all contexts. Is there a way to 
do something global to create something like a black list of caller IDs 
to block?

Thanks,
Daniel
I used bits and pieces that I got from this list and from the wiki and 
made this up - I'm using it on aah - but it should be usable with slight 
modification on any asterisk install.

If you just want busy you'll have to edit the blacklisted1 macro 
currently it plays the SIT sound and then the this number is no longer 
in service message

It usually gets me taken off their call lists pretty quickly  :)
http://www.oinko.net/astrecipes/index.php?from=0&q=astrecipes/how+to+blacklist+unwanted+callerid
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[Asterisk-Users] Polycom IP500 Ringer Volume

[John Harrison]

Hello:

 

I have searched everywhere in this list but cannot find the
.cfg file (ipmid.cfg) entry to set the initial ringer volume for an IP500.  

 

Could someone please post the XML attribute and value to set
the ringer value, to say its maximum upon the phone’s restart.

 

 

THANKS IN ADVANCE!

 

John
Harrison
Cincinnati, OH






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Re: [Asterisk-Users] UTSTARCOM Wifi handset?

[Henry Devito]

You mention the WIP-5000,  Does that handset have the ability to receive 
text messaging/instant messaging?
- Original Message - 
From: "Michael Graves" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, April 29, 2005 4:14 PM
Subject: [Asterisk-Users] UTSTARCOM Wifi handset?


Anyone use these with *? I'm curious to know how they compare to the
Hitachi WIP-5000?
Michael
--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]
o713-861-4005
o800-905-6412
c713-201-1262

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Re: [Asterisk-Users] Need info : lspci

[Mojo with Horan & Company, LLC]

Marco, I've got a clone.  X101P I think it was sold to me as.
$ lspci
<...>
00:08.0 Communication controller: Tiger Jet Network Inc. Intel 537
<...>
$ lspci -n
<...>
00:08.0 Class 0780: e159:0001
<...>
Mojo
Marco Supino wrote:
Hi,
I need some info from people with the x100p card (digium or clone), 
please send me the output of "lspci" and "lspci -n" from your linux 
machine, i am tring to find out something on my * server.

Thanks.
Marco.
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Re: [Asterisk-Users] Need info : lspci

[Derek Whitten]

00:0e.0 Communication controller: Individual Computers - Jens Schoenfeld
Intel 537

00:0e.0 Class 0780: e159:0001





On Fri, 2005-04-29 at 16:26, Marco Supino wrote:
> Hi,
> 
> I need some info from people with the x100p card (digium or clone), 
> please send me the output of "lspci" and "lspci -n" from your linux 
> machine, i am tring to find out something on my * server.
> 
> Thanks.
> 
> Marco.
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kFuQ Productions


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[Asterisk-Users] Need info : lspci

[Marco Supino]

Hi,
I need some info from people with the x100p card (digium or clone), 
please send me the output of "lspci" and "lspci -n" from your linux 
machine, i am tring to find out something on my * server.

Thanks.
Marco.
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RE: [Asterisk-Users] Asterisk Hardware Architecture Group

[Race Vanderdecken]

Sounds like a good idea to me. I would watch it.

Race Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth
Sent: Friday, April 29, 2005 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk Hardware Architecture Group

List members,

Does anyone have an interest in forming a hardware architecture group?

It seems that Asterisk is so tightly linked to specialized hardware and
its corresponding architecture that developing the software alone is
insufficient for its adoption to large scale applications.

Thank you,

Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debia
n
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Re: [Asterisk-Users] Asterisk Hardware Architecture Group

[John Todd]

At 4:57 PM -0400 on 4/29/05, Daniel Salama wrote:
On Apr 29, 2005, at 4:22 PM, snacktime wrote:
Personally I would buy an * box from someone like asaservers.com.  At
least companies like that really know their hardware, and if you tell
them the common issues with * they could probably put together a rock
solid system.
Chris
Does anyone have any experience with servers from 
siliconmechanics.com? Are they reliable? How does * run on them?

Thanks
- Daniel

I have had extensive experience with Silicon Mechanics machines, and 
I have had nothing but the best interactions with the company and 
their products.  Pricing has been decent, though I'm sure if I was 
shopping for el-cheapo hardware I could get better.  However, I base 
my purchasing decisions on quality of equipment and service, and 
their price is certainly what I consider "inexpensive" when I balance 
it against those two other criteria.

Sometimes they're a bit slow on shipping, as they build a lot of 
their gear to order, but I can typically wait a week or so.  I've 
ordered probably 30 or 40 systems from them in varying 
configurations, and I've not had a return yet.

I don't know if they actually ship anything other than SuperMicro 
motherboards, but I'd suggest specifying them in your order.  I've 
had very good luck with those MB's in my Asterisk platforms.  Plus, 
their on-line pricing and configuration tool really makes things easy 
to get a price quote without dealing with salespeople (not that their 
salespeople are bad, but it's just time-consuming trying to get 
someone on the phone.)

This thread should probably move over to asterisk-biz if it's going 
to be extended...

JT
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RE: [Asterisk-Users] * and Sipgate (UK)

[Paul Tyreman]

What are you using instead of SIPGATE in the UK ?

I also have this problem with DTMF tones not being passed to Asterisk from a 
PSTN line and my e-mails are being ignored too !

If only they sorted that problem out, it would be a great service.

Thanks, Paul.


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Rafal 
Kaniewski
Posted At: 29 April 2005 22:35
Posted To: Asterisk-Users
Conversation: [Asterisk-Users] * and Sipgate (UK)
Subject: RE: [Asterisk-Users] * and Sipgate (UK)

I gave up with sipgate after dtmp tone recognition didn't work - and found 
other who also have this problem and emails to sipgate are ignored..

Rafal Kaniewski
 


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[Asterisk-Users] Any workaround for long DISA timeout before it actually dials ?

[Robert Rozman]

Hi,
I'm finding long timeout before DISA really calls extension user entered 
annoying. I wonder what workarounds are you using for this ?

Playtones is one possibility , but it won't stop when user starts entering 
numbers...

Regards,
Rob.
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Re: [Asterisk-Users] Problems with TDM400P card

[John Novack]

I too wish I had a solution.
What I REALLY wish is that Digium would acknowledge that there is a 
whole bunch of problems, firstly with the card and MANY motherboards, 
then with reported problems some have with the FXO, either card or 
drivers? and FXS problems as well, again with the card and drivers.
Support is no support.
My ( and some others ) peculiar requirements make the use of many other 
devices difficult at best, and perhaps Digium wouldn't solve those, but 
this business of many having to go through a stable of motherboards to 
find one that works really makes it difficult to suggest using this card 
to others.

John Novack
GEOFFREY SACHS wrote:
I would also be interested in alternatives to the Tdm400p. I have had endless 
problems with a tdm400p card not being able to get the zttest numbers above 
99.975 and as a result not being able eliminate an intermitent but consistent 
echo.I have tried to date 4 different motherboard and hardware combinations as 
well as different linux versions to no avial.I would welcome some feedback on 
this.
oter   Geoffrey Sachs
- Original Message -
From: Anton Krall <[EMAIL PROTECTED]>
Date: Friday, April 29, 2005 11:54 am
Subject: RE: [Asterisk-Users] Problems with TDM400P card
 

Which card do you recommend using instead of the tdm400p? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[EMAIL PROTECTED] On Behalf Of 
|John Novack
|Sent: Viernes, 29 de Abril de 2005 09:19 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Problems with TDM400P card
|
|Anton Krall wrote:
|> Zttool shows nothing inside thebox.
|I have had similar problems with a TDM400 and CERTAIN 
|Motherboards which are PCI 2.2 but the TDM400 is not seen, in 
|my case, AT ALL The one I have reports it as an E/F but the 
|silk-screen clearly says H, Digium contends there is no 
|problem with the card, the reporting of different version 
|numbers is a "known bug" but doesn't prevent the card from 
|working, and because I can place it in another machine and get 
|it working, they refuse to acknowledge there is any defect in 
|the board.
|
|Perhaps a different motherboard? That is Digium's answer.
|Just keep going through hardware that otherwise meets 
|published specs until you find one that works.
|
|I have to conclude that, due to Digiums refusal to acknowledge 
|there are problems with the design, ( and there are more I 
|won't bore you with ) and no willingness to address the issues 
|that have been raised on this list time and time again, that 
|the TDM400 should be considered a "card of last resort" when 
|absolutely nothing else will work.
|Seems their IAXy falls into that same classification.
|Can't say about their T1/E1 cards
|
|JMO
|
|John Novack
   

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[Asterisk-Users] Caller-ID Block

[Daniel Salama]

Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a list 
of callers to be blocked. When they call, they should hear busy and 
then we hang up. We have about 100 DIDs routed to different contexts 
and I wouldn't want to have to manually edit all contexts. Is there a 
way to do something global to create something like a black list of 
caller IDs to block?

Thanks,
Daniel
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Re: [Asterisk-Users] Paging and intercom

[Henry Devito]

- Original Message - 
From: "Jacob Cazzell" <[EMAIL PROTECTED]>
To: 
Sent: Friday, April 29, 2005 4:21 PM
Subject: [Asterisk-Users] Paging and intercom


On our existing phone system, if you dial an
extention the other end will beep and then setup an intercom channel
that's hands free for the called station.  I'm not sure how this would
be duplicated in *, or is it more of a function of the phone we use?
Yes it is a function of which IP phone you use!

We also have an overhead paging system, our current system is tied
into a Valcom 3-zone paging system.  Would * support this paging
system?  How do you get a connection to it, an analog port?
You are close,  I use an analog port(ATA) attached to a Viking page adapter. 
This answers the analog port and opens talk path so a page can be made.  I 
do use valcom self amplified speakers with this setup

Henry 

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Re: [Asterisk-Users] Bouncing DTMF?

[Henry Devito]

- Original Message - 
From: "Jan Johansson" <[EMAIL PROTECTED]>
I seem to get "bounces" on DTMF.

For instance, if I turn on debug, and I dial the voicemail, and >enter 1234

as extension, I see in the logs "12234" "111234" "12344" and so >on, same

with passwords.
What type of phone SIP or analog? What is your DTMF type set for?
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RE: [Asterisk-Users] * and Sipgate (UK)

[Rafal Kaniewski]

I gave up with sipgate after dtmp tone recognition didn’t work - and
found other who also have this problem and emails to sipgate are
ignored..


Rafal Kaniewski


>-Original Message-
>From: [EMAIL PROTECTED] [mailto:asterisk-users-
>[EMAIL PROTECTED] On Behalf Of Robert P. McKenzie
>Sent: 28 April 2005 09:44
>To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] * and Sipgate (UK)
>
>Luki wrote:
>> Robert,
>>
>> It looks like you're dialing 447733322998, 44 for UK, then the area
>> code, etc. I have sipgate.de setup to dial local numbers (any German
>> number) as 0+AREA CODE+NUMBER. Always dial the area code, even if you
>> sipgate number is in the same city. For international numbers you
need
>> to dial 00+COUNTRY CODE+AREA CODE+NUMBER. I think similar rules apply
>> for sipate.co.uk, so try dialing the above as: 07733322998 or
>> 00447733322998.
>
>Doh, I had tried several combinations of dailing, however I didn't try
>just 077xxx that worked fine.  I thought it was the way I was
>dailing as other ways I'd tried had failed.  Thats got it working.
>Thanks for the wake up :)
>
>> Besides that, maybe a stupid question, but do you have money in your
>> sipgate account?
>
>Yeah :)
>
>--
>Robert P. McKenzie |   GammaRay Technical Services Ltd
>[EMAIL PROTECTED] | [EMAIL PROTECTED]
>http://www.uk-experience.com   |  http://www.gammaray-tech.com
>
>Ecademy Profile:   http://www.ecademy.com/account.php?op=view&id=64014
>
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RE: [Asterisk-Users] Paging and intercom

[Alexander Lopez]

Polycom phones and Snom phones supoprt paging. 

