Re: [Asterisk-Users] incoming limit, call_limit, or call-limit?

[Olle E Johansson]

14 mar 2006 kl. 01.10 skrev Damon Estep:

Anyone have any info on the date (or bug tracker number) of the  
change from incominglimit to call-limit, and is it call_limit or  
call-limit?

As documented in sip.conf.sample:
;call-limit=5   ; permit only 5 simultaneous outgoing  
calls to this peer



Does it work with SIP friends?

As documented in sip.conf.sample:

;call-limit=1   ; permit only 1 outgoing call and 1  
incoming call at a time

; from the phone to asterisk
; (1 for the explicit peer, 1 for  
the explicit user,
; remember that a friend equals 1  
peer and 1 user in

; memory)

However, in the test-this-branch version of svn trunk (and the  
"peermatch" branch) I've
changed this functionality so there's one combined call limit for a  
friend/peer, not two

separate limits.

Remember that if you want to support attended transfers, you need at  
least two

simultaneous calls.

/O


---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/



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Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5

[Olle E Johansson]

13 mar 2006 kl. 21.59 skrev Matt:


Hi,
I really want to start using 1.2.5, but I also really need to have the
jitter buffer.  Can anyone offer a suggestion of how to go?   I've
looked at the SIP_JB patch and there seems to be no indication of a
patch for the 1.2.5 release.


Look again. There is a new branch called "jitterbuffer-1.2" that follows
svn HEAD in the 1.2 branch. This is documented in the bug tracker
report for the jitterbuffer :-)

Please test! Thanks!

/O


---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/



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Re: [Asterisk-Users] music on hold without mpg123

[lenz]

Hi Matt,
thank you for your inputs. I updated the page at  
http://www.oinko.net/astrecipes/index.php?n=152 so that it merges in the  
new pointers you sent me.


I also think that converting to the native codec format will be the best  
choice - you can usually trade in some disk space for performance. In any  
case, even having everything as WAV will require only a relatively cheap  
encoding, and not the full mp3 -> slin -> {your codec} encoding plus the  
external process that mpg123 used to require.


I wonder: is there any way to use Asterisk as a transcoding tool, i.e.  
using its internal transcoding capabilities to transcode a given file to  
all its supported codecs? (I'm thinking of g729 and ilbc, things that are  
not supported by Sox but that might be useful in a real-life scenario)


Thank you
l.

In data Mon, 13 Mar 2006 19:29:57 +0100, Matt Roth <[EMAIL PROTECTED]> ha  
scritto:


Lenz,

This method is referred to as file-based or native MOH, and I have some  
additional information regarding it.  First, a short post on why we  
moved from the rawplayer method to native MOH on our production box,  
with a quote from Kevin Fleming regarding the impact the change would  
have on scalability.


 -  
http://lists.digium.com/pipermail/asterisk-users/2006-February/141180.html


Second, I *believe* (please correct me if I'm wrong) that in order to  
get the full benefits of native MOH, the music files should be converted  
to the codec that the calls will be in.  This allows Asterisk to play  
MOH without performing any transcoding, which lowers the resource  
utilization on the box.  Here is a guide for converting WAV files to the  
desired codec, which addresses the four characteristics that describe  
audio data.


- http://lists.digium.com/pipermail/asterisk-users/2006-March/142108.html

Please feel free to add any of this information to your site and don't  
hesitate to contact me if you spot any mistakes.


Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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--
Assum est, versa et manduca.
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Re: [Asterisk-Users] Clustering "NEW THREAD", Almost Working

[Peter Bowyer]

On 14/03/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
>
> DUNDi... pfft... forget it. No docs... it's useless.

How about qualifying these blanket put-downs something like:

"DUNDi - I couldn't find any docs that helped me with it, so I decided
not to invest any more time" ? (Which is an entirely OK position to
take - your call - declaring it as useless without giving it a try,
however, is not really helpful or accurate).

Others are busy finding it very useful indeed.

--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
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[Asterisk-Users] Re: Clustering "NEW THREAD", Almost Working

[JR Richardson]

JR Richardson wrote:
  > I'm very open to suggestions.  I feel like I'm so close but also 
still far
> away.


What happens when your DB flakes out or needs to come down for 
inevitable maintenance?
(no, I am not ripping on MySQL here)

How much Post Dial Delay do you have as your database gets more and more 
utilization?
(yes, I am ripping on realtime here)

What happens if for some reason if a phone gets registered on more than 
one server?

What happens if your single DUNDi server hangs or needs to come down?




Jeremy McNamara



Jeremy,
Good questions.  I'm in the testing and planning phase of this design,
trying to get things ginned up and working properly prior to scaling.  MySQL
will be in a High Availability arrangement, not a single server but the
database will be replicated over several as the read activity from the
registration servers increase load.

MySQL has some cool load balancing techniques that should eliminate post
dial delay as all the data request will be directed across the data cluster,
not just one server.

I've already experienced a phone being registered on 2 servers at once, this
is a product of long registration times and sip peer caching, which I will
try and address by reducing the registration time along with pruning sip
peers and realtime cache for phones that become unreachable.

The single DUNDi Master Peering Server is single for now, just testing, but
I have 2 thoughts there.  1. With this server just being used for lookups,
this function can be monitored and failed over to a hot standby server, the
cache would build back up over time, but call processing would continue with
active lookups.  2.  I could have an active secondary that also peers with
all the registration servers, and all the registration servers peer with the
2 or more peering servers.

The hardest thing here is eliminating the routing loops, mostly due to no
route summarization, not sure how to get around that just yet.  Could use
some help if you have any ideas.

JR



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RE: [Asterisk-Users] Clustering "NEW THREAD", Almost Working

[Douglas Garstang]

Holy crap. You got SIP realtime working? I've tried it twice before and it 
failed the same way twice. Do you have multiple Asterisk boxes accessing the 
same sip info (ie phones) in the same table on the same database? Digium has 
said numerous times this known not to work, although I cant' work out why as 
it's just reading from a common table.

-Original Message- 
From: JR Richardson [mailto:[EMAIL PROTECTED] 
Sent: Mon 3/13/2006 7:11 PM 
To: asterisk-users@lists.digium.com 
Cc: 
Subject: [Asterisk-Users] Clustering "NEW THREAD", Almost Working



All,

I made some progress, but it seems the further I go with clustering the
harder things get.  Hmmm, I guess if it were easy, it would be
documented..

Anyhow, I have 1 * server as the DUNDi peering master with a ttl=1.  The
only function of this server is to lookup where other sip peers are
registered and forward that info on to the requesting * server.

I have 4 * servers accepting registrations from sip users (phones).  
All the
sip phone info is stored in a MySQL database and being accessed through 
the
realtime engine, and it works great.  A phone registers to a server and 
the
server checks the database and if an entry is present, the * servers 
allows
the phone to register and dumps the sip phone into sip show peers, works
great.  I can take the sip entry out of the database and the phone will 
not
resister in realtime.  Works great.

Now the dial plan setup.  All the extension info is also in the MySQL
database, I have a switch statement in the [siptest] context pointing 
to the
database for extension logic.  This also works great.  All servers are
pointing to the same data source with all sip extensions in the database
starting with
exten => 1234,2,Answer and so on
exten => 1235,2,Answer and so on

notice the priority 2 starting point in the database, very important.

This is the good part, in sip.conf, I have regcontext=siptest in the 
general
section (because it doesn't work in the users section), so when a sip 
phone
registers on a server, * dynamically inputs an exten => 1234,1,Noop 
into the
dialplan and immediately the phone is able to be called.  This is 
working
pretty damn well also.

So at this point I have several phones registered across 4 * servers, 
all
pulling their info from MySQL, the same data source.  Now let's say 
phone
1234 and 1235 are registered to server 1 and phone 1236 and 1237 are
registered to server 2, 1234 can call 1235 and vise versa, 1236 can call
1237 and vise versa.

Now from phone 1234 on server 1, I call 1236 on server 2 and because 
1236
does not have a priority 1 entry on server 1, the call progresses to a 
DUNDi
lookup statement in the diaplan logic and request exten 1236 location 
from
the DUNDi peering master server (these registration servers all are 
peered
with the dundi peering master server with a ttl=2, so the request will 
get
past the peering master server and on to the other registration 
servers).
The request is answered from server 2 and 1234 can now complete a call 
to
1236. This is great, all is well, life is good, had a big Dallas 
barbeque
lunch to celebrate because all my sip phones are dynamically 
registering to
any one of 4 sip registration servers, and the other three servers know 
who
is registered where through DUNDi lookups.  And it only took me 2 weeks 
to
get this far.

Now then, let's break it and see what happens, dial any sip phone that 
is
not actively registered and you get an endless DUNDi lookup request 
from all
servers except the one you are dialing from.  I only had one other 
server on
at this time and within seconds produced 590+ IAX trunks initiated back 
into
a registration server before I could hang up the line.

As far as I can tell, if you make a call from server 1, exten 1234 to 
exten
1236, but 1236 is not actively registered on any other server, the other
server will get the DUNDi lookup request and not know where the phone 
is so
it keeps looking up and calling itself to find an extension that is not
there, or something, anyhow it's a bad thing.

Now intrinsically knowing that this protocol is smarter than me, I'm
guessing that I have incorrect dialplan logic that is allowing this to
happen.  I'm wondering how I can set up a dialplan flow that will do 
this:

>From Server 1, pick up phone and dial a number (phone)(exten),
1. * che

RE: [Asterisk-Users] Clustering "NEW THREAD", Almost Working

[Douglas Garstang]

I'm not the person who your replying to, but I'll jump in anyway. We're using 
MySQL in conjunction with AGI. This gives us the ultimate flexibility (and me 
the most freekin work). 
 
Databases are configured in a HA manner. Not sure exactly what form that will 
take yet, but will be either replication or clustering with some mechanism 
hopefully to perform seameless IP address modification upon database failure so 
that it's transparent to Asterisk.
 
Phones can register on any asterisk server. We have a mechanism for ensuring 
that every phone is registered on every box. Every phone can therefore reach 
every other phone through any single Asterisk system. It would have been nice 
if SIP realtime was working properly (it doesn't support having multiple 
Asterisk systems talking to a single database for reasons beyond me). So, 
anyway, for now we're just gonna have to replicate sip.conf between the 
Asterisk boxes.
 
There's various techniques you can implement on MySQL to achieve maximum 
performance as well. We have some rather expensive MySQL consultants help with 
that. :)
 
DUNDi... pfft... forget it. No docs... it's useless.
 
Doug.
 
 

-Original Message- 
From: Jeremy McNamara [mailto:[EMAIL PROTECTED] 
Sent: Mon 3/13/2006 8:34 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: Re: [Asterisk-Users] Clustering "NEW THREAD", Almost Working



JR Richardson wrote:
  > I'm very open to suggestions.  I feel like I'm so close but also
still far
> away.


What happens when your DB flakes out or needs to come down for
inevitable maintenance?
(no, I am not ripping on MySQL here)

How much Post Dial Delay do you have as your database gets more and more
utilization?
(yes, I am ripping on realtime here)

What happens if for some reason if a phone gets registered on more than
one server?

What happens if your single DUNDi server hangs or needs to come down?




Jeremy McNamara
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RE: [Asterisk-Users] Zaptel not compiling on lastest Centos 4.2 kernel.

[Alejandro Kauffmann]

RHEL 4 and therefore CentOS 4 had a bug introduced in the latest kernel.  

https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568

This bug report has a typo as well.  It should read:

#define DEFINE_RWLOCK(x) rwlock_t x = RW__LOCK_UNLOCKED

Fix the line and recompile zaptel.  All should be well.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl O. Pinc
Sent: Monday, March 13, 2006 10:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zaptel not compiling on lastest Centos 4.2
kernel.



On 03/13/2006 11:33:18 AM, Chuck Bunn wrote:
> Hi,
> 
> I made a big mistake on a Centos 4.2 box - I forgot to exclude the
> kernel from updating. Now zaptel will not do a "make linux26" see  
> below. Is there a way to roll this back or is there a patch to get  
> Zaptel to compile? I have a link to the modules using 'ln -s  
> /lib/modules/uname -r/build linux-2.6" so that I did not have to  
> specifiy the kernel version directly.

I successfly compiled zaptel 1.2.1 on Linux 2.6.  Perhaps when you upgraded
the kernel you did not install the corresponding kernel-source rpm?

Karl <[EMAIL PROTECTED]>
Free Software:  "You don't pay back, you pay forward."
  -- Robert A. Heinlein

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-- 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.375 / Virus Database: 268.2.2/280 - Release Date: 3/13/2006


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Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe

[Craig Guy]

Hmm,

I was using 0.3.0 rc24, or the unstable branch.  I see 0.2.0 is listed as 
'stable' so maybe I should have used that.  Please do keep me informed of 
your progress.


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


Sent: Tuesday, March 14, 2006 11:46 AM
Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe



-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of David Phelan
Sent: Tuesday, 14 March 2006 13:28
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one -

maybe


Faxing received by SpanDSP seems to work fine with these units.



From what I understand, receiving should be more sensitive to delays etc

than sending, so it looks like we're onto a winner here!

Thanks for reporting back. I'll post with how my testing goes when my
unit arrives.

james
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Re: [Asterisk-Users] Zaptel not compiling on lastest Centos 4.2 kernel.

[Karl O. Pinc]

On 03/13/2006 11:33:18 AM, Chuck Bunn wrote:

Hi,

I made a big mistake on a Centos 4.2 box - I forgot to exclude the  
kernel from updating. Now zaptel will not do a "make linux26" see  
below. Is there a way to roll this back or is there a patch to get  
Zaptel to compile? I have a link to the modules using 'ln -s  
/lib/modules/uname -r/build linux-2.6" so that I did not have to  
specifiy the kernel version directly.


I successfly compiled zaptel 1.2.1 on Linux 2.6.  Perhaps when you
upgraded the kernel you did not install the corresponding
kernel-source rpm?

Karl <[EMAIL PROTECTED]>
Free Software:  "You don't pay back, you pay forward."
 -- Robert A. Heinlein

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RE: [Asterisk-Users] Clustering

[Michael Collins]

> There's a book on my desk right now that disagrees with you...
> 
> ISBN: 0-596-00962-3

I believe Doug's experience with the TFOT book's DUNDi section was less
than stellar.  If memory serves, it is possible that some of the
examples from the book were out of date.  A few months back there was a
thread full of passionate emails on this subject but as of that time
there weren't a lot of people using DUNDi, or they weren't being very
vocal about their successes.

> 
> Besides, this is Linux.  Sometimes you'll simply have to use the
> internet, right?
> 
> I think you might find more willing ears if you trimmed back your
> negativity just a tad bit.

Probably true!  However, Doug's extreme difficulties with implementing
some of *'s more advanced features has been an educational experience
for many of us on the list.  And some of us find his rants entertaining!
:)

-MC
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RE: [Asterisk-Users] CDR Bug?

[Damon Estep]

Perhaps it is a problem with chan_local

The calls are not all to user agents, the call is to 1 user agent and 1
external number

Dial(sip/###&local/[EMAIL PROTECTED]/n,20)

Another detail is the CDR is mySQL

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Rodgers
Sent: Monday, March 13, 2006 9:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CDR Bug?

Unless I'm reading our CDR data wrong, such calls only generate one  
record for the actual answered call since we started way back on 1.0.9.

Here's a sample record:

"","6044378358","2380","ITS","6044378358","Zap/9-1","SIP/luv- 
c57d","Dial","SIP/luv&SIP/luv-computerroomback&SIP/luv-computerroomfron
t&SIP/luv-itsresourcec","2006-03-03 13:13:36","2006-03-03  
13:13:43","2006-03-03 13:16:02",146,139,"ANSWERED","DOCUMENTATION"

4 UAs are dialed - only one answered the call - only one CDR record.

Hope this helps.

Regards,
-- 
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp



On 13-Mar-06, at 5:32 PM, Damon Estep wrote:

> Trying to figure out if a bug report should be submitted.
>
> Can anyone on 1.2.x verify of this has been corrected?
>
> I am on CVS 8/2005
>
>
>
> If a call comes in to an extension that dials more than one channel  
> (rings at more than one phone) both calls in the CDR show a status  
> of answered when only one is answered, the source channel is  
> bridged to only one of the two destination channels, but both CDRs  
> show answered.
>
>
>
> It looks as if the status is taken from the source channel, not the  
> destination channel.
>
>
>
>
>
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RE: [Asterisk-Users] Dumb question (hang up detection/Zapata.conf)

[Michael Collins]

> Does zapata.conf have any function in systems that aren't using
zaptel(
> I suppose not)?

Just curious - what driver are you using?  (I'm not familiar with
wellgate.)

> 
> I am using an external gateway (wellgate 3701a) and don't have zaptel
> at all.
> 
> If I am not using zapata.conf (this is my guess) then is there some
> other way to configure this type of option for my FXO, or is this
> reliant on the devices setup?

Could be dependent on device setup.  Quick question: what kind of
signaling are you using in zaptel.conf?  Did you happen to try
kewl_start?  

-MC 

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Re: [Asterisk-Users] CDR Bug?

[Anthony Rodgers]

Unless I'm reading our CDR data wrong, such calls only generate one  
record for the actual answered call since we started way back on 1.0.9.


Here's a sample record:

"","6044378358","2380","ITS","6044378358","Zap/9-1","SIP/luv- 
c57d","Dial","SIP/luv&SIP/luv-computerroomback&SIP/luv-computerroomfron
t&SIP/luv-itsresourcec","2006-03-03 13:13:36","2006-03-03  
13:13:43","2006-03-03 13:16:02",146,139,"ANSWERED","DOCUMENTATION"


4 UAs are dialed - only one answered the call - only one CDR record.

Hope this helps.

Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp



On 13-Mar-06, at 5:32 PM, Damon Estep wrote:


Trying to figure out if a bug report should be submitted.

Can anyone on 1.2.x verify of this has been corrected?

I am on CVS 8/2005



If a call comes in to an extension that dials more than one channel  
(rings at more than one phone) both calls in the CDR show a status  
of answered when only one is answered, the source channel is  
bridged to only one of the two destination channels, but both CDRs  
show answered.




It looks as if the status is taken from the source channel, not the  
destination channel.






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RE: [Asterisk-Users] stop monitor on transfer

[Anton Krall]

yep 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Alexander Lopez
|Sent: Monday, March 13, 2006 5:46 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] stop monitor on transfer
|
|Did the receptionist transfer to the 'special' extension? 
|
|> -Original Message-
|> From: [EMAIL PROTECTED]
|> [mailto:[EMAIL PROTECTED] On Behalf Of Anton 
|> Krall
|> Sent: Monday, March 13, 2006 5:09 PM
|> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|> Subject: RE: [Asterisk-Users] stop monitor on transfer
|> 
|> No luck... It stopped  the recording between the recepcion and the 
|> boss when it was been attended transfer but when the call was 
|> transferred to the boss, the recoding continued so in fact 
|you end up 
|> with a recording from where the call came up until the end. :(
|> 
|> |-Original Message-
|> |From: [EMAIL PROTECTED]
|> |[mailto:[EMAIL PROTECTED] On Behalf
|> Of Alexander
|> |Lopez
|> |Sent: Monday, March 13, 2006 3:28 PM
|> |To: Asterisk Users Mailing List - Non-Commercial Discussion
|> |Subject: RE: [Asterisk-Users] stop monitor on transfer
|> |
|> |Try the StopMonitor with the extensions.conf trick below and
|> see if it
|> |works.
|> |[EMAIL PROTECTED]
|> |> [mailto:[EMAIL PROTECTED] On Behalf
|> Of Anton
|> |> Krall
|> |> Subject: RE: [Asterisk-Users] stop monitor on transfer
|> |> 
|> |> Both but mostly attended. 
|> |> 
|> |> |[mailto:[EMAIL PROTECTED] On Behalf
|> |> Of Alexander
|> |> |Lopez
|> |> |Sent: Monday, March 13, 2006 2:50 PM
|> |> |
|> |> | Ae you doing attended transfers or blind?
|> |> |
|> |> |> |Subject: RE: [Asterisk-Users] stop monitor on transfer
|> |> |> |
|> |> |> |Setup a 'non-recording' extension for the oss and transfer
|> |> |> the call to
|> |> |> |that one.
|> |> |> |
|> |> |> |Ie:
|> |> |> |
|> |> |> |7123,1,StopMonitor
|> |> |> |7123,2,Goto(123,1)
|> |> |> |
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|> |
|> |
|> 
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RE: [Asterisk-Users] slinear bandwidth

[Anton Krall]

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Roger Schreiter
|Sent: Monday, March 13, 2006 4:56 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] slinear bandwidth
|
|Anton Krall schrieb:
|> Guys, how much bandwidth does slinear comsume and what 
|quality can it 
|> be compared with? g711, gsm, g729?
|
|
|Hi,
|
|the bandwith is approx double compared to G.711, since it uses 16bit
|(signed) integers, whereas G.711 uses 8bit integers.
|
|The human ear has approximately a logarithmic sensivity, and
|G.711 has a logarithmic resolution. Thus the quality gain of 
|slinear compared to G.711 may not be very much.
|

Might be good for faxing though

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Re: [Asterisk-Users] Woooo, Polycom and * on crack - can'tregister!

[Gabriel Afana]

For anybody that read my post, I got it working again.

*AFTER* the phone decided it could not connect to my primary server and then
failed over to my secondary server (the polycoms can do this), I then had to
unplug the router and then plug it back in.  I have absolutely no idea why
this is.

My phone was still able to contacts the primary server, it just would not
authorize properly.  Maybe when I unplugged my phone and plugged it back
into the router, the router assigned it a new local port number and asterisk
was caching the NAT port and did not authorize when they did not match.

Anybody know why this happened?

- Gabe



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

Sent: Monday, March 13, 2006 5:56 PM
Subject: Re: [Asterisk-Users] W, Polycom and * on crack - can'tregister!


> I just noticed this message with "sip debug" on:
>
>
> Transmitting (no NAT) to 24.50.66.128:5060:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 24.50.66.128:5060;branch=z9hG4bK1da0c8655B1F9600;received=24.50.66.128
> From: "Gabriel Afana" ;tag=2CB88E3F-E17FB2BC
> To: ;tag=as743ca65c
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Max-Forwards: 70
> Contact: 
> WWW-Authenticate: Digest realm="asterisk", nonce="77bfdfcb"
> Content-Length: 0
>
>
> Pretty obvious, there is a problem with the registration info.  However,
my
> sip info didn't change.  Nothing changed.  All my sip.conf info matches my
> 501 exactly (as it did before).  And its not giving me the usual message
on
> the CLI saying there is an unauthorized registration;  I am not seeing
> anything on the CLI.
>
> Any ideas?  this just started happening.
>
> - Gabe
>
>
>
>
> - Original Message - 
> From: "Gabriel Afana" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Monday, March 13, 2006 5:34 PM
> Subject: [Asterisk-Users] W, Polycom and * on crack - can't register!
>
>
> > Hey guys,
> > Got a strange one.  I've been using my Polycom Phone 501 on Asterisk
> for
> > months with no problems like this.  Today I added Polycom 301 on my desk
> > next to my 501 to play with some presences features and some other
things.
> >
> > I got the 301 setup then noticed the 501 wasn't registered anymore.
> > Everything is configured correct (like ususal).  I disconnected the 301,
> > rebooted everything (Asterisk, the 501, my router...etc).  I have a
backup
> > server that the 501 registered to no problem, but it refuses to register
> to
> > my primary server.
> >
> > When I check "sip show peers", it shows (301, 302 and 303 are all
> > extensions on my Polycom 501):
> >
> > 301(Unspecified)D  0UNKNOWN
> > 302(Unspecified)D  0UNKNOWN
> > 303(Unspecified)D  0UNKNOWN
> >
> > However, when I check "sip show channels", things get interesting:
> >
> > support*CLI> sip show channels
> > Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold
Last
> > Message
> > 24.50.66.128 301 64dd711d-6f  00101/1  unkn  No
Rx:
> > SUBSCRIBE
> > 24.50.66.128 (None)  8581f21c-4a  00101/1  unkn  No
Rx:
> > REGISTER
> > 24.50.66.128 (None)  1339e59f-60  00101/1  unkn  No
Rx:
> > REGISTER
> > 24.50.66.128 (None)  f95f1c68-7d  00101/1  unkn  No
Rx:
> > REGISTER
> > 4 active SIP channels
> >
> >
> > support*CLI> sip show channels
> > Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold
Last
> > Message
> > 24.50.66.128 (None)  0e46cbb32cd  00102/0  unkn  No
> Init:
> > OPTIONS
> > 24.50.66.128 (None)  4f4ccf9b-90  00101/1  unkn  No
Rx:
> > REGISTER
> > 24.50.66.128 (None)  1e0b7033-12  00101/1  unkn  No
Rx:
> > REGISTER
> > 24.50.66.128 (None)  380745da-47  00101/1  unkn  No
Rx:
> > REGISTER
> > 4 active SIP channels
> >
> >
> > support*CLI> sip show channels
> > Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold
Last
> > Message
> > 24.50.66.128 (None)  54d4194f-33  00101/1  unkn  No
Rx:
> > REGISTER
> > 24.50.66.128 (None)  8828fe55-54  00101/1  unkn  No
Rx:
> > REGISTER
> > 24.50.66.128 (None)  8d3cde74-8d  00101/1  unkn  No
Rx:
> > REGISTER
> > 3 active SIP channels
> >
> >
> > support*CLI> sip show channels
> > Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold
Last
> > Message
> > 24.50.66.128 (None)  66d9ce2f15c  00102/0  unkn  No
> Init:
> > OPTIONS
> > 24.50.66.128 (None)  54d4194f-33  00101/1  unkn  No
Rx:
> > REGISTER
> > 24.50.66.128 (None)  8828fe55-54  00101/1  unkn  No
Rx:
> > REGISTER
> > 24.50.66.128 (None)  8d3cde74-8d  00101/1  unkn  No
Rx:
> > REGISTER
> > 4 active S

RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe

[James Harper]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of David Phelan
> Sent: Tuesday, 14 March 2006 13:28
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one -
maybe
> 
> Faxing received by SpanDSP seems to work fine with these units.

>From what I understand, receiving should be more sensitive to delays etc
than sending, so it looks like we're onto a winner here!

Thanks for reporting back. I'll post with how my testing goes when my
unit arrives.

james
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Re: [Asterisk-Users] Clustering "NEW THREAD", Almost Working

[Jeremy McNamara]

JR Richardson wrote:
 > I'm very open to suggestions.  I feel like I'm so close but also 
still far

away.



What happens when your DB flakes out or needs to come down for 
inevitable maintenance?

(no, I am not ripping on MySQL here)

How much Post Dial Delay do you have as your database gets more and more 
utilization?

(yes, I am ripping on realtime here)

What happens if for some reason if a phone gets registered on more than 
one server?


What happens if your single DUNDi server hangs or needs to come down?




Jeremy McNamara
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[Asterisk-Users] saydigits

[Jerry Geis]

Thanks, turns out I wasnt calling the application with parenthis
Saydigits(123) is what I needed...

THanks, for the help.

jerry

Jerry Geis wrote:

/ I was searching on voip-info.org for saydigits.

/>/ I see no indication it is not valid in 1.2.4 asterisk.
/>/ however, when trying to use it I get and error "no application saydigits".
/>/ 
/>/ what is the correct way to echo back digits in asterisk 1.2.4?
/>/ 
/>/ I tried "say digits 123" and "saydigits 123" both gave "no application " 
/>/ error
/>/ 
/

Jerry,

I have it on my box:

demo*CLI> show version
Asterisk 1.2.4 built by root @ demo on a i686 running Linux on 2006-02-27 
07:15:32 UTC

demo*CLI> show application saydigits
demo*CLI>
  -= Info about application 'SayDigits' =-

[Synopsis]
Say Digits

[Description]
  SayDigits(digits): This application will play the sounds that correspond
to the digits of the given number. This will use the language that is currently
set for the channel. See the LANGUAGE function for more information on setting
the language for the channel.


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[Asterisk-Users] Clustering "NEW THREAD", Almost Working

[JR Richardson]

All,

I made some progress, but it seems the further I go with clustering the
harder things get.  Hmmm, I guess if it were easy, it would be
documented..
 
Anyhow, I have 1 * server as the DUNDi peering master with a ttl=1.  The
only function of this server is to lookup where other sip peers are
registered and forward that info on to the requesting * server.

I have 4 * servers accepting registrations from sip users (phones).  All the
sip phone info is stored in a MySQL database and being accessed through the
realtime engine, and it works great.  A phone registers to a server and the
server checks the database and if an entry is present, the * servers allows
the phone to register and dumps the sip phone into sip show peers, works
great.  I can take the sip entry out of the database and the phone will not
resister in realtime.  Works great.

Now the dial plan setup.  All the extension info is also in the MySQL
database, I have a switch statement in the [siptest] context pointing to the
database for extension logic.  This also works great.  All servers are
pointing to the same data source with all sip extensions in the database
starting with 
exten => 1234,2,Answer and so on
exten => 1235,2,Answer and so on

notice the priority 2 starting point in the database, very important.

This is the good part, in sip.conf, I have regcontext=siptest in the general
section (because it doesn't work in the users section), so when a sip phone
registers on a server, * dynamically inputs an exten => 1234,1,Noop into the
dialplan and immediately the phone is able to be called.  This is working
pretty damn well also.

So at this point I have several phones registered across 4 * servers, all
pulling their info from MySQL, the same data source.  Now let's say phone
1234 and 1235 are registered to server 1 and phone 1236 and 1237 are
registered to server 2, 1234 can call 1235 and vise versa, 1236 can call
1237 and vise versa.

Now from phone 1234 on server 1, I call 1236 on server 2 and because 1236
does not have a priority 1 entry on server 1, the call progresses to a DUNDi
lookup statement in the diaplan logic and request exten 1236 location from
the DUNDi peering master server (these registration servers all are peered
with the dundi peering master server with a ttl=2, so the request will get
past the peering master server and on to the other registration servers).
The request is answered from server 2 and 1234 can now complete a call to
1236. This is great, all is well, life is good, had a big Dallas barbeque
lunch to celebrate because all my sip phones are dynamically registering to
any one of 4 sip registration servers, and the other three servers know who
is registered where through DUNDi lookups.  And it only took me 2 weeks to
get this far.