As far as your Overhead paging all you need is an FXO port on your
system. The * system will work perfectly with this. Even allowing the
zones to be set from the dialplan so your users won't need to learn any
new 'paging codes'

Email me off -list of you need some help. 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jacob
Cazzell
Sent: Friday, April 29, 2005 5:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Paging and intercom

Hello all,

We are considering implementing a new system based on Asterisk on the
back end.  I am very intrigued by the IP phones, but I have two
questions regarding paging and intercom functions.

I know that * supports these functions, but I'm not sure I fully
understand how.  On our existing phone system, if you dial an extention
the other end will beep and then setup an intercom channel that's hands
free for the called station.  I'm not sure how this would be duplicated
in *, or is it more of a function of the phone we use?

We also have an overhead paging system, our current system is tied into
a Valcom 3-zone paging system.  Would * support this paging system?  How
do you get a connection to it, an analog port?

These are probably my two biggest hurdles to overcome and I need some
pointers on how to implement or where to research my options.

Thanks!
Jacob
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RE: [Asterisk-Users] Paging and intercom

[Dean Collins]

Jacob, all of these questions have been answered numerous times before,
please search the archives.

BTW the cheapest way to set up a fxs paging is by modifying a
grandstream bt101 with auto answer per zone.

Cheers,
Dean


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jacob Cazzell
> Sent: Friday, April 29, 2005 5:21 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Paging and intercom
> 
> Hello all,
> 
> We are considering implementing a new system based on Asterisk on the
> back end.  I am very intrigued by the IP phones, but I have two
> questions regarding paging and intercom functions.
> 
> I know that * supports these functions, but I'm not sure I fully
> understand how.  On our existing phone system, if you dial an
> extention the other end will beep and then setup an intercom channel
> that's hands free for the called station.  I'm not sure how this would
> be duplicated in *, or is it more of a function of the phone we use?
> 
> We also have an overhead paging system, our current system is tied
> into a Valcom 3-zone paging system.  Would * support this paging
> system?  How do you get a connection to it, an analog port?
> 
> These are probably my two biggest hurdles to overcome and I need some
> pointers on how to implement or where to research my options.
> 
> Thanks!
> Jacob
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[Asterisk-Users] Paging and intercom

[Jacob Cazzell]

Hello all,

We are considering implementing a new system based on Asterisk on the
back end.  I am very intrigued by the IP phones, but I have two
questions regarding paging and intercom functions.

I know that * supports these functions, but I'm not sure I fully
understand how.  On our existing phone system, if you dial an
extention the other end will beep and then setup an intercom channel
that's hands free for the called station.  I'm not sure how this would
be duplicated in *, or is it more of a function of the phone we use?

We also have an overhead paging system, our current system is tied
into a Valcom 3-zone paging system.  Would * support this paging
system?  How do you get a connection to it, an analog port?

These are probably my two biggest hurdles to overcome and I need some
pointers on how to implement or where to research my options.

Thanks!
Jacob
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[Asterisk-Users] UTSTARCOM Wifi handset?

[Michael Graves]

Anyone use these with *? I'm curious to know how they compare to the
Hitachi WIP-5000?

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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Re: [Asterisk-Users] Bouncing DTMF?

[Brian McSpadden]

I've seen this happen once or twice before. Both times, different
things fixed it.

On one of them, we tweaked on the echo canceller settings, and on the
other, I believe we tweaked on the rxgain/txgain settings.

On 4/29/05, Jan Johansson <[EMAIL PROTECTED]> wrote:
> I seem to get "bounces" on DTMF.
> 
> For instance, if I turn on debug, and I dial the voicemail, and enter 1234
> as extension, I see in the logs "12234" "111234" "12344" and so on, same
> with passwords.
> 
> But dialing an extension never seem to fail this way.. Any hints?
> 
> 
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Re: RE: [Asterisk-Users] Problems with TDM400P card

[GEOFFREY SACHS]

I would also be interested in alternatives to the Tdm400p. I have had endless 
problems with a tdm400p card not being able to get the zttest numbers above 
99.975 and as a result not being able eliminate an intermitent but consistent 
echo.I have tried to date 4 different motherboard and hardware combinations as 
well as different linux versions to no avial.I would welcome some feedback on 
this.
   Geoffrey Sachs

- Original Message -
From: Anton Krall <[EMAIL PROTECTED]>
Date: Friday, April 29, 2005 11:54 am
Subject: RE: [Asterisk-Users] Problems with TDM400P card

> Which card do you recommend using instead of the tdm400p? 
> 
> |-Original Message-
> |From: [EMAIL PROTECTED] 
> |[EMAIL PROTECTED] On Behalf Of 
> |John Novack
> |Sent: Viernes, 29 de Abril de 2005 09:19 a.m.
> |To: Asterisk Users Mailing List - Non-Commercial Discussion
> |Subject: Re: [Asterisk-Users] Problems with TDM400P card
> |
> |Anton Krall wrote:
> |> Zttool shows nothing inside thebox.
> |I have had similar problems with a TDM400 and CERTAIN 
> |Motherboards which are PCI 2.2 but the TDM400 is not seen, in 
> |my case, AT ALL The one I have reports it as an E/F but the 
> |silk-screen clearly says H, Digium contends there is no 
> |problem with the card, the reporting of different version 
> |numbers is a "known bug" but doesn't prevent the card from 
> |working, and because I can place it in another machine and get 
> |it working, they refuse to acknowledge there is any defect in 
> |the board.
> |
> |Perhaps a different motherboard? That is Digium's answer.
> |Just keep going through hardware that otherwise meets 
> |published specs until you find one that works.
> |
> |I have to conclude that, due to Digiums refusal to acknowledge 
> |there are problems with the design, ( and there are more I 
> |won't bore you with ) and no willingness to address the issues 
> |that have been raised on this list time and time again, that 
> |the TDM400 should be considered a "card of last resort" when 
> |absolutely nothing else will work.
> |Seems their IAXy falls into that same classification.
> |Can't say about their T1/E1 cards
> |
> |JMO
> |
> |John Novack
> |
> |
> |> 
> |> I tried removing the x100 cards, moving the tdm card around, 
> |disabled 
> |> all usb and unnecessary stuff still, kudzu when booting up 
> shows the 
> |> card and the card shows up on /etc/sysconfig/hwconf but I 
> wonder why 
> |> it shows 2 of these and I only have 1 tdm400p card with 1 module
> |> 
> |> class: MODEM
> |> bus: PCI
> |> detached: 1
> |> driver: hisax
> |> desc: "Tiger Jet Network Inc.|Intel 537"
> |> vendorId: e159
> |> deviceId: 0001
> |> subVendorId: 8086
> |> subDeviceId: 0003
> |> pciType: 1
> |> -
> |> class: MODEM
> |> bus: PCI
> |> detached: 1
> |> driver: hisax
> |> desc: "Tiger Jet Network Inc.|Intel 537"
> |> vendorId: e159
> |> deviceId: 0001
> |> subVendorId: 8086
> |> subDeviceId: 0003
> |> pciType: 1
> |> 
> |> Still, interrupts doesn't show the card
> |> 
> |> [EMAIL PROTECTED] root]# cat /proc/interrupts
> |>CPU0
> |>   0:3994353  XT-PIC  timer
> |>   1:  2  XT-PIC  keyboard
> |>   2:  0  XT-PIC  cascade
> |>   8:  1  XT-PIC  rtc
> |>  10:  95510  XT-PIC  eth0
> |>  14: 129871  XT-PIC  ide0
> |> NMI:  0
> |> ERR:  0
> |> 
> |> And when trying to load the drvier
> |> 
> |> [EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# 
> |> modprobe wctdm
> |> /lib/modules/2.4.20-8/misc/wctdm.o: init_module: No such device
> |> Hint: insmod errors can be caused by incorrect module 
> parameters, 
> |> including invalid IO or IRQ parameters.
> |>   You may find more information in syslog or the output 
> |from dmesg
> |> /lib/modules/2.4.20-8/misc/wctdm.o: insmod 
> |> /lib/modules/2.4.20-8/misc/wctdm.o failed
> |> /lib/modules/2.4.20-8/misc/wctdm.o: insmod wctdm failed You 
> have new 
> |> mail in /var/spool/mail/root
> |> 
> |> I tried using diff. modules with no luck.,. Could be the 
> |mobo itself?
> |> 
> |>  
> |> 
> |> |-Original Message-
> |> |From: [EMAIL PROTECTED]
> |> |[EMAIL PROTECTED] On Behalf Of Rich 
> |> |Adamson
> |> |Sent: Viernes, 29 de Abril de 2005 08:59 a.m.
> |> |To: Asterisk Users Mailing List - Non-Commercial Discussion
> |> |Subject: Re: [Asterisk-Users] Problems with TDM400P card
> |> |
> |> |> I have a problem getting a TDM400P card to go.
> |> |> 
> |> |> It has 4 FXS ports (green modules) and I get this error:
> |> |> 
> |> |> [EMAIL PROTECTED] root]# ztcfg -v
> |> |> 
> |> |> Zaptel Configuration
> |> |> ==
> |> |> 
> |> |> 
> |> |> Channel map:
> |> |> 
> |> |> Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: 
> FXS 
> |> |> Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart 
> |> |> (Default)
> |> |> (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 
> |> |04) Channel
> |> |> 05: FXO Kewlstart

Re: [Asterisk-Users] Asterisk Hardware Architecture Group

[Daniel Salama]

Does anyone have any experience with servers from siliconmechanics.com? 
Are they reliable? How does * run on them?

Thanks
- Daniel
On Apr 29, 2005, at 4:22 PM, snacktime wrote:
Personally I would buy an * box from someone like asaservers.com.  At
least companies like that really know their hardware, and if you tell
them the common issues with * they could probably put together a rock
solid system.
Chris
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[Asterisk-Users] txfax and Ghostscript 8.51

[Me]

Hi all,

I'm trying to use spandsp and asterisk to send faxes. To do so I am
creating tiffs with Ghostscript. When I use Ghostscript 6.50 it seems
to work fine, but when I create the tiff using Ghostscript 8.51 (or
7.06) txfax garbles the tiff and it comes through all messed up. 
First of all is this a known problem or is it just me. More
importantly does anyone know of a way to fix this, I'd like to use
8.51 instead of 6.50.

By the way, if it makes a differnece i'm currently running
[EMAIL PROTECTED] but I've encountered the same problem with all the other
asterisk builds i've tried

thanks
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[Asterisk-Users] More TDM questions....

[Robert Webb]

Ok,

 So I am trying to still figure out my ringing issues. This time I
grabbed the butt set I own and hooked it into my pots line. With the
butt set in monitor mode, I called the pots line so I could actual hear
the AC ring. It was a low frequency ringing sound like I am accustomed
to.

I then hooked same butt set to the TDM and initiated ringing from a SIP
extension. I heard the data for the caller ID come across then I heard a
much higher pitched ringing sound. Almost like, even though the TDM
setup has been verified at 20 HZ, it is ringing at a much higher
frequency.


Any ideas?? If someone wants, I will try and do a recording of both
sounds so you can hear for yourself..

Robert




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Re: [Asterisk-Users] chan_zap graceful failure

[Andrew Kohlsmith]

On April 29, 2005 02:54 pm, Jeb Campbell wrote:
> I agree that it should be a very loud error (and possibly repeated
> notifications on the console).  But I also think that it should be able
> to limp along.  What would you think of a commercial phone system that
> completely dies when one port dies?

That's a very good point.

> I appreciate your points (thats why I wanted to discuss this), maybe we
> could satisfy both with an option in zapata.conf (keepgoing=yes)?

hehe I am not the biggest fan of allowing the system to do absolutely 
everything by "just adding an option" -- it makes the code behind the app 
hideous.

This is basically a little different that a port going bad -- this is the 
channel driver failing.  I would imagine plenty of phone systems will not 
fail nicely if a core module dies; this is akin to what chan_zap is doing.  A 
channel dying (phone unplugged, T1 red alarm) ... neither will cause asterisk 
to die.

-A.
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Re: [Asterisk-Users] Asterisk Hardware Architecture Group

[snacktime]

On 4/29/05, Daniel Salama <[EMAIL PROTECTED]> wrote:
> I think that would be a great idea. The only problem I see is that
> Asterisk is growing its feature set and maturing at such a dynamic
> rate, that I don't know in many cases, where to point the finger at.
> Sometimes it's stability of the CVS version, sometimes it's stability
> of Digium or whose ever interfaces, and yet sometimes it's issues with
> actual hardware architecture.
> 
> I wouldn't mind participating in such an effort, but that may just
> create parallel lists or problem reports that may be so tightly related
> that one list would take away knowledge from the other.
> 
> Comments?
> 

I'm thinking that the most reliable information about * hardware would
be from vendors that build and install * systems.  I know when we
purchase mission critical freebsd/linux boxes we have specific vendors
we go through because they know exactly what works and what doesn't.

Plus, Vendors have the cash to test out various systems.  If you start
an open group of some type, where are you going to get your hands on
the type of hardware that people will actually be using?

That said, I haven't actually seen any vendors that build custom
systems for stock * installations.  I've seen some dell poweredge
systems out there, but to me that means the vendor is just using the
best bang for the buck hardware and dropping in some digium cards.

Personally I would buy an * box from someone like asaservers.com.  At
least companies like that really know their hardware, and if you tell
them the common issues with * they could probably put together a rock
solid system.

Chris
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[Asterisk-Users] Bouncing DTMF?

[Jan Johansson]

I seem to get "bounces" on DTMF.

For instance, if I turn on debug, and I dial the voicemail, and enter 1234
as extension, I see in the logs "12234" "111234" "12344" and so on, same
with passwords.

But dialing an extension never seem to fail this way.. Any hints?


smime.p7s
Description: S/MIME cryptographic signature
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RE: [Asterisk-Users] Problems with TDM400P card

[Robert Webb]

> -Original Message-
> From: Anton Krall [mailto:[EMAIL PROTECTED]
> Sent: Friday, April 29, 2005 1:50 PM
> To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List -
> Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Problems with TDM400P card
>
> How do I remove it from kudzu?
>


I am looking for that now... Sorry it has taken so long to respond, I
had some errands to run.



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[Asterisk-Users] Asterisk@Home 1.0 released

[[EMAIL PROTECTED]]

[EMAIL PROTECTED] 1.0 released

This is the first production release of [EMAIL PROTECTED]
We have worked hard over the past few months to make
[EMAIL PROTECTED] easy to use and stable. Thanks for all
the help with testing and fixes from [EMAIL PROTECTED]
users all over the world. 

There are no new features in 1.0 from 0.9 but there
are a lot of small fixes to many of the components of
[EMAIL PROTECTED] For example the Auto-Config for Zaptel
hardware now works.

For those of you new to VOIP [EMAIL PROTECTED] is a
complete IP PBX that install from a CD. It includes a
web interface and many of Asterisk’s best add-on
features.

For support please read the [EMAIL PROTECTED] Handbook
http://asteriskathome.sourceforge.net/handbook/index.html

and use our support forum at
http://sourceforge.net/forum/?group_id=123387



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[Asterisk-Users] ISPCON: SIP CPE experts wanted for panel

[John Todd]

This is not quite on-topic for the Asterisk list, but is a much 
higher chance that I will find a rich network of possible candidates 
on this list than any other.  Besides, with the amount of problems 
that we all have with SIP and various CPE working with Asterisk, the 
benefits of any improvement in the SIP CPE market will somehow 
benefit us all.  :-)

I am going to be speaking at ISPCON on May 25 in Baltimore, Maryland 
USA on a panel which will be discussing SIP CPE.  The panel will have 
4 or so staff, and the purpose of the talk is to discuss the 
comparative benefits or drawbacks of certain SIP CPE for consumer and 
business use cases.  The audience, I expect, will be mostly 
consultants or ITSPs who are looking for some experienced discussion 
on equipment before they make their purchase decisions.

Some of the original invitees have had to bow out, so there are some 
positions open on the panel.

NOTE: Equipment designers or resellers are unfortunately perceived to 
have a bias, so I am afraid that if you fall into one of these 
categories we probably will not choose you, despite what I'm sure is 
an expert understanding of the market and devices.  Consultants who 
have a wide range of CPE experience are preferred.

There are no expenses paid, but my understanding is that you get a 
free day pass to the conference.  Please reply via private email.

JT
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[Asterisk-Users] Curious behaviour for pound (#) key with SIP X-lite SoftPhones

[Jeff Stokoe]

 

Hi,

 

I've been playing around with CFIM
and CFBS and came across something rather odd.  I found that a SIP X-lite phone
didn't give the expected results when running the sample CFIM/CFBS code from
the Wiki - see
http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding#comments

 

Barring the odd typo in the coded
example, it worls reasonably well for ZAP and IAX users but the dial string
which comes down the line from the SIP softphone is odd - it sends a decimal
representation of the hexadecimal for the # key (as far as I can tell) - what actually
arrives in the buffer is "%23".

 

Here are three log traces one for
each of SIP ZAP and IAX showing (in the NoOp) the actual dialstring.

 

It's not actually a problem because
I've decided to code up my dialplan a little differently, however it looks like
an inconsistency somewhere.  Anyone enlighten me? 

 

Inbound from SIP
softphone (X-Lite) 

 

-- Executing
NoOp("SIP/3000-168a", "*21*3002%23")
in new stack

    -- Executing
GotoIf("SIP/3000-168a", "1?3:4") in new stack

    -- Goto (internal,*21*3002%23,3)

    -- Executing
StripLSD("SIP/3000-168a", "3") in new stack

    -- Executing
DBput("SIP/3000-168a", "CFIM/3000=3002") in new stack

    -- DBput: family=CFIM, key=3000, value=3002

    -- Executing
Playback("SIP/3000-168a", "activated") in new stack

    -- Playing 'activated' (language
'en')

    -- Executing
Hangup("SIP/3000-168a", "") in new stack

  == Spawn extension (internal,
*21*3002, 6) exited non-zero on 'SIP/3000-168a'

    -- Executing
Hangup("SIP/3000-168a", "") in new stack

  == Spawn extension (internal, h,
1) exited non-zero on 'SIP/3000-168a'

sark*CLI>

 

 Inbound
from ZAP/1

 

 -- Starting simple switch on
'Zap/1-1'

    -- Executing
NoOp("Zap/1-1", "*21*3002#")
in new stack

    -- Executing
GotoIf("Zap/1-1", "0?3:4") in new stack

    -- Goto (internal,*21*3002#,4)

    -- Executing DBput("Zap/1-1",
"CFIM/=3002#") in new stack

    -- DBput: family=CFIM, key=, value=3002#

    -- Executing
Playback("Zap/1-1", "activated") in new stack

    -- Playing 'activated' (language
'en')

    -- Executing
Hangup("Zap/1-1", "") in new stack

  == Spawn extension (internal,
*21*3002#, 6) exited non-zero on 'Zap/1-1'

    -- Executing
Hangup("Zap/1-1", "") in new stack

  == Spawn extension (internal, h,
1) exited non-zero on 'Zap/1-1'

    -- Hungup 'Zap/1-1'

sark*CLI>

 

 Inbound
from IAX softphone (Firefly)  

 

-- Accepting AUTHENTICATED call from
192.168.1.21:

   > requested format = gsm,

   > requested prefs = (),

   > actual format = gsm,

   > host prefs =
(g729|gsm|ilbc|g726|ulaw|alaw),

   > priority = mine

    -- Executing
NoOp("IAX2/[EMAIL PROTECTED]", "*21*3002#")
in new stack

    -- Executing
GotoIf("IAX2/[EMAIL PROTECTED]", "0?3:4") in new stack

    -- Goto (internal,*21*3002#,4)

    -- Executing
DBput("IAX2/[EMAIL PROTECTED]", "CFIM/3002=3002#") in new stack

    -- DBput: family=CFIM, key=3002, value=3002#

    -- Executing
Playback("IAX2/[EMAIL PROTECTED]", "activated") in new stack

    -- Playing 'activated' (language
'en')

    -- Executing
Hangup("IAX2/[EMAIL PROTECTED]", "") in new stack

  == Spawn extension (internal,
*21*3002#, 6) exited non-zero on 'IAX2/[EMAIL PROTECTED]'

    -- Executing
Hangup("IAX2/[EMAIL PROTECTED]", "") in new stack

  == Spawn extension (internal, h,
1) exited non-zero on 'IAX2/[EMAIL PROTECTED]'

    -- Hungup 'IAX2/[EMAIL PROTECTED]'

sark*CLI>

 

Dialplan fragment  - coded to catch
the SIP dialstring.

 

;


; Unconditional Call Forward
(CFIM) 

;


; create call forward 

exten =>
_*21*.,1,NoOp,$\{EXTEN\}

exten =>
_*21*.,2,GotoIf($["$\{EXTEN:-3:3\}" = "%23"]?3:4) 

exten =>
_*21*.,3,StripLSD(3) 

exten =>
_*21*.,4,DBput(CFIM/$\{CALLERIDNUM\}=$\{EXTEN:4\}) 

exten =>
_*21*.,5,Playback(activated)

exten => _*21*.,6,Hangup 

 

 






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RE: [Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)

[Neal Walton]

Many channel banks have two T-1 connectors and support a feature called 
'drop and insert'.  This allows some of the DS0 channels to be cross 
connected from one T-1 connection to the other.  The first T-1 connection 
can go to the telco or an interface card in a computer, and the second T-1 
can go to another channel bank.  Some of the channels can be dropped off at 
the first channel bank while the rest can continue on to the second channel 
bank.  You are asking about E-1 and PBX instead of T-1 and channel bank, 
but if I understand the 'drop and insert' correctly, and if your hardware 
supports it, this may work for you.

> Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto:
>
>>Hi,
>>
>>Assume I have one E1 digium card to which I want to plug two distinct E1 
PABXs ,
>>one with 15 channels and the other with 15 channels;
>>
>>Is there a sort of E1 multiplexer devise that allows me to plug in one 
hand the
>>E1 port of the Digium card and on the other hand the two PABXs? In this 
same
>>devise, I should be able to say that 15 channels need to go to first 
Interface
>>and 15 other channels need to go to other interface.
>>
>>Or is it necessary to acquire a another E1 card although I don't need to 
process
>>more channels (30 channels are ok).
>>
>>Any help is greatly appreciated.


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Re: [Asterisk-Users] User events - a dumb question

[Thomas Miller]

http://www.voip-info.org/wiki-Asterisk+Manager+API+Action+Events


,n,Hangup
> 
> If I asterisk -r, when I dial the 888, I see
> Userevent appearing in the 
> console.
> 
> However, if I telnet to the * manager using a name
> and password that has 
> the "user" option, that telnet session sees
> everything but the user event.
> 
> What am I missing ?
> 
> manager.conf:
> 
> [event]
> secret=event
> read=system,user
> write=call,command,agent
> 

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[Asterisk-Users] Detecting DeadLocks

[Paradise Dove]

Is there any way to detect * deadlocks automatically?
i.e with a running program in background.

Paradise Dove
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Re: [Asterisk-Users] Asterisk Hardware Recommendation

[Daniel Salama]

This is an interesting question. I haven't tested it but would love to 
know if it works or not. Anyone?