Now then, let's break it and see what happens, dial any sip phone that is
not actively registered and you get an endless DUNDi lookup request from all
servers except the one you are dialing from.  I only had one other server on
at this time and within seconds produced 590+ IAX trunks initiated back into
a registration server before I could hang up the line.

As far as I can tell, if you make a call from server 1, exten 1234 to exten
1236, but 1236 is not actively registered on any other server, the other
server will get the DUNDi lookup request and not know where the phone is so
it keeps looking up and calling itself to find an extension that is not
there, or something, anyhow it's a bad thing.

Now intrinsically knowing that this protocol is smarter than me, I'm
guessing that I have incorrect dialplan logic that is allowing this to
happen.  I'm wondering how I can set up a dialplan flow that will do this:

>From Server 1, pick up phone and dial a number (phone)(exten), 
1. * checks to see if the phone is first registered and on-line on server 1
2. if so, dial it, follow standard dialplan login
3. if not, goto DUNDi switch, lookup where it may be
(this is pretty much working good)

On Server 2,
1. DUNDi lookup request comes in
2. check to see if extention is active on this server(2), if not, stop, or
at least don't continue to look for something within your own dialplan that
is not there.

I'm very open to suggestions.  I feel like I'm so close but also still far
away.

Thanks

JR



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Re: [Asterisk-Users] Can One FXO Support Multiple Phone Lines?

[James Ronald]

More then likely he used 4 conductor (two pair) cable and meant 2 lines 
(phone numbers) per cable.  They will need to be broke out to separate 
jacks.

--JR

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


Sent: Monday, March 13, 2006 9:10 PM
Subject: RE: [Asterisk-Users] Can One FXO Support Multiple Phone Lines?


Definitely one line per FXO port, but the wording of the original poster
was two numbers, not two lines, and while it may not be universally
true, "distinctive ring" should allow two (or more) phone numbers to be
present on an FXO port, and asterisk should be able to tell which one is
calling.

If the original poster did mean lines and not numbers, maybe there was
some confusion about the difference between PSTN and ISDN.

James


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Tuesday, 14 March 2006 12:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Can One FXO Support Multiple Phone

Lines?


In a word...No.

One line per FXO port.

Next question?

Michael

--Original Message Text---
From: Andrew Berman
Date: Mon, 13 Mar 2006 19:32:29 -0500

I am currently having our new office wired up with 8 PSTN lines. The

guy

asked me if he could wire it up such that one line had two phone

numbers.

I bought a Sangoma A200 with 8 FXO ports, but now I'm wondering if all

I

needed were 4 FXO ports. Is it possible to set up Asterisk with 2

numbers

per FXO?

Thanks for any help,

Andrew



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

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245

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[Asterisk-Users] Meetme

[John Bittner]

Anyone notice on latest SVN trunk that meetme no longer asks for a pin when
you try to enter a room?
I want to verify it before I post it as a bug.


John Bittner
Simlab.net

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Re: [Asterisk-Users] Australian approved 4BRI PCI adapter preliminarytesting results

[Mark Aufflick]

Sorry - I saw that in an earlier mail after I posted. This list is
pretty high traffic!

On 3/13/06, James Harper <[EMAIL PROTECTED]> wrote:
> Thought I did already, but I've been pretty absent minded recently :)
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Re: [Asterisk-Users] Can One FXO Support Multiple Phone Lines?

[Andrew Berman]

I meant two lines, my bad.  Two lines = two numbers.  In other words, I could've had two lines via one wire plugged into the FXO port.  So, it sounds like that is not possible which is what I thought.Thank you for your help,
AndrewOn 3/13/06, James Harper <[EMAIL PROTECTED]> wrote:
Definitely one line per FXO port, but the wording of the original posterwas two numbers, not two lines, and while it may not be universallytrue, "distinctive ring" should allow two (or more) phone numbers to be
present on an FXO port, and asterisk should be able to tell which one iscalling.If the original poster did mean lines and not numbers, maybe there wassome confusion about the difference between PSTN and ISDN.
James> -Original Message-> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED]] On Behalf Of Michael Graves> Sent: Tuesday, 14 March 2006 12:25> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Can One FXO Support Multiple PhoneLines?>> In a word...No.>> One line per FXO port.>> Next question?>> Michael>
> --Original Message Text---> From: Andrew Berman> Date: Mon, 13 Mar 2006 19:32:29 -0500>> I am currently having our new office wired up with 8 PSTN lines. Theguy> asked me if he could wire it up such that one line had two phone
numbers.> I bought a Sangoma A200 with 8 FXO ports, but now I'm wondering if allI> needed were 4 FXO ports. Is it possible to set up Asterisk with 2numbers> per FXO?>> Thanks for any help,
>> Andrew --> Michael Graves   [EMAIL PROTECTED]> Sr. Product Specialist  
www.pixelpower.com> Pixel Power Inc. [EMAIL PROTECTED]>> o713-861-4005> o800-905-6412
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RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe

[David Phelan]

Faxing received by SpanDSP seems to work fine with these units.
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Phelan
Sent: Tuesday, 14 March 2006 9:00 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe

HI Craig and all that is following this.
I am running a Vanilla 2.6.11 
>From cli, misdn show config

Misdn General-Config:
 ->  VERSION: 0.2.1
 ->  DEBUG_LEVEL: 1  ->  TRACEFILE: not set
 ->  TRACE_CALLS: false  ->  TRACE_DIR: /var/log/
 ->  BRIDGING: no->  STOP_TONE_AFTER_FIRST_DIGIT: yes
 ->  APPEND_DIGITS2EXTEN: yes->  L1_INFO_OK: yes
 ->  CLEAR_L3: no->  DYNAMIC_CRYPT: no
 ->  CRYPT_PREFIX: **->  CRYPT_KEYS: test,muh


So Far, no dropped calls etc
Todays testing will be faxing.

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: Monday, 13 March 2006 6:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe

We went to kernel 2.6.15, and at the same time went to mISDN 0.3.0 rc25, we
haven't had any lockups but users are reporting dropped calls. 
Unfortunately for us this means dropping chan_mISDN in favour of the Cisco
router containing BRI cards and then SIP from the Cisco to Asterisk.  It may
still be possible to use chan_capi with the mISDN drivers for the Drayteks
but for us we've run out of time which is a bit of a bummer.  I believe the
problem is in chan_mISDN which is admittedly still an experimental driver at
this stage with release candidates every few days for the past couple weeks.

I'm still interested to know how you guys get along with these adapters.  As
I said, I think the problem is within chan_mISDN at this stage rather than
in the USB adapters, so maybe using chan_CAPI on top of mISDN hardware
drivers or using chan_vISDN would be the way to go until chan_mISDN matures.

Craig

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

Sent: Monday, March 13, 2006 3:16 PM
Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe


>
> Got my 2 dreytek adapters today...
> Dropped them on to my test system.  After wadding thru my Memory of
how to
> setup mISDN, I had it up and running within about 2 hours.

You might be receiving an email from me shortly then if I get stuck. If it
wasn't for these annoying public holidays (Labour day in Victoria) mine
would probably have arrived today too :)

> Both of them operating in ptmp with no echo cancel turned on at this 
> stage.
> Seems to be happy.

That's quite comforting for initial testing.

Could you try some faxing?

And is there any way to measure latency with some hard figures, maybe by use
of a repeater? Maybe something like this:

Echo measurer -> BRI 1 -> BRI2 -> echo responder.

Where the measurer dials the responder, sends out a ping, and measures the
delay in the response.

I find it hard to believe that any USB induced latency could be measurable
in milliseconds...

> Will drop them onto my local production box next week and see how we
go :D

Let us know!

Thanks

James

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--
No virus found in this incoming message.
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Version: 7.1.375 / Virus Database: 268.2.1/279 - Release Date: 10/03/2006
 

-- 
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Checked by AVG Free Edition.
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[Asterisk-Users] Call Parking Grandstream

[Alfie Viechweg]

I have an asterisk setup using asterisk 1.2.4 and a mix of grandstream 
phones - BT 102 and GXP 200.


The problem I having is with call parking. Isn't it suppose to announce 
the extension where the call is being parked to the person parking the 
call at the time the call is being parked.


How ever in my setup I hear nothing but if I dial the default parked 
extension I get back the call.


Am I missing something? Or should I be using something like ParkAndAnnounce?

Thanks.

  -Alfie
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Re: [Asterisk-Users] priorityjumping=no

[Eric \"ManxPower\" Wieling]

Steve Kennedy wrote:

On Mon, Mar 13, 2006 at 07:38:01PM -0500, Watkins, Bradley wrote:


That depends on what you mean by default.  The supplied sample
extensions.conf contains the priorityjumping=no by default, but if this
parameter is absent then the default is to jump n+101.


OK, that explains it, just wondering why the sample extensions.conf
turns it off, while the O'Reilly Asterisk book and alomst everything you
see on the web uses it ???

I would have thought the default would be to have it on?


The default (i.e. the option not supplied) is to jump priorities (like 
Asterisk 1.0.x did).  That way 1.2 can use extensions.conf for 1.0.


Also, the priority jump won't happen if the priority doesn't exist.
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RE: Spam? Re: [Asterisk-Users] Unknown signalling method 'pri_cpe'

[Hall, Eric M.]

Good eye!

Its getting late maybe I should just stop now


Thank again! 




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Bockman
Sent: Monday, March 13, 2006 8:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [Asterisk-Users] Unknown signalling method 'pri_cpe'

Hall, Eric M. wrote:
>  [chan_zap.so] => (Zapata Telephony)
> Mar 13 20:44:26 ERROR[10829]: chan_zap.c:10598 setup_zap: Unknown 
> signalling method 'pri_cpe'

Follow the correct order in installing Asterisk as shown on the download
page at http://www.asterisk.org

zaptel, libpri, asterisk


Kevin
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RE: [Asterisk-Users] Can One FXO Support Multiple Phone Lines?

[James Harper]

Definitely one line per FXO port, but the wording of the original poster
was two numbers, not two lines, and while it may not be universally
true, "distinctive ring" should allow two (or more) phone numbers to be
present on an FXO port, and asterisk should be able to tell which one is
calling.

If the original poster did mean lines and not numbers, maybe there was
some confusion about the difference between PSTN and ISDN.

James

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Graves
> Sent: Tuesday, 14 March 2006 12:25
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Can One FXO Support Multiple Phone
Lines?
> 
> In a word...No.
> 
> One line per FXO port.
> 
> Next question?
> 
> Michael
> 
> --Original Message Text---
> From: Andrew Berman
> Date: Mon, 13 Mar 2006 19:32:29 -0500
> 
> I am currently having our new office wired up with 8 PSTN lines. The
guy
> asked me if he could wire it up such that one line had two phone
numbers.
> I bought a Sangoma A200 with 8 FXO ports, but now I'm wondering if all
I
> needed were 4 FXO ports. Is it possible to set up Asterisk with 2
numbers
> per FXO?
> 
> Thanks for any help,
> 
> Andrew
> 
> 
> 
> --
> Michael Graves   [EMAIL PROTECTED]
> Sr. Product Specialist  www.pixelpower.com
> Pixel Power Inc. [EMAIL PROTECTED]
> 
> o713-861-4005
> o800-905-6412
> c713-201-1262
> fwd 54245
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Re: [Asterisk-Users] Woooo, Polycom and * on crack - can't register!

[Gabriel Afana]

I just noticed this message with "sip debug" on:


Transmitting (no NAT) to 24.50.66.128:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
24.50.66.128:5060;branch=z9hG4bK1da0c8655B1F9600;received=24.50.66.128
From: "Gabriel Afana" ;tag=2CB88E3F-E17FB2BC
To: ;tag=as743ca65c
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: 
WWW-Authenticate: Digest realm="asterisk", nonce="77bfdfcb"
Content-Length: 0


Pretty obvious, there is a problem with the registration info.  However, my
sip info didn't change.  Nothing changed.  All my sip.conf info matches my
501 exactly (as it did before).  And its not giving me the usual message on
the CLI saying there is an unauthorized registration;  I am not seeing
anything on the CLI.

Any ideas?  this just started happening.

- Gabe




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

Sent: Monday, March 13, 2006 5:34 PM
Subject: [Asterisk-Users] W, Polycom and * on crack - can't register!


> Hey guys,
> Got a strange one.  I've been using my Polycom Phone 501 on Asterisk
for
> months with no problems like this.  Today I added Polycom 301 on my desk
> next to my 501 to play with some presences features and some other things.
>
> I got the 301 setup then noticed the 501 wasn't registered anymore.
> Everything is configured correct (like ususal).  I disconnected the 301,
> rebooted everything (Asterisk, the 501, my router...etc).  I have a backup
> server that the 501 registered to no problem, but it refuses to register
to
> my primary server.
>
> When I check "sip show peers", it shows (301, 302 and 303 are all
> extensions on my Polycom 501):
>
> 301(Unspecified)D  0UNKNOWN
> 302(Unspecified)D  0UNKNOWN
> 303(Unspecified)D  0UNKNOWN
>
> However, when I check "sip show channels", things get interesting:
>
> support*CLI> sip show channels
> Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last
> Message
> 24.50.66.128 301 64dd711d-6f  00101/1  unkn  No   Rx:
> SUBSCRIBE
> 24.50.66.128 (None)  8581f21c-4a  00101/1  unkn  No   Rx:
> REGISTER
> 24.50.66.128 (None)  1339e59f-60  00101/1  unkn  No   Rx:
> REGISTER
> 24.50.66.128 (None)  f95f1c68-7d  00101/1  unkn  No   Rx:
> REGISTER
> 4 active SIP channels
>
>
> support*CLI> sip show channels
> Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last
> Message
> 24.50.66.128 (None)  0e46cbb32cd  00102/0  unkn  No
Init:
> OPTIONS
> 24.50.66.128 (None)  4f4ccf9b-90  00101/1  unkn  No   Rx:
> REGISTER
> 24.50.66.128 (None)  1e0b7033-12  00101/1  unkn  No   Rx:
> REGISTER
> 24.50.66.128 (None)  380745da-47  00101/1  unkn  No   Rx:
> REGISTER
> 4 active SIP channels
>
>
> support*CLI> sip show channels
> Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last
> Message
> 24.50.66.128 (None)  54d4194f-33  00101/1  unkn  No   Rx:
> REGISTER
> 24.50.66.128 (None)  8828fe55-54  00101/1  unkn  No   Rx:
> REGISTER
> 24.50.66.128 (None)  8d3cde74-8d  00101/1  unkn  No   Rx:
> REGISTER
> 3 active SIP channels
>
>
> support*CLI> sip show channels
> Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last
> Message
> 24.50.66.128 (None)  66d9ce2f15c  00102/0  unkn  No
Init:
> OPTIONS
> 24.50.66.128 (None)  54d4194f-33  00101/1  unkn  No   Rx:
> REGISTER
> 24.50.66.128 (None)  8828fe55-54  00101/1  unkn  No   Rx:
> REGISTER
> 24.50.66.128 (None)  8d3cde74-8d  00101/1  unkn  No   Rx:
> REGISTER
> 4 active SIP channels
>
>
>
>
> What the hell?  Any ideas??  I am not getting any errors or
anything
> on the CLI (verbose 100).
>
> - Gabe
>
>
>
> - Original Message - 
> From: "Jerry Geis" <[EMAIL PROTECTED]>
> To: 
> Sent: Monday, March 13, 2006 4:54 PM
> Subject: [Asterisk-Users] saydigits
>
>
> > I was searching on voip-info.org for saydigits.
> > I see no indication it is not valid in 1.2.4 asterisk.
> > however, when trying to use it I get and error "no application
saydigits".
> >
> > what is the correct way to echo back digits in asterisk 1.2.4?
> >
> > I tried "say digits 123" and "saydigits 123" both gave "no application "
> > error
> >
> > Thanks
> > jerry
> > ___
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
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Re: [Asterisk-Users] Unknown signalling method 'pri_cpe'

[Kevin Bockman]

Hall, Eric M. wrote:

 [chan_zap.so] => (Zapata Telephony)
Mar 13 20:44:26 ERROR[10829]: chan_zap.c:10598 setup_zap: Unknown
signalling method 'pri_cpe'


Follow the correct order in installing Asterisk as shown on the download 
page at http://www.asterisk.org


zaptel, libpri, asterisk


Kevin
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[Asterisk-Users] Unknown signalling method 'pri_cpe'

[Hall, Eric M.]

Anyone have any idea what this is talking about.


Here is my zapata.conf
[channels]
switchtype=5ess
signalling=pri_cpe
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
group=1
context=default
musiconhold=default
faxdetect=incoming
channel => 1-23 


Here is my zaptel.conf
span=1,1,0,esf,b8zs
bchan=1-23 # set this to 1-15,17-31 for E1
dchan=24 # set this to 16 for E1

defaultzone=us
loadzone=us

---

Running asterisk in debug give me this!


asterisk -vgcd
Parsing /etc/asterisk/asterisk.conf
Parsing /etc/asterisk/extconfig.conf
Asterisk SVN-trunk-r7498, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>

=
Asterisk Event Logger Started /var/log/asterisk/event_log
Asterisk Dynamic Loader loading preload modules:
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  == Manager registered action Redirect
  == Manager registered action Originate
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxStatus
  == Manager registered action MailboxCount
  == Manager registered action ListCommands
Mar 13 20:44:26 NOTICE[10829]: cdr.c:1166 do_reload: CDR simple logging
enabled.
  == RTP Allocating from port range 1 -> 2
Asterisk PBX Core Initializing
Registering builtin applications:
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
  == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [ExecIfTime]
  == Registered application 'ExecIfTime'
 [Hangup]
  == Registered application 'Hangup'
 [NoOp]
  == Registered application 'NoOp'
 [Progress]
  == Registered application 'Progress'
 [ResetCDR]
  == Registered application 'ResetCDR'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
 [SayAlpha]
  == Registered application 'SayAlpha'
 [SayPhonetic]
  == Registered application 'SayPhonetic'
 [SetAMAFlags]
  == Registered application 'SetAMAFlags'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [Set]
  == Registered application 'Set'
 [ImportVar]
  == Registered application 'ImportVar'
 [Wait]
  == Registered application 'Wait'
 [WaitExten]
  == Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
 [res_musiconhold.so] => (Music On Hold Resource)
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
  == Registered application 'StartMusicOnHold'
  == Registered application 'StopMusicOnHold'
 [res_indications.so] => (Indications Configuration)
-- Registered indication country 'at'
-- Registered indication country 'au'
-- Registered indication country 'br'
-- Registered indication country 'be'
-- Registered indication country 'ch'
-- Registered indication country 'cl'
-- Registered indication country 'cn'
-- Registered indication country 'cz'
-- Registered indication country 'de'
-- Registered indication country 'dk'
-- Registered indication country 'ee'
-- Registered indication country 'es'
-- Registered indication country 'fi'
-- Registered indication country 'fr'
-- Registered indication country 'gr'
-- Registered indication country 'hu'
-- Registered indication country 'it'
-- Registered indication country 'lt'
-- Registered indication country 'mx'
-- Registered indication country 'nl'
-- Registered indication country 'no'
-- Registered indication country 'nz'
-- Registered indication country 'pl'
-- Registered indication country 'pt'
-- Registered indication country 'ru'
-- Registered indication country 'se'
-- Registered indication country 'sg'
-- Registered indication country 'uk'
-- Registered indication country 'us'
-- Registered indication country 'us-o'
-- Registered indication country 'tw'
-- Registered indication country 'za'
-- Setting default indication country to 'us'
  == Registered application 'PlayTones'
  == Registered application 'StopPlayTones'
 [res_agi.so] => (Asterisk Gateway Interface (AGI))
  == Registered application 'DeadAGI'
  == Registered application 'EAGI'
  == Registered application 'AGI'
 [res_odbc.so] => (ODBC Resource)
Mar 13 20:44:26 NOTICE[10829]: res_odbc.c:265 load_odbc_config: Adding
ENV var: INFORMIXSERVER

Re: [Asterisk-Users] saydigits

[El Flynn]

Jerry Geis wrote:

I was searching on voip-info.org for saydigits.
I see no indication it is not valid in 1.2.4 asterisk.
however, when trying to use it I get and error "no application saydigits".

what is the correct way to echo back digits in asterisk 1.2.4?

I tried "say digits 123" and "saydigits 123" both gave "no application " 
error




Jerry,

I have it on my box:

demo*CLI> show version
Asterisk 1.2.4 built by root @ demo on a i686 running Linux on 2006-02-27 
07:15:32 UTC

demo*CLI> show application saydigits
demo*CLI>
  -= Info about application 'SayDigits' =-

[Synopsis]
Say Digits

[Description]
  SayDigits(digits): This application will play the sounds that correspond
to the digits of the given number. This will use the language that is currently
set for the channel. See the LANGUAGE function for more information on setting
the language for the channel.

demo*CLI>

You might want to check if the application is loaded.


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[Asterisk-Users] Woooo, Polycom and * on crack - can't register!

[Gabriel Afana]

Hey guys,
Got a strange one.  I've been using my Polycom Phone 501 on Asterisk for
months with no problems like this.  Today I added Polycom 301 on my desk
next to my 501 to play with some presences features and some other things.

I got the 301 setup then noticed the 501 wasn't registered anymore.
Everything is configured correct (like ususal).  I disconnected the 301,
rebooted everything (Asterisk, the 501, my router...etc).  I have a backup
server that the 501 registered to no problem, but it refuses to register to
my primary server.

When I check "sip show peers", it shows (301, 302 and 303 are all
extensions on my Polycom 501):

301(Unspecified)D  0UNKNOWN
302(Unspecified)D  0UNKNOWN
303(Unspecified)D  0UNKNOWN

However, when I check "sip show channels", things get interesting:

support*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last
Message
24.50.66.128 301 64dd711d-6f  00101/1  unkn  No   Rx:
SUBSCRIBE
24.50.66.128 (None)  8581f21c-4a  00101/1  unkn  No   Rx:
REGISTER
24.50.66.128 (None)  1339e59f-60  00101/1  unkn  No   Rx:
REGISTER
24.50.66.128 (None)  f95f1c68-7d  00101/1  unkn  No   Rx:
REGISTER
4 active SIP channels


support*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last
Message
24.50.66.128 (None)  0e46cbb32cd  00102/0  unkn  No   Init:
OPTIONS
24.50.66.128 (None)  4f4ccf9b-90  00101/1  unkn  No   Rx:
REGISTER
24.50.66.128 (None)  1e0b7033-12  00101/1  unkn  No   Rx:
REGISTER
24.50.66.128 (None)  380745da-47  00101/1  unkn  No   Rx:
REGISTER
4 active SIP channels


support*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last
Message
24.50.66.128 (None)  54d4194f-33  00101/1  unkn  No   Rx:
REGISTER
24.50.66.128 (None)  8828fe55-54  00101/1  unkn  No   Rx:
REGISTER
24.50.66.128 (None)  8d3cde74-8d  00101/1  unkn  No   Rx:
REGISTER
3 active SIP channels


support*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last
Message
24.50.66.128 (None)  66d9ce2f15c  00102/0  unkn  No   Init:
OPTIONS
24.50.66.128 (None)  54d4194f-33  00101/1  unkn  No   Rx:
REGISTER
24.50.66.128 (None)  8828fe55-54  00101/1  unkn  No   Rx:
REGISTER
24.50.66.128 (None)  8d3cde74-8d  00101/1  unkn  No   Rx:
REGISTER
4 active SIP channels




What the hell?  Any ideas??  I am not getting any errors or anything
on the CLI (verbose 100).

- Gabe



- Original Message - 
From: "Jerry Geis" <[EMAIL PROTECTED]>
To: 
Sent: Monday, March 13, 2006 4:54 PM
Subject: [Asterisk-Users] saydigits


> I was searching on voip-info.org for saydigits.
> I see no indication it is not valid in 1.2.4 asterisk.
> however, when trying to use it I get and error "no application saydigits".
>
> what is the correct way to echo back digits in asterisk 1.2.4?
>
> I tried "say digits 123" and "saydigits 123" both gave "no application "
> error
>
> Thanks
> jerry
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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[Asterisk-Users] CDR Bug?

[Damon Estep]

Trying to figure out if a bug report should be submitted.

Can anyone on 1.2.x verify of this has been corrected?

I am on CVS 8/2005

 

If a call comes in to an extension that dials more than one
channel (rings at more than one phone) both calls in the CDR show a status of
answered when only one is answered, the source channel is bridged to only one
of the two destination channels, but both CDRs show answered.

 

It looks as if the status is taken from the source channel,
not the destination channel.

 

 






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Re: [Asterisk-Users] Can One FXO Support Multiple Phone Lines?

[Michael Graves]

In a word...No. 



One line per FXO port.



Next question?



Michael



--Original Message Text---

From: Andrew Berman

Date: Mon, 13 Mar 2006 19:32:29 -0500



I am currently having our new office wired up with 8 PSTN lines.  The guy asked me if he could wire it up such that one line had two phone numbers.  I bought a Sangoma A200 with 8 FXO ports, but now I'm wondering if all I needed were 4 FXO ports.  Is it possible to set up Asterisk with 2 numbers per FXO? 



Thanks for any help,



Andrew











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

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245
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Re: [Asterisk-Users] MWI to 7960's sometimes delayed or lost. Please advise.

[Doug Lytle]

Paul wrote:


I have a large group of 7960G’s running Asterisk 1.2.4 using SIP 7.5. 
Most of the time the message waiting indicator works fine. Once or 
twice a day either an MWI




People have noted the 7.5 SIP image seems to be a little flaky, try 7.4 
or 8.2


Doug

--
Ben Franklin quote:

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


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[Asterisk-Users] saydigits

[Jerry Geis]

I was searching on voip-info.org for saydigits.
I see no indication it is not valid in 1.2.4 asterisk.
however, when trying to use it I get and error "no application saydigits".

what is the correct way to echo back digits in asterisk 1.2.4?

I tried "say digits 123" and "saydigits 123" both gave "no application " 
error


Thanks
jerry
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Re: [Asterisk-Users] priorityjumping=no

[Steve Kennedy]

On Mon, Mar 13, 2006 at 07:38:01PM -0500, Watkins, Bradley wrote:

> That depends on what you mean by default.  The supplied sample
> extensions.conf contains the priorityjumping=no by default, but if this
> parameter is absent then the default is to jump n+101.

OK, that explains it, just wondering why the sample extensions.conf
turns it off, while the O'Reilly Asterisk book and alomst everything you
see on the web uses it ???

I would have thought the default would be to have it on?


Steve

-- 
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Re: [Asterisk-Users] priorityjumping=no

[Gabriel Afana]

Thats up to you.  With priorityjumping=yes, they all will.  If you want to
have jumping *only* for certain calls/priorities, add a "j" in the options
of the dial command:

[inbound-trunk]
exten => 441234123456,1,Dial(SIP/s1a,20,rj)
exten => 441234123456,102,Dial(SIP/s2a,20,rj)
exten => 441234123456,203,Dial(SIP/s1b,20,rj)
exten => 441234123456,304,Dial(SIP/s2a,20,rj)

This should make them jump to n+101 without having to define
priorityjumping=yes and make all calls jump.

- Gabe




- Original Message - 
From: "Steve Kennedy" <[EMAIL PROTECTED]>
To: 
Sent: Monday, March 13, 2006 4:29 PM
Subject: Re: [Asterisk-Users] priorityjumping=no


> On Mon, Mar 13, 2006 at 04:25:20PM -0800, Gabriel Afana wrote:
>
> > I think in 1.2.x, this jumping feature was disabled by default.
>
> So should priorities still increase when the Dial returns busy (i.e.
> jumping to priority + 101)?
>
> Or should something else be done?
>
>
> Steve
>
> -- 
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> UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
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RE: [Asterisk-Users] priorityjumping=no

[Watkins, Bradley]

That depends on what you mean by default.  The supplied sample
extensions.conf contains the priorityjumping=no by default, but if this
parameter is absent then the default is to jump n+101.

Regards,
- Brad

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Afana
Sent: Monday, March 13, 2006 7:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] priorityjumping=no


I think in 1.2.x, this jumping feature was disabled by default.

- Gabe


- Original Message - 
From: "Steve Kennedy" <[EMAIL PROTECTED]>
To: 
Sent: Monday, March 13, 2006 11:19 AM
Subject: [Asterisk-Users] priorityjumping=no


> I've been trying to use a set-up whereby I have several TA's connected 
> to an Asterisk server (1.2.4) and they act like they're in a 
> hunt-group i.e. try the first, if busy jump to the next etc.
> 
> in my extensions.conf I had something like
> [inbound-trunk]
> exten => 441234123456,1,Dial(SIP/s1a,20,r)
> exten => 441234123456,102,Dial(SIP/s2a,20,r)
> exten => 441234123456,203,Dial(SIP/s1b,20,r)
> exten => 441234123456,304,Dial(SIP/s2a,20,r)
> 
> i.e. try the first, if busy try the next etc.
> 
> It seemed to consistently fail.
> 
> in [globals]
> priorityjumping=no
> 
> was set, which came from the samples (i.e. make samples when 
> installing Asterisk).
> 
> I changed that to yes (i.e. priorityjumping=yes) and it started to 
> work.
> 
> If that was the problem (which it seems to be), is that the wrong 
> default? Or am I missing something here completely?
> 
> 
> Steve
> 
> --
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> UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
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Re: [Asterisk-Users] Clustering

[Nick Hoffman]

On Tue March 14 2006 10:15, "Gabriel Afana" <[EMAIL PROTECTED]> wrote:
> Haha, Buuurn.
>
> I have the book on my desk too.  I am going to go step-by-step to setup
> DUDNi.  If I can get it working, I'll post step-by-step details for you
> guys on how to do it yourself.
>
> - Gabe


Hi Gabe. Are you referring to Asterisk - TFOT?
-- Nick
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f: +61 7 5591 6588

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[Asterisk-Users] Can One FXO Support Multiple Phone Lines?