- Daniel
On Apr 29, 2005, at 3:38 AM, Michael Welter wrote:
I haven't seen this before--can an agent log into a queue on a remote 
(i.e. over IAX) Asterisk server?
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Re: [Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)

[ht]

I just called this company. They seem to do what is required. Now remains the
pricing part of it. I will wait for their feedback.

http://www.megatelindustries.com/products.htm

Hakem,


Selon Julio Arruda <[EMAIL PROTECTED]>:

> Matteo Brancaleoni wrote:
> > yes, some multiplexer allows that, but they're quite expensive
> > compared to another E1 card for asterisk.
> > I think you'll need at least 1k $$$ for a such splitter.
> >
>
>
> Matteo, would you have any reference for this 'mux/splitter' ?
> I would guess it need to be smart enough to dig into the signalling,
> since is not only the PCM DS0s that would need to be "Y-splitted".
> [], 
>
> >
> > Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto:
> >
> >>Hi,
> >>
> >>Assume I have one E1 digium card to which I want to plug two distinct E1
> PABXs ,
> >>one with 15 channels and the other with 15 channels;
> >>
> >>Is there a sort of E1 multiplexer devise that allows me to plug in one hand
> the
> >>E1 port of the Digium card and on the other hand the two PABXs? In this
> same
> >>devise, I should be able to say that 15 channels need to go to first
> Interface
> >>and 15 other channels need to go to other interface.
> >>
> >>Or is it necessary to acquire a another E1 card although I don't need to
> process
> >>more channels (30 channels are ok).
> >>
> >>Any help is greatly appreciated.
>
>
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RE: [Asterisk-Users] Traffic Testing

[Anton Krall]

Any url? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|René Mayorga
|Sent: Viernes, 29 de Abril de 2005 12:29 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Traffic Testing
|
|I'm using "sip-tester" you should try it gnuws:~# apt-cache 
|search sip-tester sip-tester - a performance testing tool for 
|the SIP protocol gnuws:~# 
|
|
|On Fri, 2005-04-29 at 14:14 +0200, Nils Ohlmeier wrote:
|> The homepage http://sipsak.org contains some examples. If you need 
|> help with special cases drop me a line.
|> 
|> Regards
|>   Nils Ohlmeier
|> 
|> On Friday 29 April 2005 02:54, Anton Krall wrote:
|> > Can you send some command line examples on how to use it?
|> >
|> > Thx!
|> >
|> > |-Original Message-
|> > |From: [EMAIL PROTECTED]
|> > |[mailto:[EMAIL PROTECTED] On Behalf Of 
|> > |[EMAIL PROTECTED]
|> > |Sent: Jueves, 28 de Abril de 2005 07:05 p.m.
|> > |To: asterisk-users@lists.digium.com
|> > |Subject: RE: [Asterisk-Users] Traffic Testing
|> > |
|> > |> -Original Message-
|> > |> From: [EMAIL PROTECTED]
|> > |> [mailto:[EMAIL PROTECTED] Behalf Of 
|> > |> Anton Krall
|> > |> Sent: Thursday, April 28, 2005 6:07 PM
|> > |> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|> > |> Subject: [Asterisk-Users] Traffic Testing
|> > |>
|> > |>
|> > |> Guys, is there any way to generate simulated traffic via sip or 
|> > |> IAX2 for testing cpu load and asterisk? (sip client 
|simulation, etc)?
|> > |
|> > |yes, use  sipsak utility
|> > |
|> > |--
|> > |___
|> > |Asterisk-Users mailing list
|> > |Asterisk-Users@lists.digium.com
|> > |http://lists.digium.com/mailman/listinfo/asterisk-users
|> > |To UNSUBSCRIBE or update options visit:
|> > |   http://lists.digium.com/mailman/listinfo/asterisk-users
|> >
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|> > http://lists.digium.com/mailman/listinfo/asterisk-users
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|> >http://lists.digium.com/mailman/listinfo/asterisk-users
|> 
|--
|René Mayorga
|Internet & Data
|El Salvador Telecom S.A. de S.V.
|Tel:(503) 247-7246
|(503) 247-7156
|Cel:(503) 962-8205
|
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RE: [Asterisk-Users] Traffic Testing

[Anton Krall]

Thx Rene, Ill give it a try 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|René Mayorga
|Sent: Viernes, 29 de Abril de 2005 12:29 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Traffic Testing
|
|I'm using "sip-tester" you should try it gnuws:~# apt-cache 
|search sip-tester sip-tester - a performance testing tool for 
|the SIP protocol gnuws:~# 
|
|
|On Fri, 2005-04-29 at 14:14 +0200, Nils Ohlmeier wrote:
|> The homepage http://sipsak.org contains some examples. If you need 
|> help with special cases drop me a line.
|> 
|> Regards
|>   Nils Ohlmeier
|> 
|> On Friday 29 April 2005 02:54, Anton Krall wrote:
|> > Can you send some command line examples on how to use it?
|> >
|> > Thx!
|> >
|> > |-Original Message-
|> > |From: [EMAIL PROTECTED]
|> > |[mailto:[EMAIL PROTECTED] On Behalf Of 
|> > |[EMAIL PROTECTED]
|> > |Sent: Jueves, 28 de Abril de 2005 07:05 p.m.
|> > |To: asterisk-users@lists.digium.com
|> > |Subject: RE: [Asterisk-Users] Traffic Testing
|> > |
|> > |> -Original Message-
|> > |> From: [EMAIL PROTECTED]
|> > |> [mailto:[EMAIL PROTECTED] Behalf Of 
|> > |> Anton Krall
|> > |> Sent: Thursday, April 28, 2005 6:07 PM
|> > |> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|> > |> Subject: [Asterisk-Users] Traffic Testing
|> > |>
|> > |>
|> > |> Guys, is there any way to generate simulated traffic via sip or 
|> > |> IAX2 for testing cpu load and asterisk? (sip client 
|simulation, etc)?
|> > |
|> > |yes, use  sipsak utility
|> > |
|> > |--
|> > |___
|> > |Asterisk-Users mailing list
|> > |Asterisk-Users@lists.digium.com
|> > |http://lists.digium.com/mailman/listinfo/asterisk-users
|> > |To UNSUBSCRIBE or update options visit:
|> > |   http://lists.digium.com/mailman/listinfo/asterisk-users
|> >
|> > ___
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|> > Asterisk-Users@lists.digium.com
|> > http://lists.digium.com/mailman/listinfo/asterisk-users
|> > To UNSUBSCRIBE or update options visit:
|> >http://lists.digium.com/mailman/listinfo/asterisk-users
|> 
|--
|René Mayorga
|Internet & Data
|El Salvador Telecom S.A. de S.V.
|Tel:(503) 247-7246
|(503) 247-7156
|Cel:(503) 962-8205
|
|___
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|http://lists.digium.com/mailman/listinfo/asterisk-users
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|   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Asterisk Hardware Architecture Group

[Daniel Salama]

I think that would be a great idea. The only problem I see is that  
Asterisk is growing its feature set and maturing at such a dynamic  
rate, that I don't know in many cases, where to point the finger at.  
Sometimes it's stability of the CVS version, sometimes it's stability  
of Digium or whose ever interfaces, and yet sometimes it's issues with  
actual hardware architecture.

I wouldn't mind participating in such an effort, but that may just  
create parallel lists or problem reports that may be so tightly related  
that one list would take away knowledge from the other.

Comments?
- Daniel
On Apr 29, 2005, at 2:42 PM, Matt Roth wrote:
List members,
Does anyone have an interest in forming a hardware architecture group?
It seems that Asterisk is so tightly linked to specialized hardware  
and its corresponding architecture that developing the software alone  
is insufficient for its adoption to large scale applications.

Thank you,
Matthew Roth
http://voip-info.org/tiki-index.php? 
page=Running%20Asterisk%20on%20Debian
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Re: [Asterisk-Users] Traffic Testing

[Mailing List]

SIPp is a free Open Source test tool / traffic generator for the SIP protocol
http://sipp.sourceforge.net/
On Fri, 2005-04-29 at 14:14 +0200, Nils Ohlmeier wrote:
The homepage http://sipsak.org contains some examples. If you need help with
special cases drop me a line.
Regards
  Nils Ohlmeier
On Friday 29 April 2005 02:54, Anton Krall wrote:
> Can you send some command line examples on how to use it?
>
> Thx!
>
> |-Original Message-
> |From: [EMAIL PROTECTED]
> |[mailto:[EMAIL PROTECTED] On Behalf Of
> |[EMAIL PROTECTED]
> |Sent: Jueves, 28 de Abril de 2005 07:05 p.m.
> |To: asterisk-users@lists.digium.com
> |Subject: RE: [Asterisk-Users] Traffic Testing
> |
> |> -Original Message-
> |> From: [EMAIL PROTECTED]
> |> [mailto:[EMAIL PROTECTED] Behalf Of Anton
> |> Krall
> |> Sent: Thursday, April 28, 2005 6:07 PM
> |> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> |> Subject: [Asterisk-Users] Traffic Testing
> |>
> |>
> |> Guys, is there any way to generate simulated traffic via sip or IAX2
> |> for testing cpu load and asterisk? (sip client simulation, etc)?
> |
> |yes, use  sipsak utility
> |
> |--
> |___
> |Asterisk-Users mailing list
> |Asterisk-Users@lists.digium.com
> |http://lists.digium.com/mailman/listinfo/asterisk-users
> |To UNSUBSCRIBE or update options visit:
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--
René Mayorga
Internet & Data
El Salvador Telecom S.A. de S.V.
Tel:(503) 247-7246
   (503) 247-7156
Cel:(503) 962-8205
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RE: [Asterisk-Users] Problems with TDM400P card

[Anton Krall]

Which card do you recommend using instead of the tdm400p? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|John Novack
|Sent: Viernes, 29 de Abril de 2005 09:19 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Problems with TDM400P card
|
|Anton Krall wrote:
|> Zttool shows nothing inside thebox.
|I have had similar problems with a TDM400 and CERTAIN 
|Motherboards which are PCI 2.2 but the TDM400 is not seen, in 
|my case, AT ALL The one I have reports it as an E/F but the 
|silk-screen clearly says H, Digium contends there is no 
|problem with the card, the reporting of different version 
|numbers is a "known bug" but doesn't prevent the card from 
|working, and because I can place it in another machine and get 
|it working, they refuse to acknowledge there is any defect in 
|the board.
|
|Perhaps a different motherboard? That is Digium's answer.
|Just keep going through hardware that otherwise meets 
|published specs until you find one that works.
|
|I have to conclude that, due to Digiums refusal to acknowledge 
|there are problems with the design, ( and there are more I 
|won't bore you with ) and no willingness to address the issues 
|that have been raised on this list time and time again, that 
|the TDM400 should be considered a "card of last resort" when 
|absolutely nothing else will work.
|Seems their IAXy falls into that same classification.
|Can't say about their T1/E1 cards
|
|JMO
|
|John Novack
|
|
|> 
|> I tried removing the x100 cards, moving the tdm card around, 
|disabled 
|> all usb and unnecessary stuff still, kudzu when booting up shows the 
|> card and the card shows up on /etc/sysconfig/hwconf but I wonder why 
|> it shows 2 of these and I only have 1 tdm400p card with 1 module
|> 
|> class: MODEM
|> bus: PCI
|> detached: 1
|> driver: hisax
|> desc: "Tiger Jet Network Inc.|Intel 537"
|> vendorId: e159
|> deviceId: 0001
|> subVendorId: 8086
|> subDeviceId: 0003
|> pciType: 1
|> -
|> class: MODEM
|> bus: PCI
|> detached: 1
|> driver: hisax
|> desc: "Tiger Jet Network Inc.|Intel 537"
|> vendorId: e159
|> deviceId: 0001
|> subVendorId: 8086
|> subDeviceId: 0003
|> pciType: 1
|> 
|> Still, interrupts doesn't show the card
|> 
|> [EMAIL PROTECTED] root]# cat /proc/interrupts
|>CPU0
|>   0:3994353  XT-PIC  timer
|>   1:  2  XT-PIC  keyboard
|>   2:  0  XT-PIC  cascade
|>   8:  1  XT-PIC  rtc
|>  10:  95510  XT-PIC  eth0
|>  14: 129871  XT-PIC  ide0
|> NMI:  0
|> ERR:  0
|> 
|> And when trying to load the drvier
|> 
|> [EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# 
|> modprobe wctdm
|> /lib/modules/2.4.20-8/misc/wctdm.o: init_module: No such device
|> Hint: insmod errors can be caused by incorrect module parameters, 
|> including invalid IO or IRQ parameters.
|>   You may find more information in syslog or the output 
|from dmesg
|> /lib/modules/2.4.20-8/misc/wctdm.o: insmod 
|> /lib/modules/2.4.20-8/misc/wctdm.o failed
|> /lib/modules/2.4.20-8/misc/wctdm.o: insmod wctdm failed You have new 
|> mail in /var/spool/mail/root
|> 
|> I tried using diff. modules with no luck.,. Could be the 
|mobo itself?
|> 
|>  
|> 
|> |-Original Message-
|> |From: [EMAIL PROTECTED]
|> |[mailto:[EMAIL PROTECTED] On Behalf Of Rich 
|> |Adamson
|> |Sent: Viernes, 29 de Abril de 2005 08:59 a.m.
|> |To: Asterisk Users Mailing List - Non-Commercial Discussion
|> |Subject: Re: [Asterisk-Users] Problems with TDM400P card
|> |
|> |> I have a problem getting a TDM400P card to go.
|> |> 
|> |> It has 4 FXS ports (green modules) and I get this error:
|> |> 
|> |> [EMAIL PROTECTED] root]# ztcfg -v
|> |> 
|> |> Zaptel Configuration
|> |> ==
|> |> 
|> |> 
|> |> Channel map:
|> |> 
|> |> Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS 
|> |> Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart 
|> |> (Default)
|> |> (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 
|> |04) Channel
|> |> 05: FXO Kewlstart (Default) (Slaves: 05) Channel 06: FXO Kewlstart
|> |> (Default) (Slaves: 06)
|> |> 
|> |> 6 channels configured.
|> |> 
|> |> ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you 
|> |> forget that FXS interfaces are configured with FXO
|> |signalling and that
|> |> FXO interfaces use FXS signalling?
|> |> 
|> |> My zaptel.conf reads:
|> |> 
|> |> [EMAIL PROTECTED] root]# more /etc/zaptel.conf
|> |> fxsks=1
|> |> fxsks=2
|> |> fxoks=3-6
|> |> loadzone=us
|> |> defaultzone=us
|> |> 
|> |> And my rc.local loads:
|> |> 
|> |> /sbin/modprobe zaptel
|> |> /sbin/modprobe wcfxo
|> |> /sbin/modprobe wctdm
|> |> 
|> |> The 2 100p cards load perfectly but the TDM is not. 
|> |> 
|> |> Any ideas?
|> |
|> |What does zttool indicate?
|> |
|> |Have you tried moving the cards around in different slots?
|> |
|> |Any shared interrupt 