[Andrew Berman]

I am currently having our new office wired up with 8 PSTN lines.  The guy asked me if he could wire it up such that one line had two phone numbers.  I bought a Sangoma A200 with 8 FXO ports, but now I'm wondering if all I needed were 4 FXO ports.  Is it possible to set up Asterisk with 2 numbers per FXO?
Thanks for any help,Andrew
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Re: [Asterisk-Users] priorityjumping=no

[Steve Kennedy]

On Mon, Mar 13, 2006 at 04:25:20PM -0800, Gabriel Afana wrote:

> I think in 1.2.x, this jumping feature was disabled by default.

So should priorities still increase when the Dial returns busy (i.e.
jumping to priority + 101)?

Or should something else be done?


Steve

-- 
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Re: [Asterisk-Users] priorityjumping=no

[Gabriel Afana]

I think in 1.2.x, this jumping feature was disabled by default.

- Gabe


- Original Message - 
From: "Steve Kennedy" <[EMAIL PROTECTED]>
To: 
Sent: Monday, March 13, 2006 11:19 AM
Subject: [Asterisk-Users] priorityjumping=no


> I've been trying to use a set-up whereby I have several TA's connected
> to an Asterisk server (1.2.4) and they act like they're in a hunt-group
> i.e. try the first, if busy jump to the next etc.
> 
> in my extensions.conf I had something like
> [inbound-trunk]
> exten => 441234123456,1,Dial(SIP/s1a,20,r)
> exten => 441234123456,102,Dial(SIP/s2a,20,r)
> exten => 441234123456,203,Dial(SIP/s1b,20,r)
> exten => 441234123456,304,Dial(SIP/s2a,20,r)
> 
> i.e. try the first, if busy try the next etc.
> 
> It seemed to consistently fail.
> 
> in [globals]
> priorityjumping=no
> 
> was set, which came from the samples (i.e. make samples when installing
> Asterisk).
> 
> I changed that to yes (i.e. priorityjumping=yes) and it started to work.
> 
> If that was the problem (which it seems to be), is that the wrong
> default? Or am I missing something here completely?
> 
> 
> Steve
> 
> -- 
> NetTek Ltd  UK mob +44-(0)7775 755503
> UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
> Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
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Re: [Asterisk-Users] Dialplan woes

[Gabriel Afana]

Hmm, I dont have pbx_config.so in my modules.conf file?

Maybe because I have "autoload=yes"

- Gabe

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

Sent: Sunday, March 12, 2006 10:07 PM
Subject: Re: [Asterisk-Users] Dialplan woes


> Hi Gabe,
>
> The issue was because I didn't load pbx_config.so in modules.conf :)
>
> Thanks,
>
> Dave.
>
> On Sun, 2006-03-12 at 13:15 -0800, Gabriel Afana wrote:
> > Hi,
> > After updating your sip.conf and extensions.conf, did you reload
> > asterisk?  Asterisk caches the config files and does not re-read them
unless
> > you issue a "sip reload", "extensions reload" or an all-in-one "restart
when
> > convenient" at the CLI.
> >
> > - Gabe
> >
> > - Original Message - 
> > From: "Dave Hope" <[EMAIL PROTECTED]>
> > To: 
> > Sent: Sunday, March 12, 2006 2:31 AM
> > Subject: [Asterisk-Users] Dialplan woes
> >
> >
> > > Hello all,
> > >
> > > Inspired by the Asterisk talks at FOSDEM 2006, I've decided to give it
a
> > > whirl. I'm having some newbie problems with my dialplan and was
> > > wondering if anyone could be of assistance Smile
> > >
> > > When trying to dial 500, 600 or 601 I get the following notice:
> > >
> > >
> > > pbx.c:1330 pbx_extension_helper: Cannot find extension context
> > > Internal'
> > >
> > >
> > > Any suggestions would be GREATLY appreciated!
> > >
> > > extensions.conf:
> > > [general]
> > > static  =>  yes
> > > writeprotect=>  yes
> > >
> > > [External]
> > > include =>  Internal
> > >
> > > [Internal]
> > > exten   =>  500,1,Dial(SIP/Dave)
> > > exten   =>  600,1,Echo()
> > > exten  =>  601,1,Answer()
> > > exten  =>  601,2,Playback(demo-echotest)
> > > exten  =>  601,3,Echo
> > > exten  =>  601,4,Playback(demo-echodone)
> > > exten  =>  601,5,Hangup
> > >
> > >
> > > sip.conf
> > > [general]
> > > context =   External
> > > srvlookup   =   yes
> > >
> > > [Dave]
> > > type=   friend
> > > ;username   =   Dave
> > > secret  =   dave
> > > host=   dynamic
> > > allow   =   all
> > > context =   Internal
> > >
> > > Thanks for any input,
> > >
> > > Dave.
> > >
> > >
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Re: [Asterisk-Users] Clustering

[Gabriel Afana]

Haha, Buuurn.

I have the book on my desk too.  I am going to go step-by-step to setup
DUDNi.  If I can get it working, I'll post step-by-step details for you guys
on how to do it yourself.

- Gabe


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

Sent: Monday, March 13, 2006 12:13 PM
Subject: RE: [Asterisk-Users] Clustering



There's a book on my desk right now that disagrees with you...

ISBN: 0-596-00962-3

Besides, this is Linux.  Sometimes you'll simply have to use the
internet, right?

I think you might find more willing ears if you trimmed back your
negativity just a tad bit.


Bob McDowell

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Monday, March 13, 2006 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Clustering

Yes, people mention DUNDi ocassionaly. It's a shame it's completely
useless as their is no documentation for it.

-Original Message-
From: Kristian Larsson [mailto:[EMAIL PROTECTED]
Sent: Monday, March 13, 2006 8:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Clustering


On Mon, Mar 13, 2006 at 08:12:40AM -0700, Douglas Garstang wrote:
> Thanks Kristian. It isn't clear how this means a registration on one
Asterisk system magically appear on the other though...
I'm not quite certain as I build my call routing on scripts instead of
Asterisk built in commands, but I beleive Dundi should be able to help
you out in situations like this.

   Kristian
>
> -Original Message-
> From: Kristian Larsson [mailto:[EMAIL PROTECTED]
> Sent: Monday, March 13, 2006 12:22 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Clustering
>
>
> On Fri, Mar 10, 2006 at 08:20:27PM -0700, Douglas Garstang wrote:
> > Kevin,
> >
> > From the voip wiki at
http://www.voip-info.org/wiki-Asterisk+sip+regcontext:
> >
> > "If regcontext is specified, Asterisk will dynamically create and
destroy a NoOp priority 1 extension for a given peer who registers or
unregisters with us"
> Pretend we have peer 123456, then put
>
> exten => 123456,2,Dial(SIP/123456)
>
> in your extensions.conf
> When phone 123456 becomes available and registers to the Asterisk, the

> dialplan will look like:
>
> exten => 123456,1,NoOp
> exten => 123456,2,Dial(SIP/123456)
>
> and as you know the dialplan always begin on priority 1 so if the
> phone is not registered you don't automatically move to priority 2.
>
> What I'm curious to know is whether there is a way to use this with
> SIP RealTime... there doesn't seem to exist a setting for both
> regexten and regcontext. Any pointers?
>
>Kristian.
>
> > What does this mean exactly? How is it used? I've read the same
piece of information dozens of times over the last few months and it
makes as much sense to me today, as it did back then, which is about
zero.
> >
> > Wow... IAX can be used to share registration info? I've never seen
that mentioned anywhere. After reading the patchy docs on DUNDi, I kind
of got the impression that it _might_ be able to do that sort of thing,
but the docs where so bad they where useless. And while we're on the
discussion topic, why doesn't Digium release some docs on DUNDi? It's
their baby after all. It seems to be that almost no one uses it, simply
because there's no docs that explain how to do it.
> >
> > Alternatively, if you don't have time, can you point me to anywhere
where instructions on how to use regcontent is succinctly and clearly
documented and explained?
> >
> > Doug.
> >
> >
> > -Original Message- 
> > From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
> > Sent: Fri 3/10/2006 8:05 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Cc:
> > Subject: Re: [Asterisk-Users] Clustering
> >
> >
> >
> > Douglas Garstang wrote:
> >
> > > I'd just die to see an example of that. I've never seen an
example that actually works. I quite distinctly remember reading
somewhere (sorry, forget where) that this command was broken.
> >
> > It's not broken. If you find some official documentation that
says so,
> > then it needs to be fixed. If you read it somewhere else, then
that
> > source is not something you should trust.
> >
> > regexten in sip.conf works just fine; it can easily be used to
make an
> > extension 'appear' and 'disappear' from the desired context
based on the
> > status of the peer's registration. If that context is then
shared among
> > the Asterisk servers (via DUNDi, IAX2 switches or some other
technique),
> > then calls to that extension will be handled by the server it
registered
> > to automatically.
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> >http://lists.digium

[Asterisk-Users] Simple php script to monitor asterisk calls

[Mojo with Horan & Company, LLC]

Hiya, hope I don't bore anybody with this.  There are certainly a lot of 
monitor-y things out there and they just didn't fit my need, so maybe 
this will fit someone's besides mine.


http://horanappraisals.com/asterisk/pbxmonitor/ contains two files.  one 
is a php script called pbxmonitor, and one is a flat file of extensions 
to extension name mappings of internal users.  It contains example data 
that needs editing to fit your scenario.


so the pbxmonitor.db might have (separated by tabs):
SIP/2000Receptionist
SIP/2001Username 2
SIP/2002Username 3

an internal call might say:
Username 2 talking to Receptionist

an outgoing call might say:
Username 3 talking to 18005551212

an incoming call (already answered) might say:
18005551212 talking to Receptionist

It's pretty self explanatory I guess.  Run it and hope it does stuff.

so, pbxmonitor, in our application, is called from watch, like so:

watch -t -n 1 pbxmonitor

but you could implement it into a refreshing webpage or otherwise parse 
it for your needs.


[sidenote]
We use putty to connect to the asterisk box, and there's an account 
called monitor with a key login instead of password login, and the 
monitor user's .bashrc runs this watch line at startup, followed by an 
exit.  I call tell putty to auto login a username, and via the command 
line, make it load this connection at startup without asking for any 
info, so it's pretty seamless for the end user.  But all that is neither 
here nor there related to my post.

[/sidenote]

I don't have parked calls in yet, but will soon.  I don't have meetme 
conferences in soon, don't know if I will.  It doesn't do non-bridged 
calls yet, this will be soon, as it is important to us.  This should 
give indications of people checking their voicemail, people in echo 
rooms and meetme conferences, and people in IVR things. Not sure what 
else I'll have in it eventually, we'll see.  It's only tested with SIP, 
IAX should work but dunno. I'll post back when I improve it.


Comments, suggestions welcome!

Moj


--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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[Asterisk-Users] DISA & SPA3000 issues

[Alchaemist]

Hi,

These days I run into something quite odd.
I have an [EMAIL PROTECTED] that was modified to meet our requirements.
We have a completely funtional DISA which we use pretty much all the 
time.
I works flawlessly with incomming SIP calls from several providers, 
IAX calls from FWD and with ZAP.

Recently we came out with a situation where it doesn't work... with 
a SPA3000 PSTN Line.
You can call, navigate de IVR, log in into our app, and then when 
you go to DISA, and DISA plays the dialtone... whatever you dial is not 
recognized...

This was REALLY odd... so I made a network capture with Ethereal, 
and... the SPA actually STOPS sending the RTP Events after the second 
dialtone...

To verify this, I created an IVR which played the dialtone, and 
verified that it was true no RTP DTMF events (RFC2833) are sent after 
the SPA listens the second dialtone.

I just reviewed the 87 pages PDF of the SPA3000... and didn't find 
anything about such "feature".
Now I am going to try to figure out if it has something to do with 
the tones recognition of the SPA.
I the meanwhile I had to write a little DISA-like app, based on 
something I found on this forum, without the dialtone.

Did anyone find out anything about this issue before?

REGARDS!!!
Alchaemist

 



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[Asterisk-Users] incoming limit, call_limit, or call-limit?

[Damon Estep]

Anyone have any info on the date (or bug tracker number) of
the change from incominglimit to call-limit, and is it call_limit or
call-limit?

 

Does it work with SIP friends?

 

Running CVS head 8/24 (right before 1.2 release).






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Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time

[Omar A. Sabek]

To be totally honest, I have 7.5 running on many phones and I have yet
to receive a report on a firmware related issue.

Omar

On 3/13/06, Tomislav Parcina <[EMAIL PROTECTED]> wrote:
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Omar A. Sabek
> > Sent: 9. ozujak 2006 18:12
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
> >
> > This issue has been fixed in SIP firmware 7.5
> >
> > Omar A. Sabek
>
> Yes, and I read that SIP 7.5 firmware have some other issues. They recommend 
> using 7.4 firmware. I'm not sure how good in new 8.2 firmware.
>
>
> Tomislav
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Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time

[Omar A. Sabek]

The 8.2 firmware displays Caller ID as @...
this becomes problematic for users that want to dial from their
'Missed Calls' log.

Omar

On 3/13/06, Nathan Bowyer <[EMAIL PROTECTED]> wrote:
> On 3/13/06, Chris Stenton <[EMAIL PROTECTED]> wrote:
> > I have had no issues with 8.2 so far!
> >
> > Chris
> >
>
> Except the Caller ID issue reported in another thread?
>
> > >>
> > >> This issue has been fixed in SIP firmware 7.5
> > >>
> > >> Omar A. Sabek
> > >
> > > Yes, and I read that SIP 7.5 firmware have some other issues. They
> > > recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware.
> > >
> > >
> > > Tomislav
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RE: [Asterisk-Users] stop monitor on transfer

[Alexander Lopez]

Did the receptionist transfer to the 'special' extension? 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Anton Krall
> Sent: Monday, March 13, 2006 5:09 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] stop monitor on transfer
> 
> No luck... It stopped  the recording between the recepcion 
> and the boss when it was been attended transfer but when the 
> call was transferred to the boss, the recoding continued so 
> in fact you end up with a recording from where the call came 
> up until the end. :( 
> 
> |-Original Message-
> |From: [EMAIL PROTECTED]
> |[mailto:[EMAIL PROTECTED] On Behalf 
> Of Alexander 
> |Lopez
> |Sent: Monday, March 13, 2006 3:28 PM
> |To: Asterisk Users Mailing List - Non-Commercial Discussion
> |Subject: RE: [Asterisk-Users] stop monitor on transfer
> |
> |Try the StopMonitor with the extensions.conf trick below and 
> see if it 
> |works.
> |[EMAIL PROTECTED]
> |> [mailto:[EMAIL PROTECTED] On Behalf 
> Of Anton 
> |> Krall
> |> Subject: RE: [Asterisk-Users] stop monitor on transfer
> |> 
> |> Both but mostly attended. 
> |> 
> |> |[mailto:[EMAIL PROTECTED] On Behalf
> |> Of Alexander
> |> |Lopez
> |> |Sent: Monday, March 13, 2006 2:50 PM
> |> |
> |> | Ae you doing attended transfers or blind?
> |> |
> |> |> |Subject: RE: [Asterisk-Users] stop monitor on transfer
> |> |> |
> |> |> |Setup a 'non-recording' extension for the oss and transfer
> |> |> the call to
> |> |> |that one.
> |> |> |
> |> |> |Ie:
> |> |> |
> |> |> |7123,1,StopMonitor
> |> |> |7123,2,Goto(123,1)
> |> |> |
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> |
> 
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Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time

[Nathan Bowyer]

On 3/13/06, Chris Stenton <[EMAIL PROTECTED]> wrote:
> I have had no issues with 8.2 so far!
>
> Chris
>

Except the Caller ID issue reported in another thread?