Re: [Asterisk-Users] chan_zap graceful failure

[Jeb Campbell]

Andrew Kohlsmith wrote:
No; if the driver didn't load that's a major problem.  Remember that if the 
channel doesn't exist all the subsequent channels "move up"...  serious 
potential security issues.
Good points.  What if it kept the number (so nothing "moved up"), but 
marked the channel inuse (or even better a new non working status code).

I'd rather have the system as it is, where it fails out with an error that is 
easy to understand so I can fix the problem.
I agree that it should be a very loud error (and possibly repeated 
notifications on the console).  But I also think that it should be able 
to limp along.  What would you think of a commercial phone system that 
completely dies when one port dies?

I appreciate your points (thats why I wanted to discuss this), maybe we 
could satisfy both with an option in zapata.conf (keepgoing=yes)?

Thanks,
Jeb
--
Jeb Campbell
[EMAIL PROTECTED]
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[Asterisk-Users] Asterisk Hardware Architecture Group

[Matt Roth]

List members,
Does anyone have an interest in forming a hardware architecture group?
It seems that Asterisk is so tightly linked to specialized hardware and its 
corresponding architecture that developing the software alone is insufficient 
for its adoption to large scale applications.
Thank you,
Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
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Re: [Asterisk-Users] quadbri bristuff ztcfg fail

[Kristof Hardy]

Sander wrote:
I compiled the bristuff drivers and then I do
--
When doing lsmod I can see qozap is loaded with zaptel but no entry in 
/proc/zaptel/
Did the compiling go correct?
What version of bristuff are you using? (latest? 0.2.0rc8a)
What linux distro are you running and wich kernel?
Your zaptel.conf looks okay, but you should have something in 
/proc/zaptel indeed..

Cheers.
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Re: [Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)

[Julio Arruda]

Matteo Brancaleoni wrote:
yes, some multiplexer allows that, but they're quite expensive
compared to another E1 card for asterisk.
I think you'll need at least 1k $$$ for a such splitter.

Matteo, would you have any reference for this 'mux/splitter' ?
I would guess it need to be smart enough to dig into the signalling, 
since is not only the PCM DS0s that would need to be "Y-splitted".
[], 

Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto:
Hi,
Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs ,
one with 15 channels and the other with 15 channels;
Is there a sort of E1 multiplexer devise that allows me to plug in one hand the
E1 port of the Digium card and on the other hand the two PABXs? In this same
devise, I should be able to say that 15 channels need to go to first Interface
and 15 other channels need to go to other interface.
Or is it necessary to acquire a another E1 card although I don't need to process
more channels (30 channels are ok).
Any help is greatly appreciated.

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RE: [Asterisk-Users] Asterisk Hardware Recommendation

[Ken N. March]

Hi Matt, 

> Does anyone have experience with using NAS
> (http://en.wikipedia.org/wiki/Network-attached_storage) or SAN
> (http://en.wikipedia.org/wiki/Storage_area_network) for this 
> application?

I've had our agent/queue recordings dumped both to local disk and SAN
(currently using local disk as the SAN is being used for some other
stuff).

With both SAN (2GB FC) and local disk, we haven't had any problems like
the ones described by Daniel.  One of our live servers has 4 PRI's going
with an average of about 40-50 calls at any given time during the day
(60-70 peak), all being recorded, and we've had zero issues.  The other
two servers have similar configurations, but lower call volumes (5-20
calls depending on time of day).

I'd be leary about doing it over NFS or Samba or any other sort of
networked filesystem though.  For our servers, that'd be extra I/O
that'd have to go over either one of the network interfaces (both of
which are plenty busy already with IAX2 and/or SIP).  I guess it depends
on your network card and how well behaved it is in terms of
interrupts/etc..  You could say the same thing for local disk if you had
slower drives and/or disk controllers.

Ken.
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[Asterisk-Users] GR-303 zaptel and zapata configurations

[Chris A. Icide]

-BEGIN PGP SIGNED MESSAGE-

Does anyone have any working example GR-303 configurations for zaptel
and 
zapata conf?

The information available on the wiki as well as in the sample 
configurations just doesn't seem to be enough to bridge the gap for
me.

In Zaptel.conf,

Do you set up a GR-303 circuit like a PRI with b and d channels or do
you 
set fxo or fxs, ks signalling?
How do you configure the channels for the control and timing channels
(12 
and 24 on a T1 from a 5ESS switch)?

In zapata.conf,

Trunk groups require 1 or more signalling channels, so this would
indicate 
that you need to set up the circuits in zaptel as PRI?

signalling can be gr303fxoks_cpe/net?

Do you assign CRV's instead of channels, or do you assign channels
and crv's?


I would appreciate any help here.  I'm trying to set up an asterisk
system 
as a GR-303 CPE device off of a 5ESS switch using 2 T1 circuits.

- -Chris

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Re: [Asterisk-Users] Asterisk Hardware Recommendation

[Steve Totaro]

Maybe something like this would be good.
http://www.pcmicrostore.com/PartDetail.aspx?q=p:10502197
- Original Message - 
From: "Matt Roth" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, April 29, 2005 2:11 PM
Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation


Does anyone have experience with using NAS 
(http://en.wikipedia.org/wiki/Network-attached_storage) or SAN 
(http://en.wikipedia.org/wiki/Storage_area_network) for this application?

Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
Daniel Salama wrote:
Sure.
I setup a small lab on a machine with 4 T1s and 36 agents logged in.  The 
system was configured to Monitor all outbound calls as well as  monitor 
all calls distributed by Queue app (monitor-format setting in 
queues.conf).

When recording to local disk, everything was working fine. Agents were 
busy 99.5% and there were at least 30 calls waiting in Queue to be 
distributed. Average call conversation length was about 7.5 minutes.

Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E.
The moment we pushed the load on the Asterisk machine, everything  worked 
for about 40 seconds. Then call quality started suffering  significantly. 
Chopped audio. Bad audio. No audio. Good audio. You  could imagine. So we 
stopped the test.

Then we unmounted the NFS drive and repeated the test again. Everything 
worked fine again.

The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM. 
During all tests, CPU utilization was about 55% on the average (for  each 
CPU). Memory usage was under 1G.

I would say I need to try more troubleshooting. Maybe there was 
congestion on the Fast-E, although preliminary analysis indicates there 
were no CRC errors, collisions, or packet loss.

The NFS machine was completely idle.
Last, we repeated the test over a 1 hour period. This time, Monitor was 
recording on local drives and we were copying files every 15 minutes 
with a background process (perl script) to NFS mount point. Everything 
worked fine as well.

I don't know if these tests are conclusive yet. However, from the 
results so far, I would recommend staying away from recording to NFS 
mounted point. I will continue running simulations to see if anything 
else can be identified.

Thanks,
Daniel
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[Asterisk-Users] SIP/IAX softphone with g729/723

[M. Ehsanul Karim]

Hello,
   I am in search for a SIP or IAX softphone that works with * and
supports commercial codecs like g729 and g723.1. It can be commercial
license . I have been through Xten and SJphone.

   Let me know anyone can offer this. I need it on an urgent basis.

Thanks.

Ehsanul Karim
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Re: [Asterisk-Users] T1 Technology and VoIP Gateway Primer

[Matt Roth]

Callum,
Matt, is this similar to the idea that you have for your project ?  
Similar, except we are looking to have a single Asterisk server attached 
to the Gateway for centralized queuing, reportings, call recoring, etc.  
We are a call center, so having everything in a single environment is a 
huge advantage for us.

What Cisco hardware have you looked at so far ?  
Same as you, the AS series of voice gateways. I'm not the person looking 
specifically at hardware anymore, though. Our entire proposed setup is 
going to be posted here for review once it's known.

How many E1/T1 lines are you going to have terminating on your setup ?
16 to 20 to begin with.  Odds are it will scale up over time, to what 
number I'm not sure.

Yours truly,
Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
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[Asterisk-Users] Adtran 600

[Steve Totaro]

is it possible to program an adtran 600 to act as 
the network and asterisk to be cpe?
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Re: [Asterisk-Users] Realtime feature

[Matthew Boehm]

> I don't have any failure I just want to know if the next release will
> be 1.0.8 or 1.2.

Oh good grief. My fault..for some reason I read the subject and
processed "failure" instead of 'feature'. Thank goodness its friday.

Anyway, there will probably be a 1.0.8 release. But remember that 1.0.*
releases are for bug fixes only. No new features are incorporated into the
1.0 branch.

-Matthew

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Re: [Asterisk-Users] Asterisk Hardware Recommendation

[Matt Roth]

Does anyone have experience with using NAS 
(http://en.wikipedia.org/wiki/Network-attached_storage) or SAN 
(http://en.wikipedia.org/wiki/Storage_area_network) for this application?

Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
Daniel Salama wrote:
Sure.
I setup a small lab on a machine with 4 T1s and 36 agents logged in.  
The system was configured to Monitor all outbound calls as well as  
monitor all calls distributed by Queue app (monitor-format setting in  
queues.conf).

When recording to local disk, everything was working fine. Agents 
were  busy 99.5% and there were at least 30 calls waiting in Queue to 
be  distributed. Average call conversation length was about 7.5 minutes.

Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E.
The moment we pushed the load on the Asterisk machine, everything  
worked for about 40 seconds. Then call quality started suffering  
significantly. Chopped audio. Bad audio. No audio. Good audio. You  
could imagine. So we stopped the test.

Then we unmounted the NFS drive and repeated the test again. 
Everything  worked fine again.

The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM.  
During all tests, CPU utilization was about 55% on the average (for  
each CPU). Memory usage was under 1G.

I would say I need to try more troubleshooting. Maybe there was  
congestion on the Fast-E, although preliminary analysis indicates 
there  were no CRC errors, collisions, or packet loss.

The NFS machine was completely idle.
Last, we repeated the test over a 1 hour period. This time, Monitor 
was  recording on local drives and we were copying files every 15 
minutes  with a background process (perl script) to NFS mount point. 
Everything  worked fine as well.

I don't know if these tests are conclusive yet. However, from the  
results so far, I would recommend staying away from recording to NFS  
mounted point. I will continue running simulations to see if anything  
else can be identified.

Thanks,
Daniel
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[Asterisk-Users] Re: Sip endpoints that support re-invite??

[Gene Willingham]

> Hi,
> 
> I am doing some testing with asterisk using Cisco IP Phones 7960's and
> EyeBeam. I have canreinvite=yes on all my devices but the media still
> goes through the asterisk box. Does it mean that Cisco and Xten do not
> support re-invites? If yes can you recommend SIP phones or adapters
> that support re-invites.
> 
> Thanks in advance.
> 
> Hamza Moore.
> 

  Do you have the option "t" or "T" in the dial statement?  If you do
asterisk ignores the re-invite so it can support transfer of call using
asterisk # transfer.

   


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Re: [Asterisk-Users] Asterisk Hardware Recommendation

[Matt Roth]

Thanks Daniel,
We may end up replicating your tests in order to confirm some of your 
results.  I don't know if it will be anytime soon, because we don't have 
the hardware yet. Regardless, I will share my results with the list.

Anyone out there have any ideas on why the NFS mount affected call 
quality?  It seems backwards, since it should have relieved some of the 
load from the Asterisk machine.

Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
Daniel Salama wrote:
Sure.
I setup a small lab on a machine with 4 T1s and 36 agents logged in.  
The system was configured to Monitor all outbound calls as well as  
monitor all calls distributed by Queue app (monitor-format setting in  
queues.conf).

When recording to local disk, everything was working fine. Agents 
were  busy 99.5% and there were at least 30 calls waiting in Queue to 
be  distributed. Average call conversation length was about 7.5 minutes.

Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E.
The moment we pushed the load on the Asterisk machine, everything  
worked for about 40 seconds. Then call quality started suffering  
significantly. Chopped audio. Bad audio. No audio. Good audio. You  
could imagine. So we stopped the test.

Then we unmounted the NFS drive and repeated the test again. 
Everything  worked fine again.

The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM.  
During all tests, CPU utilization was about 55% on the average (for  
each CPU). Memory usage was under 1G.

I would say I need to try more troubleshooting. Maybe there was  
congestion on the Fast-E, although preliminary analysis indicates 
there  were no CRC errors, collisions, or packet loss.

The NFS machine was completely idle.
Last, we repeated the test over a 1 hour period. This time, Monitor 
was  recording on local drives and we were copying files every 15 
minutes  with a background process (perl script) to NFS mount point. 
Everything  worked fine as well.

I don't know if these tests are conclusive yet. However, from the  
results so far, I would recommend staying away from recording to NFS  
mounted point. I will continue running simulations to see if anything  
else can be identified.

Thanks,
Daniel
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RE: [Asterisk-Users] Problems with TDM400P card

[Anton Krall]

How do I remove it from kudzu? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Robert Webb
|Sent: Viernes, 29 de Abril de 2005 08:57 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] Problems with TDM400P card
|
|> 
|> Zttool shows nothing inside thebox.
|> 
|> I tried removing the x100 cards, moving the tdm card around, 
|disabled 
|> all usb and unnecessary stuff still, kudzu when booting up shows the 
|> card and the card shows up on /etc/sysconfig/hwconf but I wonder why 
|> it shows 2 of these and I only have 1 tdm400p card with 1 module
|> 
|
|
|If I remember correctly, when I installed [EMAIL PROTECTED] and it 
|did its reboot, the TDM was removed from kudzu as it loaded 
|the linux zaptel and you want to load the zaptel obtained from 
|Digium. Try removing it permanantly from kudzu then try 
|loading your modules.
|
|Robert
|
|
|
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|
|

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RE: [Asterisk-Users] IAX2 one way audio

[Ken N. March]

Yep, seeing the exact same problem here if it's a trunked IAX2
connection.  A CVS checkout I had from early April did the same thing.

Try setting trunk=no and see if it works.  Seemed to fix the problem
here for us with our development cluster.

To quote bkw (from earlier this week in IRC), "trunking is BROKEN" in
CVS.  I've been meaning to finish getting all the info together to bug
it, but I've been pretty swamped here at work.  Hopefully I'll get to it
today if someone else doesn't.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, April 29, 2005 11:03 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] IAX2 one way audio

Upgraded one of my asterisk servers to the latest cvs head version last
nigh now I get one way audio on IAX2 channels when calling other
asterisk servers. Anyone seeing the some problems? 

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Re: [Asterisk-Users] Realtime feature

[Rodrigo P. Telles]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Matthew,
Matthew Boehm escreveu:
| Rodrigo P. Telles wrote:
|
|
|>Does someone knows if the next release of Asterisk (1.0.8?) will have
|>Realtime support and when we will have the next Asterisk release
|>with Realtime features?
|
|
| Where is your failure? I don't see anything. The next stable release of
| asterisk will be 1.2 and it will have RealTime. The current CVS-HEAD (aka
| 1.1) has it now and it is very stable.
I don't have any failure I just want to know if the next release will be 1.0.8
or 1.2.
|
| The eta on 1.2 is unknown. You can help 1.2 along by downloading it and
| running it to help fix bugs.
Great, I'll do that!
Thanks for your answer Matthew.
regards.
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sqrNa7Lz30lpWcxxP9ciN1E=
=3UYk
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Re: [Asterisk-Users] IAX2 one way audio

[Richard Lyman]

Duane Cox wrote:
Do you get 2-way audio that sometimes drops off to 1-way audio then picks
back up as 2-way? (Thats what I see)
Not sure if my problem is a lost packet issue as I am sending IAX off net.
Duane Cox
- Original Message - 
From: <[EMAIL PROTECTED]>
To: 
Sent: Friday, April 29, 2005 12:03 PM
Subject: [Asterisk-Users] IAX2 one way audio

 

Upgraded one of my asterisk servers to the latest cvs head version last
nigh now I get one way audio on IAX2 channels when calling other
asterisk servers. Anyone seeing the some problems?
___
   

do you still have trunk=yes in your iax.conf?
if so, remove them for CVS-HEAD stuff.
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Re: [Asterisk-Users] Group/Broadcast Voicemail

[Chris Stinson]

I have one with 33. but I can't get the voicemail to copy to more than 
20 mailboxes.
-

Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
[EMAIL PROTECTED]
Eric Wieling aka ManxPower wrote:
Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes?
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Re: [Asterisk-Users] first few seconds of call is lost

[snacktime]

On 4/29/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
> 
> > I'm testing this strange behavior using livevoip, teliax, and
> > voicepulse connect.  I'm calling our office phone which picks up after
> > two rings and plays a greeting.  With livevoip and teliax I hear 3-4
> > rings and when the line answers I find myself a few  seconds into the
> > initial greeting.  With voicepulse I hear two rings and then hear the
> > complete greeting, which is the same as if I call using a pots line.
> > Doesn't seem to make a difference whether I use iax or sip.
> >
> > This has happened consistantly and since day one of using teliax and
> > livevoip, while voicepulse has never had this problem.
> 
> I'm using:
> [bus-ivr-main]
> exten => s,1,Wait,1
> exten => s,2,Answer
> exten => s,3,DigitTimeout,5
> exten => s,4,ResponseTimeout,15
> exten => s,5,Background(npi-greeting)  ; "Thanks for calling press 1 for"
> 
> for both livevoip.com and teliax.com (both with iax), no problems.
> If you want to listen to it, call 913-440- and listen for the
> number of rings before the ivr audio.
> 
>
My problem is on outgoing calls via livevoip/teliax to a normal pstn
number.  livevoip/teliax DID's are not in the picture.  Our office
phone is connected via pstn and is on a nortel norstar mics.

Chris
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Re: [Asterisk-Users] Traffic Testing

[René Mayorga]

I'm using "sip-tester" you should try it
gnuws:~# apt-cache search sip-tester
sip-tester - a performance testing tool for the SIP protocol
gnuws:~# 


On Fri, 2005-04-29 at 14:14 +0200, Nils Ohlmeier wrote:
> The homepage http://sipsak.org contains some examples. If you need help with 
> special cases drop me a line.
> 
> Regards
>   Nils Ohlmeier
> 
> On Friday 29 April 2005 02:54, Anton Krall wrote:
> > Can you send some command line examples on how to use it?
> >
> > Thx!
> >
> > |-Original Message-
> > |From: [EMAIL PROTECTED]
> > |[mailto:[EMAIL PROTECTED] On Behalf Of
> > |[EMAIL PROTECTED]
> > |Sent: Jueves, 28 de Abril de 2005 07:05 p.m.
> > |To: asterisk-users@lists.digium.com
> > |Subject: RE: [Asterisk-Users] Traffic Testing
> > |
> > |> -Original Message-
> > |> From: [EMAIL PROTECTED]
> > |> [mailto:[EMAIL PROTECTED] Behalf Of Anton
> > |> Krall
> > |> Sent: Thursday, April 28, 2005 6:07 PM
> > |> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > |> Subject: [Asterisk-Users] Traffic Testing
> > |>
> > |>
> > |> Guys, is there any way to generate simulated traffic via sip or IAX2
> > |> for testing cpu load and asterisk? (sip client simulation, etc)?
> > |
> > |yes, use  sipsak utility
> > |
> > |--
> > |___
> > |Asterisk-Users mailing list
> > |Asterisk-Users@lists.digium.com
> > |http://lists.digium.com/mailman/listinfo/asterisk-users
> > |To UNSUBSCRIBE or update options visit:
> > |   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
-- 
René Mayorga
Internet & Data 
El Salvador Telecom S.A. de S.V.
Tel:(503) 247-7246
(503) 247-7156
Cel:(503) 962-8205

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Re: [Asterisk-Users] call a peer over the asterisk manager with a php script

[Richard Lyman]

Guy Boehm wrote:
wau thank you it works!! but,
 
first it says that e loop is detected,
 
and secondary what must I do to hand over the new working channel to 
my x-lite to use it???
 
 
DENGENS 

Richard Lyman <[EMAIL PROTECTED]> wrote:
Guy Boehm wrote:
> fputs($socket, "Channel: 6159bfb47b9\r\n\r\n");
>
>Response: Error
>Message: Invalid channel
>
>
>
>
the Channel: var needs to be in the form of type/dev/numbertocall
like Channel: IAX2/user:[EMAIL PROTECTED]/14085551212
i have no clue what you meant by 'e loop', as for handing over the 
call.. i think you really need to read the handbook and get a base 
knowledge of what asterisk is and how it works.

without that, you would be in here 5 times aday asking questions and 
probably getting flamed like crazy.

fire up a brower and goto www.digium.com  click the documentation link 
on the left side. 

there is a getting started section, read the FAQ
there is a reference doc section, read the asterisk project handbook, 
version 2
there is also get http://www.digium.com/handbook-draft.pdf 

good luck
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Re: [Asterisk-Users] Re: Re: T1/DS1/ISDN PRI

[Matt Roth]

David Josephson,
Not off-base, but you haven't made it all the way home yet. This is 
another layer of the puzzle, and again we are not talking about apples 
and apples here. "Circuit switched" means that there is a (real or 
virtual) circuit that takes data on an input port and delivers it to 
an output port somewhere. "Packet switched" means that each packet of 
data is examined by each port it passes, to see where it should be 
sent. Normally this layer of VoIP traffic is handled not in Asterisk, 
but in a router. You could run the router on the same Linux box that's 
running Asterisk (and send packets to different Ethernet ports 
depending on their destination address) but normally this task is 
handled by a separate router. There is a small computational overhead 
associated with adding and decoding Ethernet packets but the main 
routing work is done outside Asterisk, and isn't too intensive. You 
could read up on TCP/IP routing and understand how this works in more 
detail.
We plan on using a Gb switch with 100 Mbps ports to handle the routing.
It's not something you can "take a look at" in my experience. Some of 
the Bell System training material that comes up on eBay is good. You 
need to follow the progress from circuit-switched voice telephony 
circa 1930 through modern TDM, and then look at the development of 
TCP/IP switching separately.
75 years of telephony and network technology to cover, eh?  Looks like 
it's going to be a long weekend.  ; )

No sound card, no monitor. Recording to the various file formats is 
possible, as Herman mentioned.
This seems like an odd limitation to me.  Any idea why it's designed so 
that you must have a sound card to digitally record calls?  They could 
always be moved to another box in order to listen to them.