> >>
> >> This issue has been fixed in SIP firmware 7.5
> >>
> >> Omar A. Sabek
> >
> > Yes, and I read that SIP 7.5 firmware have some other issues. They
> > recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware.
> >
> >
> > Tomislav
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Re: [Asterisk-Users] RE: Delay in ringing

[Ira]

At 08:12 AM 03/13/2006, you wrote:

I am currently experiencing a delay in ringing by around 12 seconds.

Is there something I need to adjust in the dial plan for this?


I had this problem at some point and then fixed it, watch the console 
while it's ringing and see where the call goes in that 12 
seconds.  callerid accounts for 1 ring time delay, but in my case the 
rest was a misspelling or something like that.


Ira 



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[Asterisk-Users] MWI to 7960's sometimes delayed or lost. Please advise.

[Paul]

I have a large group of 7960G’s running Asterisk 1.2.4
using SIP 7.5.  Most of the time the message waiting indicator works
fine.  Once or twice a day either an MWI is not sent to the phone or the
phone misses it.  I have not noticed duplicates being sent and I can’t
figure out the time, if any, a resend is made. Anyone know why I’m
missing, loosing or not sending an MWI to the 7960’s?

 

Short of writing a program to scan the mailboxes for new
messages and bump the count then reduce it to force another MWI does anyone
know how to force MWI’s (such as a notify record moved to the outbound
queue – or a configuration item somewhere in SIP)

 

I want to change the frequency of checking for messages as
well as change the frequency an MWI is resent for the same mailbox.  

 

This seems to be an annoying problem.  Is there a way
to put an outgoing message in the proper format to force the manager to send an
MWI.  What is the format of an MWI notify message for the outgoing queue?  With
this knowledge, I could then write a cron job to scan the mailboxes and move a
notify message in the outgoing queue to generate the MWI.

 

Any help or information would be appreciated.

 

Thanks all..

Paul

 

 

 






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RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe

[David Phelan]

HI Craig and all that is following this.
I am running a Vanilla 2.6.11 
>From cli, misdn show config  

Misdn General-Config:
 ->  VERSION: 0.2.1
 ->  DEBUG_LEVEL: 1  ->  TRACEFILE: not set
 ->  TRACE_CALLS: false  ->  TRACE_DIR: /var/log/
 ->  BRIDGING: no->  STOP_TONE_AFTER_FIRST_DIGIT: yes
 ->  APPEND_DIGITS2EXTEN: yes->  L1_INFO_OK: yes
 ->  CLEAR_L3: no->  DYNAMIC_CRYPT: no
 ->  CRYPT_PREFIX: **->  CRYPT_KEYS: test,muh


So Far, no dropped calls etc
Todays testing will be faxing.

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: Monday, 13 March 2006 6:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe

We went to kernel 2.6.15, and at the same time went to mISDN 0.3.0 rc25, we
haven't had any lockups but users are reporting dropped calls. 
Unfortunately for us this means dropping chan_mISDN in favour of the Cisco
router containing BRI cards and then SIP from the Cisco to Asterisk.  It may
still be possible to use chan_capi with the mISDN drivers for the Drayteks
but for us we've run out of time which is a bit of a bummer.  I believe the
problem is in chan_mISDN which is admittedly still an experimental driver at
this stage with release candidates every few days for the past couple weeks.

I'm still interested to know how you guys get along with these adapters.  As
I said, I think the problem is within chan_mISDN at this stage rather than
in the USB adapters, so maybe using chan_CAPI on top of mISDN hardware
drivers or using chan_vISDN would be the way to go until chan_mISDN matures.

Craig

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

Sent: Monday, March 13, 2006 3:16 PM
Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe


>
> Got my 2 dreytek adapters today...
> Dropped them on to my test system.  After wadding thru my Memory of
how to
> setup mISDN, I had it up and running within about 2 hours.

You might be receiving an email from me shortly then if I get stuck. If
it wasn't for these annoying public holidays (Labour day in Victoria)
mine would probably have arrived today too :)

> Both of them operating in ptmp with no echo cancel turned on at this
> stage.
> Seems to be happy.

That's quite comforting for initial testing.

Could you try some faxing?

And is there any way to measure latency with some hard figures, maybe by
use of a repeater? Maybe something like this:

Echo measurer -> BRI 1 -> BRI2 -> echo responder.

Where the measurer dials the responder, sends out a ping, and measures
the delay in the response.

I find it hard to believe that any USB induced latency could be
measurable in milliseconds...

> Will drop them onto my local production box next week and see how we
go :D

Let us know!

Thanks

James

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Re: [Asterisk-Users] slinear bandwidth

[Roger Schreiter]

Anton Krall schrieb:

Guys, how much bandwidth does slinear comsume and what quality can it be
compared with? g711, gsm, g729?



Hi,

the bandwith is approx double compared to G.711, since it uses 16bit
(signed) integers, whereas G.711 uses 8bit integers.

The human ear has approximately a logarithmic sensivity, and
G.711 has a logarithmic resolution. Thus the quality gain of
slinear compared to G.711 may not be very much.


Roger.


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Re: [Asterisk-Users] Professional Recordings

[Martinez Felix]

Our friend record all the annoucements, so we dont have disparity problems...

She can also record the annoucements in Spanish...On 3/13/06, Alexander Lopez <[EMAIL PROTECTED]> wrote:
 Placing plasic bag over ones head whist listnting to Allison Promptsmakes the 'breathy-ness' go away.
> -Original Message-> From: [EMAIL PROTECTED]> [mailto:
[EMAIL PROTECTED]] On Behalf Of Matt> Sent: Monday, March 13, 2006 12:46 PM>> While I like Allison... sometimes she just sounds a little> too breathy for my liking.
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[Asterisk-Users] (no subject)

[Hector medina]

 
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[Asterisk-Users] slinear bandwidth

[Anton Krall]

Guys, how much bandwidth does slinear comsume and what quality can it be
compared with? g711, gsm, g729?

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RE: [Asterisk-Users] stop monitor on transfer

[Anton Krall]

No luck... It stopped  the recording between the recepcion and the boss when
it was been attended transfer but when the call was transferred to the boss,
the recoding continued so in fact you end up with a recording from where the
call came up until the end. :( 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Alexander Lopez
|Sent: Monday, March 13, 2006 3:28 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] stop monitor on transfer
|
|Try the StopMonitor with the extensions.conf trick below and 
|see if it works. 
|[EMAIL PROTECTED] 
|> [mailto:[EMAIL PROTECTED] On Behalf Of Anton 
|> Krall
|> Subject: RE: [Asterisk-Users] stop monitor on transfer
|> 
|> Both but mostly attended. 
|> 
|> |[mailto:[EMAIL PROTECTED] On Behalf
|> Of Alexander
|> |Lopez
|> |Sent: Monday, March 13, 2006 2:50 PM
|> |
|> | Ae you doing attended transfers or blind?
|> |
|> |> |Subject: RE: [Asterisk-Users] stop monitor on transfer
|> |> |
|> |> |Setup a 'non-recording' extension for the oss and transfer
|> |> the call to
|> |> |that one.
|> |> |
|> |> |Ie:
|> |> |
|> |> |7123,1,StopMonitor
|> |> |7123,2,Goto(123,1)
|> |> |
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Re: RE : [Asterisk-Users] Voice problem

[Gabriel Afana]

Ooops.  My mistake.  I was thinking of a VPN.  Too many late-night reading
sessions...all the acronyms begin to blend together :-)

- Gabe


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

Sent: Monday, March 13, 2006 3:19 AM
Subject: Re: RE : [Asterisk-Users] Voice problem


> On Sun, 2006-03-12 at 13:33 -0800, Gabriel Afana wrote:
> > Andrew,
> > From what I've read, ISDN is *not* a very good platform for VoIP
> > because it introduces a great deal of latency and jitter.  Latency
> > will cause communication to be difficult.  Jitter will cause the calls
> > to be choppy sounding.
>
> Where did you get that idea? ISDN is a digital TDM technology and as
> such does not have jitter and negligible latency (read up on TDM). ISDN
> and VoIP don't have anything to do with each other other than that an
> Asterisk box might be a SIP/IAX2 <--> PSTN gateway using Basic Rate of
> Primary Rate ISDN on the trunk side on the Asterisk box.
>
> Regards,
> Patrick
>
>
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[Asterisk-Users] Seperate music on hold for SIP extensions

[james.texter]

I have a requirement to play different hold messages depending upon the 
extension that originated the call.  I noticed a musicclass setting in 
sip.conf, but it appears this is global.  I tried setting this on all of my 
individual extensions, but it didn't have any affect.  Is there a way to 
achieve this, either through sip.conf or in the dial plan?

Thanks,

James

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RE: [Asterisk-Users] OT call recording (was stop monitor on transfer)

[Bob McDowell]

I'd wager that would depended on whether you were in front of a state
judge or a federal one...

In general, I think no.  I think the smaller the government body, the
more restrictive it can be.  Unless directly contradicted by a federal
law, anyway.


Bob McDowell

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Monday, March 13, 2006 3:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT call recording (was stop monitor on
transfer)

Martin Joseph wrote:
>
> On Mar 13, 2006, at 12:00 PM, Bob McDowell wrote:
>
>>
>> It depends
>>
>> http://www.callcorder.com/phone-recording-law-america.htm
>>
> Thanks for the info!
>
> 12 states require, under most circumstances, the consent of all
> parties to a conversation. Those jurisdictions are California,
> Connecticut, Florida, Illinois, Maryland, Massachusetts, Michigan,
> Montana, Nevada, New Hampshire, Pennsylvania and Washington.
>

Do Federal rules trump state rules?

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Re: [Asterisk-Users] Need help implementing call center features ofAsterisk

[pdhales]

Where are you located?

We are in Melbourne, Australia.

regards

PaulH

- Original Message - 
From: "Naren Koka" <[EMAIL PROTECTED]>
To: 
Sent: Monday, March 13, 2006 11:25 PM
Subject: [Asterisk-Users] Need help implementing call center features
ofAsterisk


> I am looking for help in implementing call center on Asterisk server.
> How can we implement predictive dialing? How does it communicate with
> a CRM system?  Are there consultants who can help us setup the system?
>
> Thank you,
> Naren
>
>
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[Asterisk-Users] cisco 7912 ringlist.dat file format

[Zachary C. Whitley]

Anyone know what the file format for the cisco 7912 RINGLIST.DAT file
is?

I have a RINGLIST.DAT file for a 7960 that is working fine. It's just a
list of the ring name and the file name but it doesn't work for the
7912. I've read somewhere that the 7912 needs the ringlist.dat file to
be in xml but I can't figure out how exactly it is constructed.

Thanks.
-- 

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[Asterisk-Users] cisco 7912 ringlist.dat file format

[Zachary C. Whitley]

Anyone know what the file format for the cisco 7912 RINGLIST.DAT file
is?

I have a RINGLIST.DAT file for a 7960 that is working fine. It's just a
list of the ring name and the file name but it doesn't work for the
7912. I've read somewhere that the 7912 needs the ringlist.dat file to
be in xml but I can't figure out how exactly it is constructed.

Thanks.
-- 


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[Asterisk-Users] Phase locked mode

[Aaron Daniel]

Anyone have a hint on what this means:
Mar 13 15:14:58 DEBUG[16772] channel.c: Generator got voice, switching 
to phase locked mode


Just had a user call me and tell me that several times through a 
conversation, the line would just cut out for a little bit and then come 
back, the person on the other end had no clue it was happening.  This is 
what showed in the logs a couple times throughout the conversation, so 
I'm thinking it's the culprit, just don't know what it is exactly.


Aaron
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Re: [Asterisk-Users] OT call recording (was stop monitor on transfer)

[Rich Adamson]

Martin Joseph wrote:


On Mar 13, 2006, at 12:00 PM, Bob McDowell wrote:



It depends

http://www.callcorder.com/phone-recording-law-america.htm


Thanks for the info!

12 states require, under most circumstances, the consent of all parties 
to a conversation. Those jurisdictions are California, Connecticut, 
Florida, Illinois, Maryland, Massachusetts, Michigan, Montana, Nevada, 
New Hampshire, Pennsylvania and Washington.




Do Federal rules trump state rules?

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RE: [Asterisk-Users] stop monitor on transfer

[Alexander Lopez]

Try the StopMonitor with the extensions.conf trick below and see if it
works. 
[EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Anton Krall
> Subject: RE: [Asterisk-Users] stop monitor on transfer
> 
> Both but mostly attended. 
> 
> |[mailto:[EMAIL PROTECTED] On Behalf 
> Of Alexander 
> |Lopez
> |Sent: Monday, March 13, 2006 2:50 PM
> |
> | Ae you doing attended transfers or blind?
> |
> |> |Subject: RE: [Asterisk-Users] stop monitor on transfer
> |> |
> |> |Setup a 'non-recording' extension for the oss and transfer
> |> the call to
> |> |that one.
> |> |
> |> |Ie:
> |> |
> |> |7123,1,StopMonitor
> |> |7123,2,Goto(123,1)
> |> |
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[Asterisk-Users] Calls not tearing down properly

[McQuiggan, Mark xt46480]

I have the 
following configuration:
 
Bell CO. 
<---ISDN---> Definity <---ZAP PRI--->  Asterisk  <--ZAP 
PRI---> Nortel BCM.
 
With SIP 
users on the Asterisk box.
 
I have found 
a problem with call tear-downs.  When a caller calls in from Outside, or 
from a Definity station to a BCM station, _and_ the BCM hangs up first, the 
Definity - Asterisk call does not tear down properly.  I get a problem tone 
on the Definity station, or a "not in service" message from Bell for outside 
callers.
 
All other 
calls tear down properly, in either direction, no matter who calls or who hangs 
up first.  
 
I have been 
having problems with my Definity Asterisk connection dropping its D-channel (and 
all connections existing at the time of the D-channel drop).  I believe 
that they may be linked.  In using gdb, I was seeing problems with 
pri_disconnect_timeout.
 
Please 
help.
 
Regards,
 
Mark.
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RE: [Asterisk-Users] Professional Recordings

[Alexander Lopez]

 Placing plasic bag over ones head whist listnting to Allison Prompts
makes the 'breathy-ness' go away.