Your reference picture is fine ... but note that Asterisk can be the 
TDM/VoIP gateway, particularly when Digium releases their DS3 card 
(644 voice channels!) working, a lot more cheaply than a standalone 
box from some hardware vendor.
I'm not sure that the DS3000P is in our timeframe.  I am interested in 
knowing how it will perform, considering more than two Digium quad-span 
cards currently overload the CPU with interrupts.  It seems that Monitor 
cannot handle digitally recording more than ~50 concurrent calls, 
either.  Maybe these limitations are being addressed as we speak.

Thank you for sharing your knowledge with me,
Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
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Re: [Asterisk-Users] IAX2 one way audio

[Duane Cox]

Do you get 2-way audio that sometimes drops off to 1-way audio then picks
back up as 2-way? (Thats what I see)
Not sure if my problem is a lost packet issue as I am sending IAX off net.

Duane Cox


- Original Message - 
From: <[EMAIL PROTECTED]>
To: 
Sent: Friday, April 29, 2005 12:03 PM
Subject: [Asterisk-Users] IAX2 one way audio


> Upgraded one of my asterisk servers to the latest cvs head version last
> nigh now I get one way audio on IAX2 channels when calling other
> asterisk servers. Anyone seeing the some problems?
>
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Re: [Asterisk-Users] Asterisk Hardware Recommendation

[Steve Totaro]

Daniel,
Thanks alot for this post.  You were right on time with valuable 
information.

Thanks again,
Steve
- Original Message - 
From: "Daniel Salama" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, April 29, 2005 12:37 PM
Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation


Sure.
I setup a small lab on a machine with 4 T1s and 36 agents logged in.  The 
system was configured to Monitor all outbound calls as well as  monitor 
all calls distributed by Queue app (monitor-format setting in 
queues.conf).

When recording to local disk, everything was working fine. Agents were 
busy 99.5% and there were at least 30 calls waiting in Queue to be 
distributed. Average call conversation length was about 7.5 minutes.

Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E.
The moment we pushed the load on the Asterisk machine, everything  worked 
for about 40 seconds. Then call quality started suffering  significantly. 
Chopped audio. Bad audio. No audio. Good audio. You  could imagine. So we 
stopped the test.

Then we unmounted the NFS drive and repeated the test again. Everything 
worked fine again.

The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM. 
During all tests, CPU utilization was about 55% on the average (for  each 
CPU). Memory usage was under 1G.

I would say I need to try more troubleshooting. Maybe there was 
congestion on the Fast-E, although preliminary analysis indicates there 
were no CRC errors, collisions, or packet loss.

The NFS machine was completely idle.
Last, we repeated the test over a 1 hour period. This time, Monitor was 
recording on local drives and we were copying files every 15 minutes  with 
a background process (perl script) to NFS mount point. Everything  worked 
fine as well.

I don't know if these tests are conclusive yet. However, from the  results 
so far, I would recommend staying away from recording to NFS  mounted 
point. I will continue running simulations to see if anything  else can be 
identified.

Thanks,
Daniel
On Apr 28, 2005, at 7:26 PM, Matt Roth wrote:
Daniel,
Could you expand upon your experience recording to an NFS mounted  drive.
We are looking to use a TDM-VoIP gateway to route 16+ spans to a  single 
Asterisk server.  We were hoping to Monitor using the following  scheme:

- Monitor application executed on Asterisk server (no 'm' flag)
- Pick a codec that the Monitor application can record in natively so 
that no transcoding is done on the leg files (can this be done?)
- Record the leg files to an NFS mounted drive on a remote machine
- Have soxmix take care of mixing and transcoding the leg files into  the 
desired format on the remote machine

According to you this now looks like a VERY BAD IDEA.
Does anyone out there have any experience using monitor to digitally 
record large numbers of spans (16+) on a single Asterisk server using  a 
VoIP gateway instead of TDM cards?  Is it feasible?  We are trying  to 
keep the Asterisk server as slim as possible, but would like to  stick to 
one box so that we can have centralized queuing,  configuration, and 
reporting.

Has anyone had any luck using Monitor to record to an NFS mounted  drive? 
Are there any other options to remove the overhead of the disk  subsystem 
when recording calls?

Thanks,
Matthew Roth
http://voip-info.org/tiki-index.php? 
page=Running%20Asterisk%20on%20Debian

Daniel Salama wrote:
Thank you again. I will definitely do that. By "cheaper" asterisk 
servers, do you mean single-CPU machines that can handle Quad T1s and 
still do the call monitoring?

BTW, I tried the monitoring without the 'm' option and mounted the 
audio directory via NFS. Big NO NO for everyone. Just do what Matt 
says: copy the -in and -out to archive server separately several  times 
a day :) - don't record to NFS mounted drive.

Thanks,
Daniel
On Apr 28, 2005, at 6:42 PM, mattf wrote:
I have never been able to do more than 50 concurrent recordings with 
Zap ->
SIP phone calls without the audio skipping and/or breaking up. Also, 
if you
are using Digium TE4XXP and want to do a lot of recording I would 
recommend
against a SCSI RAID card because of the interrupt conflicts that you 
will
run into over time. I would recommend a couple of cheaper Asterisk 
servers
with a dual T1 or Quad T1 board in them and SATA drives, with a nice 
big
archive server that the audio will be copied to several times a day. 
Also,
do not record(Monitor) with the 'm' flag on because this will also 
lead to
more disk read-write while you are already trying to write another  100 
or so
streams. Offload the -in and -out files to the archive server and  let 
it
soxmix them together instead. This is the method that we have  settled 
on for
our 12 Asterisk servers and it works rather well for us.

MATT---
-Original Message-
From: Daniel Salama [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 28, 2005 5:56 PM
To: Asterisk Users Mailing List -

Re: [Asterisk-Users] Asterisk Manager interface, setting global vars

[Johann]

There isn't a specific command in the manager API itself to do it.  
However there is a CLI command and you can use the manager command 
action to get the information.  Below is an example, you will need to 
parse the response part to see who is connected.

Action: Command
Command: show manager connected
Response: Follows
 Username IP Address
 something127.0.0.1
As far as I know, there isn't a way to modify or look at the global 
variables directly.  You could make a kludge that would call to a 
special extension that runs NoOp or something that can be seen from an 
Event, but thats not going to be fun.

--johann
Umar Sear wrote:
Hi all, 

Does anyone know of a way to setup global var using the manager interface. 

Basically I want to be able to have multiple manager clients login,
however in a sort of master slave scenario. So the first client that
logs in, sets a global variable which tells subsequent clients at
least one client is already logged in.
The Master would then set additional variables which the slaves would
periodically read.
Is this possible ?
Thanks in advance for any help. 

Umar
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[Asterisk-Users] User events - a dumb question

[Asterisk]

Ok, this is probably stupid question of the week. I have
exten => 888,1,whatever
exten => 888,n,UserEvent(Event|Data)
exten => 888,n,Hangup
If I asterisk -r, when I dial the 888, I see Userevent appearing in the 
console.

However, if I telnet to the * manager using a name and password that has 
the "user" option, that telnet session sees everything but the user event.

What am I missing ?
manager.conf:
[event]
secret=event
read=system,user
write=call,command,agent
Julian
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[Asterisk-Users] IAX2 one way audio

[geek]

Upgraded one of my asterisk servers to the latest cvs head version last
nigh now I get one way audio on IAX2 channels when calling other
asterisk servers. Anyone seeing the some problems? 

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Re: [Asterisk-Users] chan_zap graceful failure

[Andrew Kohlsmith]

On April 29, 2005 12:38 pm, Jeb Campbell wrote:
> While I like the idea (and will look into it -- might need a wait, etc),
> as I said in original post, unloading and reloading did not fix the
> problem.  It took a clean shutdown (unload and restart) to fix the problem.

Hmm; that is odd...

> So regardless of why the card has failed, I would like to discuss making
> chan_zap fail gracefully.  For example if you have a
> Dial(Zap/3/${NUMBER}, and Zap/3 does not exist, asterisk will spit a
> warning (not fail to startup).  However if you have that channel => 3 in
> zapata.conf, chan_zap will fail and prevent asterisk from starting.
>
> I would think that everyone would prefer asterisk to start and have
> parts of the dialplan fail, rather than have asterisk not load at all.

No; if the driver didn't load that's a major problem.  Remember that if the 
channel doesn't exist all the subsequent channels "move up"...  serious 
potential security issues.

I'd rather have the system as it is, where it fails out with an error that is 
easy to understand so I can fix the problem.

-A.
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[Asterisk-Users] Sip endpoints that support re-invite??

[Hamza Moore]

Hi,

I am doing some testing with asterisk using Cisco IP Phones 7960's and
EyeBeam. I have canreinvite=yes on all my devices but the media still
goes through the asterisk box. Does it mean that Cisco and Xten do not
support re-invites? If yes can you recommend SIP phones or adapters
that support re-invites.

Thanks in advance.

Hamza Moore.
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[Asterisk-Users] Problems with MusicOnHold

[Nathan Bowyer]

Greetings,

I have two machines.  One is a P3 Dell Dimension 4100, the other is a
PowerEdge SC420.  Both are running Asterisk 1.0.7, the PowerEdge has a
TE405P card in it, the Dimension has a Digium X100P present (although
not modprobed).  Each machine has mpg123 0.59r loaded, and is using
the exact same set of MP3s for music on hold (both the distributed
ones and some of our own).  Neither box is sharing any interrupts.  I
use the same 7960G to test the Music on Hold.

On the Dimension 4100, MusicOnHold works flawlessly.  No static, no
glitches, nothing.
On the PowerEdge SC420, MusicOnHold has a lot of static, pops,
crackles, and almost everything you can imagine.

I can't think of anything else that is applicable.  Basically, the
machines seem pretty much identical to me.  I expected MoH to work the
same as well, but it isn't.  If anyone has any ideas, please let me
know.

Thanks,
Nathan
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Re: [Asterisk-Users] Asterisk Hardware Recommendation

[Daniel Salama]

Well, I don't think I'm ready to spend that much money :)
I understand your point regarding that load depends on usage. 
SIP_Agents are simply agents answering calls. Average call length would 
be about 8 minutes. During some of these calls (maybe 25%), agents will 
conference the call (PSTN channel) with internal IVR script.

I like Scenario 6. Will look into that further. However, if the above 
information gives you more grounds to make additional comments, please 
do so :)

Thanks,
Daniel
On Apr 29, 2005, at 10:21 AM, mattf wrote:
If price would truly not an option just get one of the Signate 
Telephony
5000 servers(http://www.signate.com/pbx.php) They are about $18,000 and
allow you to have upto 5000 SIP streams go through it. You could have 
that
be your gateway and do the SIP->IAX through that machine and scale 
upto 100
T1s if you want.

But that is a bit steep. So on to your choices. I would really say 
that the
setup you choose will depend on what kind of users you have as well as 
how
often you need to change/add users to the system and how the users are 
using
the system at what times. Any of them that you listed could work 
depending
on how they are used, but in some cases you may not want to use some 
of the
scenarios listed because they would either be incapable of meeting your
needs or overly complex to manage.

The easiest and cheapest one would actually not be listed:
Scenario 6:
Direct SIP->Zap on two separate servers half SIP users on each server
PSTN <--2xT1--> A1 <> SIP_Agents
PSTN <--2xT1--> A2 <> SIP_Agents
There is really no reason to have another 2 servers running IAX to the 
T1
servers, and this is simple and easy to set up and involves only 2 
machines.

The next setup I would recommend would be Scenario 4, although you 
will have
to get a machine with a fast/wide BUS(like an Apple G5) to handle ever
increasing numbers of SIP->IAX streams as the system would grow.

If you can explain more about what kind of use this system will have I 
can
give a better recommendation.

MATT---
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Re: [Asterisk-Users] chan_zap graceful failure

[Jeb Campbell]

Andrew Kohlsmith wrote:
It has nothing to do with not being unloaded; I've seen the wctdm driver fail 
to detect modules correctly.  Run it again and it works just fine.  Some kind 
of minor tweak is in order, I believe.

As an interim solution, your asterisk starup script should try to unload any 
modules and reload them upon asterisk failure...  preferably in a loop:

while(1) {
 unload modules
 sleep 1
 load modules
 start asterisk
 sleep 5
 }
I imagine at this point in time your startup script either does not loop, or 
it doesn't try to unload/load the modules inside the loop.
While I like the idea (and will look into it -- might need a wait, etc), 
as I said in original post, unloading and reloading did not fix the 
problem.  It took a clean shutdown (unload and restart) to fix the problem.

So regardless of why the card has failed, I would like to discuss making 
chan_zap fail gracefully.  For example if you have a 
Dial(Zap/3/${NUMBER}, and Zap/3 does not exist, asterisk will spit a 
warning (not fail to startup).  However if you have that channel => 3 in 
zapata.conf, chan_zap will fail and prevent asterisk from starting.

I would think that everyone would prefer asterisk to start and have 
parts of the dialplan fail, rather than have asterisk not load at all.

As I said, I have not checked the behavior of cvs-head, I just wanted to 
discuss making asterisk more resilient.

Thanks for the tip and I will look into it.
Jeb
--
Jeb Campbell
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk Hardware Recommendation

[Daniel Salama]

Sure.
I setup a small lab on a machine with 4 T1s and 36 agents logged in.  
The system was configured to Monitor all outbound calls as well as  
monitor all calls distributed by Queue app (monitor-format setting in  
queues.conf).

When recording to local disk, everything was working fine. Agents were  
busy 99.5% and there were at least 30 calls waiting in Queue to be  
distributed. Average call conversation length was about 7.5 minutes.

Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E.
The moment we pushed the load on the Asterisk machine, everything  
worked for about 40 seconds. Then call quality started suffering  
significantly. Chopped audio. Bad audio. No audio. Good audio. You  
could imagine. So we stopped the test.

Then we unmounted the NFS drive and repeated the test again. Everything  
worked fine again.

The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM.  
During all tests, CPU utilization was about 55% on the average (for  
each CPU). Memory usage was under 1G.

I would say I need to try more troubleshooting. Maybe there was  
congestion on the Fast-E, although preliminary analysis indicates there  
were no CRC errors, collisions, or packet loss.

The NFS machine was completely idle.
Last, we repeated the test over a 1 hour period. This time, Monitor was  
recording on local drives and we were copying files every 15 minutes  
with a background process (perl script) to NFS mount point. Everything  
worked fine as well.

I don't know if these tests are conclusive yet. However, from the  
results so far, I would recommend staying away from recording to NFS  
mounted point. I will continue running simulations to see if anything  
else can be identified.

Thanks,
Daniel
On Apr 28, 2005, at 7:26 PM, Matt Roth wrote:
Daniel,
Could you expand upon your experience recording to an NFS mounted  
drive.

We are looking to use a TDM-VoIP gateway to route 16+ spans to a  
single Asterisk server.  We were hoping to Monitor using the following  
scheme:

- Monitor application executed on Asterisk server (no 'm' flag)
- Pick a codec that the Monitor application can record in natively so  
that no transcoding is done on the leg files (can this be done?)
- Record the leg files to an NFS mounted drive on a remote machine
- Have soxmix take care of mixing and transcoding the leg files into  
the desired format on the remote machine

According to you this now looks like a VERY BAD IDEA.
Does anyone out there have any experience using monitor to digitally  
record large numbers of spans (16+) on a single Asterisk server using  
a VoIP gateway instead of TDM cards?  Is it feasible?  We are trying  
to keep the Asterisk server as slim as possible, but would like to  
stick to one box so that we can have centralized queuing,  
configuration, and reporting.

Has anyone had any luck using Monitor to record to an NFS mounted  
drive?  Are there any other options to remove the overhead of the disk  
subsystem when recording calls?

Thanks,
Matthew Roth
http://voip-info.org/tiki-index.php? 
page=Running%20Asterisk%20on%20Debian

Daniel Salama wrote:
Thank you again. I will definitely do that. By "cheaper" asterisk  
servers, do you mean single-CPU machines that can handle Quad T1s and  
still do the call monitoring?

BTW, I tried the monitoring without the 'm' option and mounted the  
audio directory via NFS. Big NO NO for everyone. Just do what Matt  
says: copy the -in and -out to archive server separately several  
times a day :) - don't record to NFS mounted drive.

Thanks,
Daniel
On Apr 28, 2005, at 6:42 PM, mattf wrote:
I have never been able to do more than 50 concurrent recordings with  
Zap ->
SIP phone calls without the audio skipping and/or breaking up. Also,  
if you
are using Digium TE4XXP and want to do a lot of recording I would  
recommend
against a SCSI RAID card because of the interrupt conflicts that you  
will
run into over time. I would recommend a couple of cheaper Asterisk  
servers
with a dual T1 or Quad T1 board in them and SATA drives, with a nice  
big
archive server that the audio will be copied to several times a day.  
Also,
do not record(Monitor) with the 'm' flag on because this will also  
lead to
more disk read-write while you are already trying to write another  
100 or so
streams. Offload the -in and -out files to the archive server and  
let it
soxmix them together instead. This is the method that we have  
settled on for
our 12 Asterisk servers and it works rather well for us.

MATT---
-Original Message-
From: Daniel Salama [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 28, 2005 5:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk Hardware Recommendation
Hi,
I've been reading on the wiki as well as on this list, different
suggestions of what to look for when designing an asterisk server  
with
a lot of traffic. By "a lot" of traffic, I mean a box with a a  
TE4XXP,
that will be hit to 

Re: [Asterisk-Users] Recording in a call center

[Dana Olson]

Wouldn't introducing Samba into the mix be even worse?

I would think it would add more processing power and network use to be
constantly writing over the network as opposed to recording on the
same box.

If it's such a critical system, it should have the power to do that,
but that's not the point... If I had such a critical system, I'm not
so sure that I would be saving files in real-time over the network via
Samba.

My question is, what's the difference between writing to the local
disk and over the network? What will happen if the network link goes
down? I've had bad experiences with Samba and NFS both, as far as
connectivity issue handling is concerned.

--
Dana




On 4/29/05, sjaak imap <[EMAIL PROTECTED]> wrote:
> 
> 
> You need something like this ??
> 
> exten => _0.,1,SetVar(CALLFILENAME=${CALLERIDNUM}-${EXTEN}-${TIMESTAMP})
> exten => _0.,2,Monitor(wav,${CALLFILENAME},m)
> exten => _0.,3,Dial,SIP/[EMAIL PROTECTED]
> 
> and mount another server with NFS or SAMBA on /var/spool/asterisk/monitor
> 
> That would be the job.
> 
> 
> Sjaak
> 
> 
> > I would like to record two months of calls.  The call center does not
> > have a huge volume, probably like 60 calls a day and average about 15
> > min a call.  I am using a quad port e1 card from digium.  i would like
> > to record the calls on a seperate server than the one running asterisk
> > to avoid any problems.
> >
> > any ideas?
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Re: [Asterisk-Users] the beginning of voice menu is cutted

[Josiah Bryan]

On Friday 29 April 2005 12:12 pm, Kib Eki wrote:
> Hi,
>
> when I dial  my voicemenu the menu voice is always cutted so that i only
> hear 'word from password.
> What do i have to configure so that i hear the full text from the
> beginning?
>
> thanks, Kib

You might try inserting a Wait in your menu ...e.g...

exten => s,1,Answer ; answer the channel
exten => s,n,Wait(2) ; give the channel time to initalize (2seconds)
exten => s,n,Background(some-recording) 

The 'Wait' supposedly gives the channel time to 'initalize' and get ready to 
send audio. If you start dumping audio ('Background') down a channel not 
initalized, you wont hear anything until the channel is initalized, even if 
the audio has already started.


At least, thats my non-developer-ish understanding of the sequence of events 
after having the same problem myself...

HTH,
-josiah


-- 
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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[Asterisk-Users] the beginning of voice menu is cutted

[Kib Eki]

Hi,
when I dial  my voicemenu the menu voice is always cutted so that i only 
hear 'word from password.
What do i have to configure so that i hear the full text from the beginning?

thanks, Kib
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Re: [Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)

[Matteo Brancaleoni]

yes, some multiplexer allows that, but they're quite expensive
compared to another E1 card for asterisk.
I think you'll need at least 1k $$$ for a such splitter.

Matteo.

Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto:
> Hi,
> 
> Assume I have one E1 digium card to which I want to plug two distinct E1 
> PABXs ,
> one with 15 channels and the other with 15 channels;
> 
> Is there a sort of E1 multiplexer devise that allows me to plug in one hand 
> the
> E1 port of the Digium card and on the other hand the two PABXs? In this same
> devise, I should be able to say that 15 channels need to go to first Interface
> and 15 other channels need to go to other interface.
> 
> Or is it necessary to acquire a another E1 card although I don't need to 
> process
> more channels (30 channels are ok).
> 
> Any help is greatly appreciated.
> 
> 
> 
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-- 
Matteo Brancaleoni
System Administrator
Tel  +39.02.70633354
Sip  [EMAIL PROTECTED]
Iax2 [EMAIL PROTECTED]

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Re: [Asterisk-Users] Asterisk on VMWare ESX/blade servers

[itamar]

I am running on usermodelinux 

Itamar Reis Peixoto
+55 (34) 3238 3845
e-mail : [EMAIL PROTECTED]
http://vps.ispbrasil.com.br --->>> servidores linux
Has anyone had any experience (good or bad) running Asterisk under
VMWare ESX server on a blade chassis?  This application will (fairly
obviously) not include Zap channelsactually, it will be SIP-only.
Please feel free to contact me off-list and I'll summarize for the list
later.
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Re: [Asterisk-Users] Realtime feature

[Matthew Boehm]

Rodrigo P. Telles wrote:

> Does someone knows if the next release of Asterisk (1.0.8?) will have
> Realtime support and when we will have the next Asterisk release
> with Realtime features?

Where is your failure? I don't see anything. The next stable release of
asterisk will be 1.2 and it will have RealTime. The current CVS-HEAD (aka
1.1) has it now and it is very stable.

The eta on 1.2 is unknown. You can help 1.2 along by downloading it and
running it to help fix bugs.

-Matthew

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Re: [Asterisk-Users] chan_zap graceful failure

[Andrew Kohlsmith]

On April 29, 2005 11:22 am, Jeb Campbell wrote:
> As soon as power came back, the server started.  However when it loaded
> wcfxs, port 3 on the card failed the tests (I assume from the module not
> being unloaded before power off).  Because this one port failed the
> test, chan_zap failed to load and asterisk will not start.

It has nothing to do with not being unloaded; I've seen the wctdm driver fail 
to detect modules correctly.  Run it again and it works just fine.  Some kind 
of minor tweak is in order, I believe.

> While the unclean shutdown can be controlled in the future, I have had
> ports go bad and when they do asterisk will not start until the
> offending lines are removed from zapata.conf.  This is not a very
> resilient solution (especially if you are not on site).  I would much
> prefer for asterisk to keep running with what it has got.

As an interim solution, your asterisk starup script should try to unload any 
modules and reload them upon asterisk failure...  preferably in a loop:

while(1) {
 unload modules
 sleep 1
 load modules
 start asterisk
 sleep 5
 }

I imagine at this point in time your startup script either does not loop, or 
it doesn't try to unload/load the modules inside the loop.

 -A.
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