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Monday, March 13, 2006 12:46 PM
> 
> While I like Allison... sometimes she just sounds a little 
> too breathy for my liking.
> 
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RE: [Asterisk-Users] stop monitor on transfer

[Anton Krall]

Both but mostly attended. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Alexander Lopez
|Sent: Monday, March 13, 2006 2:50 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] stop monitor on transfer
|
| Ae you doing attended transfers or blind?
|
|
|> -Original Message-
|> From: [EMAIL PROTECTED]
|> [mailto:[EMAIL PROTECTED] On Behalf Of Anton 
|> Krall
|> Sent: Monday, March 13, 2006 10:50 AM
|> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|> Subject: RE: [Asterisk-Users] stop monitor on transfer
|> 
|> Ive tried that via agis and that doesn't seem to work because the 
|> stopmonitor is applied to the call between the receptionist and the 
|> boss not the original call between caller and reception 
|which is later 
|> transferred to the boss.
|>  
|> 
|> |-Original Message-
|> |From: [EMAIL PROTECTED]
|> |[mailto:[EMAIL PROTECTED] On Behalf
|> Of Alexander
|> |Lopez
|> |Sent: Monday, March 13, 2006 9:25 AM
|> |To: Asterisk Users Mailing List - Non-Commercial Discussion
|> |Subject: RE: [Asterisk-Users] stop monitor on transfer
|> |
|> |Setup a 'non-recording' extension for the oss and transfer
|> the call to
|> |that one.
|> |
|> |Ie:
|> |
|> |7123,1,StopMonitor
|> |7123,2,Goto(123,1)
|> |
|> | 
|> |
|> |> -Original Message-
|> |> From: [EMAIL PROTECTED]
|> |> [mailto:[EMAIL PROTECTED] On Behalf
|> Of Adrian
|> |> Carter
|> |> Sent: Monday, March 13, 2006 1:35 AM
|> |> To: Asterisk Users Mailing List - Non-Commercial Discussion
|> |> Subject: Re: [Asterisk-Users] stop monitor on transfer
|> |> 
|> |> I'd teach the boss to appreciate recorded calls and just
|> ensure they
|> |> are secure.
|> |> 
|> |> I know mine actually loves that his calls are recorded - not many 
|> |> people counter-claim or argue about conversations once you
|> can trot
|> |> out them actually making the statement they claim they 
|never did...
|> |> *shrug*
|> |> 
|> |> horses for courses I guess - but other than the obvious (make em 
|> |> appreciate and embrace rather than shun and dismiss) im not
|> |sure what
|> |> you could do - Maybe just running stopmonitor again will stop the 
|> |> first recording ? try just calling it twice on those calls
|> |> 
|> |> Anton Krall wrote:
|> |> > Guys.
|> |> >
|> |> > This idea has been banging my headfor days now and I feel
|> |> the need to
|> |> > share with you.
|> |> >
|> |> > Imagine this scenario: all calls come in thru a
|> |> receptionist, asterisk
|> |> > records all incoming calls, the receptionist's work is to
|> |> transfer the
|> |> > calls to internal people but some of them are bosses and
|> |> you know how
|> |> > bosses are, they don't want their calls to be recorded,
|> so, I have
|> |> > been trying to figure a way on how to stop monitoring /
|> |> recoring calls
|> |> > once they are transferred to a bosses extension while othe
|> |> transferd
|> |> > to other people stay on record mode.
|> |> >
|> |> > Anybody has done this or know of a way? 
|> |> >
|> |> > I tried with stopmonitor but stopmonitor will stop
|> |> recording the call
|> |> > between the receptionist and the boss but once the call is
|> |> transferred
|> |> > and since the initial call come thru the recepcionist, the
|> |> call stays on record.
|> |> >
|> |> > What do you think guys?
|> |> >
|> |> > ___
|> |> > --Bandwidth and Colocation provided by Easynews.com --
|> |> >
|> |> > Asterisk-Users mailing list
|> |> > To UNSUBSCRIBE or update options visit:
|> |> >http://lists.digium.com/mailman/listinfo/asterisk-users
|> |> >
|> |> >   
|> |> 
|> |> --
|> |> Adrian Carter
|> |> Technical Manager
|> |> Leading Edge Internet
|> |> 
|> |> Web http://www.lei.net.au http://support.lei.net.au
|> |> Direct+61 2 6163 6162  Support 1 300 662 415
|> |> E-mail[EMAIL PROTECTED]
|> |> ___
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|> |> 
|> |> Asterisk-Users mailing list
|> |> 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
|> 
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|
|

_

[Asterisk-Users] SIP Jitter Buffer for 1.2.5

[Matt]

Hi,
I really want to start using 1.2.5, but I also really need to have the
jitter buffer.  Can anyone offer a suggestion of how to go?   I've
looked at the SIP_JB patch and there seems to be no indication of a
patch for the 1.2.5 release.
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Re: [Asterisk-Users] OT call recording (was stop monitor on transfer)

[Martin Joseph]

On Mar 13, 2006, at 12:00 PM, Bob McDowell wrote:



It depends

http://www.callcorder.com/phone-recording-law-america.htm


Thanks for the info!

12 states require, under most circumstances, the consent of all parties 
to a conversation. Those jurisdictions are California, Connecticut, 
Florida, Illinois, Maryland, Massachusetts, Michigan, Montana, Nevada, 
New Hampshire, Pennsylvania and Washington.


I live in washington  ;~)

Marty

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RE: [Asterisk-Users] stop monitor on transfer

[Alexander Lopez]

 Ae you doing attended transfers or blind?


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Anton Krall
> Sent: Monday, March 13, 2006 10:50 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] stop monitor on transfer
> 
> Ive tried that via agis and that doesn't seem to work because 
> the stopmonitor is applied to the call between the 
> receptionist and the boss not the original call between 
> caller and reception which is later transferred to the boss.
>  
> 
> |-Original Message-
> |From: [EMAIL PROTECTED]
> |[mailto:[EMAIL PROTECTED] On Behalf 
> Of Alexander 
> |Lopez
> |Sent: Monday, March 13, 2006 9:25 AM
> |To: Asterisk Users Mailing List - Non-Commercial Discussion
> |Subject: RE: [Asterisk-Users] stop monitor on transfer
> |
> |Setup a 'non-recording' extension for the oss and transfer 
> the call to 
> |that one.
> |
> |Ie:
> |
> |7123,1,StopMonitor
> |7123,2,Goto(123,1)
> |
> | 
> |
> |> -Original Message-
> |> From: [EMAIL PROTECTED]
> |> [mailto:[EMAIL PROTECTED] On Behalf 
> Of Adrian 
> |> Carter
> |> Sent: Monday, March 13, 2006 1:35 AM
> |> To: Asterisk Users Mailing List - Non-Commercial Discussion
> |> Subject: Re: [Asterisk-Users] stop monitor on transfer
> |> 
> |> I'd teach the boss to appreciate recorded calls and just 
> ensure they 
> |> are secure.
> |> 
> |> I know mine actually loves that his calls are recorded - not many 
> |> people counter-claim or argue about conversations once you 
> can trot 
> |> out them actually making the statement they claim they never did...
> |> *shrug*
> |> 
> |> horses for courses I guess - but other than the obvious (make em 
> |> appreciate and embrace rather than shun and dismiss) im not
> |sure what
> |> you could do - Maybe just running stopmonitor again will stop the 
> |> first recording ? try just calling it twice on those calls
> |> 
> |> Anton Krall wrote:
> |> > Guys.
> |> >
> |> > This idea has been banging my headfor days now and I feel
> |> the need to
> |> > share with you.
> |> >
> |> > Imagine this scenario: all calls come in thru a
> |> receptionist, asterisk
> |> > records all incoming calls, the receptionist's work is to
> |> transfer the
> |> > calls to internal people but some of them are bosses and
> |> you know how
> |> > bosses are, they don't want their calls to be recorded, 
> so, I have 
> |> > been trying to figure a way on how to stop monitoring /
> |> recoring calls
> |> > once they are transferred to a bosses extension while othe
> |> transferd
> |> > to other people stay on record mode.
> |> >
> |> > Anybody has done this or know of a way? 
> |> >
> |> > I tried with stopmonitor but stopmonitor will stop
> |> recording the call
> |> > between the receptionist and the boss but once the call is
> |> transferred
> |> > and since the initial call come thru the recepcionist, the
> |> call stays on record.
> |> >
> |> > What do you think guys?
> |> >
> |> > ___
> |> > --Bandwidth and Colocation provided by Easynews.com --
> |> >
> |> > Asterisk-Users mailing list
> |> > To UNSUBSCRIBE or update options visit:
> |> >http://lists.digium.com/mailman/listinfo/asterisk-users
> |> >
> |> >   
> |> 
> |> --
> |> Adrian Carter
> |> Technical Manager
> |> Leading Edge Internet
> |> 
> |> Web  http://www.lei.net.au http://support.lei.net.au
> |> Direct+61 2 6163 6162  Support 1 300 662 415
> |> E-mail[EMAIL PROTECTED]
> |> ___
> |> --Bandwidth and Colocation provided by Easynews.com --
> |> 
> |> Asterisk-Users mailing list
> |> To UNSUBSCRIBE or update options visit:
> |>http://lists.digium.com/mailman/listinfo/asterisk-users
> |> 
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> |
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> |To UNSUBSCRIBE or update options visit:
> |   http://lists.digium.com/mailman/listinfo/asterisk-users
> |
> 
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Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?

[Aaron Daniel]

We rolled back to 7.4 cause of that too.  7.5 has a strange bug where if 
the server loses connection, the phone's just don't try re-registering.


Aaron

Tim Connolly wrote:
Just curious, why not 7.5 ? 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Monday, March 13, 2006 2:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?


I'm using P0S3-08-2-00.. I noticed the callerID started showing
up 
with the number, then @... So the callerID on the phone 
looks like: [EMAIL PROTECTED] which of course is logged in the 
missed calls exactly like that, and completely foobars the dialing 
string if you try to dial a missed call by simply hitting the dial 
button. Can anyone else verify this problem?


Yeah, that bothered me so I rolled back to SIP 7.4.

Nabeel

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RE: [Asterisk-Users] Looking for docs on adjusting txgain/rxgain

[Bob McDowell]

Are these the droids you're looking for?:

http://lists.digium.com/pipermail/asterisk-users/2004-November/064312.ht
ml

I have corrected/edited the entry in the wiki.

Also, is Kris Boutilier still around?  Can anyone verify if this
information has signifigantly changed in the last 18 months?


Thanks,

Bob McDowell

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert La
Ferla
Sent: Sunday, March 12, 2006 1:09 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Looking for docs on adjusting txgain/rxgain

I am looking for docs on how to diagnose and adjust the rx/tx gain in
zapata.conf.  The wiki has a link to this article but it no longer
exists on the server.

http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.ht
ml

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[Asterisk-Users] Outgoing calls via Sipgate

[Dave Hope]

Hello all, 

With some help from people in #asterisk on freenode, I've managed to get
incoming SIP calls working. 

Outgoing calls however are however a different matter. My whole working
(incoming calls only) SIPgate configuration can be found here. [1]

When I uncommon what's in there, nothing works.  There doesn't appear to
be any useful error being logged , even when debug is enabled for
console and file logs.

If anyone could take a look and show me what needs adding in order for
outgoing calls to work, that would be superb!

My long term goal is to get asterisk running at home, and then persuade
the boss to ditch the Avaya setup we have at the office. But since I'd
likely be the one implementing it, I want to try and get something
working before I commit myself :)

Thanks!, 

Dave.

[1] http://files.davehope.co.uk/home.tar

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Re: [Asterisk-Users] Failed installing zaptel

[Rich Adamson]

Chuck Bunn wrote:

Hi,

I am having the same exact problem. I am assuming that it was a problem 
with a kernel update I did. I am in the process of rolling back to an 
older kernel... I will let you let know if this works. There is also a 
patch for zaptel but I believe this is for going from 1.3 to 1.4?


If memory serves correctly, I think someone submitted a change to a 
makefile to handle an issue with recent kernels. Thought it was related 
to fc4 (or something like that), but might be what you're looking for.


Think the change was submitted this weekend.

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[Asterisk-Users] Re: Regexten & Regcontext, working now

[JR Richardson]

Just figured it out, I think.  I put regcontext=mycontext into the [general] 
section in sip.conf instead of the the [user] section and when the sip user 
registered, the NoOp extension priority 1 came right up in the dial plan.

All is well again, so far.

Clarity of sight becomes infinitely greater with head removed from rectum.


>> 
> Hi All,
> 
> I've been trying to get regexten and regcontext going for some sip peers but 
> following the examples on the wiki is not working, as far as I can tell, 
> nothing is happening.  the phone registers, sip show peers is ok, but the 
> NoOp priority 1 extension never gets created or added to the dialplan.  Has 
> anyone got this working?
> 
> Thanks.
> 
> JR
> 
> JR Richardson
> Engineering for the Masses
> 


JR Richardson
Engineering for the Masses

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[Asterisk-Users] priorityjumping=no

[Steve Kennedy]

I've been trying to use a set-up whereby I have several TA's connected
to an Asterisk server (1.2.4) and they act like they're in a hunt-group
i.e. try the first, if busy jump to the next etc.

in my extensions.conf I had something like
[inbound-trunk]
exten => 441234123456,1,Dial(SIP/s1a,20,r)
exten => 441234123456,102,Dial(SIP/s2a,20,r)
exten => 441234123456,203,Dial(SIP/s1b,20,r)
exten => 441234123456,304,Dial(SIP/s2a,20,r)

i.e. try the first, if busy try the next etc.

It seemed to consistently fail.

in [globals]
priorityjumping=no

was set, which came from the samples (i.e. make samples when installing
Asterisk).

I changed that to yes (i.e. priorityjumping=yes) and it started to work.

If that was the problem (which it seems to be), is that the wrong
default? Or am I missing something here completely?


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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[Asterisk-Users] chan_zap ast_pickup_call issue redux

[Andrew Kirch]

I'm running latest asterisk and zaptel, I have loaded wctdm
and lsmod shows that it is in the kernel.  I have configured the FXS and FXO
ports on my TDM400P, and ztcfg shows both as configured with no errors.  When I
start asterisk I get the following error: Mar 13 14:07:41 WARNING[10958]:
loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined
symbol: ast_pickup_cal.  A search of the web and this mailing list shows issues
related to the module not being loaded or zaptel not having been compiled
before asterisk.  I recompiled asterisk to ensure that it was linked against
zaptel and manually deleted the previously installed version of chan_zap.so
before doing make install.  After following this resolution the issue persists
with the same error as before.  Any help in getting zaptel working would be
greatly appreciated.






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Re: [Asterisk-Users] how to connect 3 or more servers via IAX ?

[Anthony Rodgers]

Hi Jean-Louis,

We have 3 servers connected togther - we do it by creating specific  
trunks between each one.


### iax.conf from asterix server:

; IAX Trunks

[dogmatix-in]
type=user
auth=md5
host=voip.dogmatix.dnv.org
secret=
context=international
trunk=yes

[dogmatix-out]
type=peer
auth=md5
host=voip.dogmatix.dnv.org
username=asterix-in
secret=
context=international
trunk=yes

[obelix-in]
type=user
auth=md5
host=voip.obelix.dnv.org
secret=
context=international
trunk=yes

[obelix-out]
type=peer
auth=md5
host=voip.obelix.dnv.org
username=asterix-in
secret=
context=international
trunk=yes

### iax.conf from dogmatix server

; IAX Trunks

[asterix-in]
type=user
auth=md5
host=voip.asterix.dnv.org
secret=
context=international
trunk=yes

[asterix-out]
type=peer
auth=md5
host=voip.asterix.dnv.org
username=dogmatix-in
secret=
context=international
trunk=yes

The iax.conf from the obelix server would be similar. Hope this gives  
the idea OK - let me know if you need any more information.


Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On 11-Mar-06, at 8:04 AM, Jean-Louis curty wrote:


Hi,

I successfully connected 2 servers via IAX but I'm pulling my hair  
to connect 2 extra servers , Anyone connected 3 or 4 servers  
together ? is it possible ?


I d like to share the dialplan so _2 goes to server A _3  
goes to serverB _4x goes to server C etc from the 4 servers


any example of which one is peer, which one is user or friend would  
help me  :-)


thanks
jl
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Re: [Asterisk-Users] Problem compiling zaptel on latest RHEL kernel(2.6.9-34.EL)

[Nilesh Londhe]

Thanks Russ. I updated the Makefile under /usr/src/zaptel and issued rebuild_zaptel. it worked flawlessly:)
On 3/13/06, Anthony Rodgers <[EMAIL PROTECTED]> wrote:
Many thanks, Russ - I'll give this a try.Thank goodness a) for test servers and b) for the ability of Linux to
rollback with a simple change to grub.conf :-)Regards,--Anthony RodgersBusiness Systems AnalystDistrict of North VancouverWeb: http://www.dnv.orgRSS Feed: 
http://www.dnv.org/rss.aspOn 11-Mar-06, at 7:33 AM, Russ Price wrote:> Anthony Rodgers wrote:> > Greetings,> >> > I have just updated our test server to 
2.6.9-34.EL and get the> following> > error messages when compiling zaptel:> >> > make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686'> >   CC [M]  /usr/src/zaptel/zaptel-
1.2.1/zaptel.o> > /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: error: syntax error> before> > "zone_lock">> [snipped]>> This bit me with CentOS 4.2 as well.  The problem is actually a
> typo in> the kernel spinlock.h file. See:>> >> and>> <
https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568>>> for more information.>> Here's a quick fix.  In your zaptel Makefile, add the following> (line 38> for 1.2.4) - THIS SHOLD BE ALL ONE LINE:
>> CFLAGS+=$(shell if uname -r | grep -q 2.6.9-34.EL; then echo> "-Drw_lock_t=\"rwlock_t\""; fi)>> This way, if this is fixed in the next kernel release, you won't
> need to> make another change to the Makefile.>> Russ> ___> --Bandwidth and Colocation provided by Easynews.com
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RE: [Asterisk-Users] Analog Desktop Phone

[asterisk]

On Mon, 13 Mar 2006, Kerry Garrison wrote:

system) you do NOT want to use a cheap phone on this system. At a minimum go
with a Linksys SPA941 or a Snom 360. You will have either one working in a


I would wait until snom fixes the issues with the 360 firmware.

-Dan
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Re: [Asterisk-Users] Professional Recordings

[Matt]

While I like Allison... sometimes she just sounds a little too breathy
for my liking.

On 3/12/06, Zach A <[EMAIL PROTECTED]> wrote:
>
>
>
> Allison Smith is the best. Her voice can be obtained at
> thevoice.digium.com. See her demos at www.theivrvoice.com and you'll be
> impressed. Plus all the voices in asterisk are from her, and I think all the
> voices in an ivr system should be of the same person. If you get anybody
> else recorded your prompts, what will you do with the voicemail, directory
> and some other system prompts? Or you'll need to change all the required
> sound files too to make all the voices consistent.
>
>
>
>
> Zach A
>
>
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>
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Re: [Asterisk-Users] Failed installing zaptel

[Chuck Bunn]

Hi,

I am having the same exact problem. I am assuming that it was a problem 
with a kernel update I did. I am in the process of rolling back to an 
older kernel... I will let you let know if this works. There is also a 
patch for zaptel but I believe this is for going from 1.3 to 1.4?


Thanks

Hall, Eric M. wrote:


Group
Having trouble installing zaptel. Below is my server specs

Intel Motherboard D101GGC
TE405P
CentOS-4.2-i386



Here is the output trying to do a 'make'
===

make clean
rm -f torisatool makefw tor2fw.h radfw.h
rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so *.lo
rm -f *.ko *.mod.c .*o.cmd
rm -rf .tmp_versions
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f fxotune
rm -f core
rm -f ztcfg-shared fxstest
[EMAIL PROTECTED] zaptel]# make
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c   -o makefw
./makefw tormenta2.rbt tor2fw > tor2fw.h
Loaded 69900 bytes from file
./makefw pciradio.rbt radfw > radfw.h
Loaded 42096 bytes from file
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o zonedata.lo
zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o tonezone.lo
tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o libtonezone.a -lm
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o torisatool.o torisatool.c
cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztmonitor.o ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o zttool.o zttool.c
cc -o zttool zttool.o -lnewt
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c   -o zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o fxotune.o fxotune.c
cc -o fxotune fxotune.o -lm
/lib/modules/2.6.9-34.ELsmp/build
make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel
modules
make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686'
 CC [M]  /usr/src/zaptel/zaptel.o
/usr/src/zaptel/zaptel.c:372: error: syntax error before "zone_lock"
/usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in
declaration of `zone_lock'
/usr/src/zaptel/zaptel.c:372: error: incompatible types in
initialization
/usr/src/zaptel/zaptel.c:372: error: initializer element is not constant
/usr/src/zaptel/zaptel.c:372: warning: data definition has no type or
storage class
/usr/src/zaptel/zaptel.c:373: error: syntax error before "chan_lock"
/usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in
declaration of `chan_lock'
/usr/src/zaptel/zaptel.c:373: error: incompatible types in
initialization
/usr/src/zaptel/zaptel.c:373: error: initializer element is not constant
/usr/src/zaptel/zaptel.c:373: warning: data definition has no type or
storage class
/usr/src/zaptel/zaptel.c: In function `free_tone_zone':
/usr/src/zaptel/zaptel.c:1022: warning: passing arg 1 of `_write_lock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c:1025: warning: passing arg 1 of `_write_unlock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone':
/usr/src/zaptel/zaptel.c:1035: warning: passing arg 1 of `_write_lock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c:1042: warning: passing arg 1 of `_write_unlock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c: In function `set_tone_zone':
/usr/src/zaptel/zaptel.c:1083: warning: passing arg 1 of `_read_lock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c:1095: warning: passing arg 1 of `_read_unlock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c: In function `zt_chan_reg':
/usr/src/zaptel/zaptel.c:1176: warning: passing arg 1 of
`_write_lock_irqsave' from incompatible pointer type
/usr/src/zaptel/zaptel.c:1199: warning: passing arg 1 of
`_write_unlock_irqrestore' from incompatible pointer type
/usr/src/zaptel/zaptel.c: In function `zt_chan_unreg':
/usr/src/zaptel/zaptel.c:1572: warning: passing arg 1 of
`_write_lock_irqsave' from incompatible pointer type
/usr/src/zaptel/zaptel.c:1608: warning: passing arg 1 of
`_write_unlock_

Re: [Asterisk-Users] Re: transfers/parked calls + polycom 501

[Mojo with Horan & Company, LLC]

When you hit the polycom's transfer button, a softkey appears on the 
screen that says "Blind" -- hitting this changes the transfer from 
attended to blind, and the blind button then disappears to show this. 
There's no real way I know to make this permanent.


Andrew Kohlsmith wrote:

On Monday 13 March 2006 10:20, Noah Miller wrote:
  The transer button on the polycom phone does not seem to transfer/park 
the call properly.  I have to use the # -> 700  to park  the call.

If I recall, using the Polycom transfer, you have to make sure it is done
as a blind transfer.  The Polycom attended transfer (default) option does
not work.


How is this configured?  That is, how do I configure the Polycom's transfer 
button to be a blind transfer?


-A.
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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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[Asterisk-Users] Re: Asterisk large scale, help needed

[Noah Miller]

Hi - 

>>  I was able to install Asterisk and configure many of it's features.
>> Currently I am using Extensions.conf for giving all my contexts and
>> extensions. Whenever I change my extensions or add a new context I have to
>> reload extensions.conf and practically it is not possible reloading many
>> times as we update or add contexts many times. Please tell me what could be
> 
> Why is it not possible to reload? whats wrong with reloading many times?

I think maybe there's some confusion here with the OP.  Reloading does not
interrupt calls in progress.  They will keep on going through as many
reloads as you want.


>> the best solution to avoid all this and if possible extensions.conf itself.
>> I came to know about scripts using AGI but I am a newbie totally and I do
>> not have any idea using them. I have seen a article in voip-info site
>> showing some examples on AGI and PHP. I want to do something like this:

Actually, if you're talking about a large scale deployment, AGI scripts
could conceivably be very bad.  Depending on how they are implemented, they
may add considerable processing overhead, which would be compounded on a
heavily taxed server.

Realtime is probably your best bet.


- Noah

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Re: [Asterisk-Users] Hardware timing source for MeetMe

[Andrew D Kirch]

Mike Clark wrote:
Will the low cost X100P clones available on ebay provide a solid 
hardware timing source? Our experience shows that while using ztdummy 
with no zaptel hardware does allow MeetMe to function, we experience 
unacceptable levels of delay after four ot five users join the 
conference. With both TDM400 and Sangoma A101 hardware, we have had 
20+ users with no problems.


We have a pure VoIP system installed, that has nor PRI or analog 
lines, but does have a need for MeetMe. If a $15 card will do the 
trick, we would obviously rather do that than spend a couple hundred 
bucks for the same thing. This card would not be used for voice, just 
timing.


Thanks,

Mike Clark
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Those do not have timing interfaces on them that I am aware of.
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[Asterisk-Users] Re: transfers/parked calls + polycom 501

[Noah Miller]

Hi Andrew - 

> On Monday 13 March 2006 10:20, Noah Miller wrote:
>>>   The transer button on the polycom phone does not seem to transfer/park
>>> the call properly.  I have to use the # -> 700  to park  the call.
>> 
>> If I recall, using the Polycom transfer, you have to make sure it is done
>> as a blind transfer.  The Polycom attended transfer (default) option does
>> not work.
> 
> How is this configured?  That is, how do I configure the Polycom's transfer
> button to be a blind transfer?

>From what I know, you can't configure the polycom transfer button to do
blind transfers by default.  You just have to make sure to manually press
the blind softkey every time you do a transfer for the parking lot.  My
solution was to set '#' as the asterisk transfer key, and remap the Polycom
transfer key to '#'.  Actually, my even more simplified solution was to hack
parking as a feature in features.conf.  I then set the '*' key to use the
parking feature, and remapped the services key to '*'.  I have a patch for
this, if you want it.

Now I need to do my part and test out the new metermaid feature in Olle's
test-this-branch ;-)


- Noah


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[Asterisk-Users] Zaptel not compiling on lastest Centos 4.2 kernel.

[Chuck Bunn]

Hi,

I made a big mistake on a Centos 4.2 box - I forgot to exclude the 
kernel from updating. Now zaptel will not do a "make linux26" see below. 
Is there a way to roll this back or is there a patch to get Zaptel to 
compile? I have a link to the modules using 'ln -s /lib/modules/uname 
-r/build linux-2.6" so that I did not have to specifiy the kernel 
version directly.



cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o fxotune.o fxotune.c

cc -o fxotune fxotune.o -lm
/lib/modules/2.6.9-34.ELsmp/build
make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel-1.2.4 
XPPMOD= modules

make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686'
 CC [M]  /usr/src/zaptel-1.2.4/zaptel.o
/usr/src/zaptel-1.2.4/zaptel.c:384: error: syntax error before "zone_lock"
/usr/src/zaptel-1.2.4/zaptel.c:384: warning: type defaults to `int' in 
declaration of `zone_lock'
/usr/src/zaptel-1.2.4/zaptel.c:384: error: incompatible types in 
initialization
/usr/src/zaptel-1.2.4/zaptel.c:384: error: initializer element is not 
constant
/usr/src/zaptel-1.2.4/zaptel.c:384: warning: data definition has no type 
or storage class

/usr/src/zaptel-1.2.4/zaptel.c:385: error: syntax error before "chan_lock"
/usr/src/zaptel-1.2.4/zaptel.c:385: warning: type defaults to `int' in 
declaration of `chan_lock'
/usr/src/zaptel-1.2.4/zaptel.c:385: error: incompatible types in 
initialization
/usr/src/zaptel-1.2.4/zaptel.c:385: error: initializer element is not 
constant
/usr/src/zaptel-1.2.4/zaptel.c:385: warning: data definition has no type 
or storage class

/usr/src/zaptel-1.2.4/zaptel.c: In function `free_tone_zone':
/usr/src/zaptel-1.2.4/zaptel.c:1034: warning: passing arg 1 of 
`_write_lock' from incompatible pointer type
/usr/src/zaptel-1.2.4/zaptel.c:1037: warning: passing arg 1 of 
`_write_unlock' from incompatible pointer type

/usr/src/zaptel-1.2.4/zaptel.c: In function `zt_register_tone_zone':
/usr/src/zaptel-1.2.4/zaptel.c:1047: warning: passing arg 1 of 
`_write_lock' from incompatible pointer type
/usr/src/zaptel-1.2.4/zaptel.c:1054: warning: passing arg 1 of 
`_write_unlock' from incompatible pointer type

/usr/src/zaptel-1.2.4/zaptel.c: In function `set_tone_zone':
/usr/src/zaptel-1.2.4/zaptel.c:1095: warning: passing arg 1 of 
`_read_lock' from incompatible pointer type
/usr/src/zaptel-1.2.4/zaptel.c:1107: warning: passing arg 1 of 
`_read_unlock' from incompatible pointer type

/usr/src/zaptel-1.2.4/zaptel.c: In function `zt_chan_reg':
/usr/src/zaptel-1.2.4/zaptel.c:1188: warning: passing arg 1 of 
`_write_lock_irqsave' from incompatible pointer type
/usr/src/zaptel-1.2.4/zaptel.c:1211: warning: passing arg 1 of 
`_write_unlock_irqrestore' from incompatible pointer type

/usr/src/zaptel-1.2.4/zaptel.c: In function `zt_chan_unreg':
/usr/src/zaptel-1.2.4/zaptel.c:1584: warning: passing arg 1 of 
`_write_lock_irqsave' from incompatible pointer type
/usr/src/zaptel-1.2.4/zaptel.c:1620: warning: passing arg 1 of 
`_write_unlock_irqrestore' from incompatible pointer type

/usr/src/zaptel-1.2.4/zaptel.c: In function `zt_ctl_ioctl':
/usr/src/zaptel-1.2.4/zaptel.c:3343: warning: passing arg 1 of 
`_write_lock' from incompatible pointer type
/usr/src/zaptel-1.2.4/zaptel.c:3345: warning: passing arg 1 of 
`_write_unlock' from incompatible pointer type

/usr/src/zaptel-1.2.4/zaptel.c: In function `zt_init':
/usr/src/zaptel-1.2.4/zaptel.c:6553: error: incompatible types in assignment
/usr/src/zaptel-1.2.4/zaptel.c: At top level:
/usr/src/zaptel-1.2.4/zaptel.c:188: warning: 'fcstab' defined but not used
make[2]: *** [/usr/src/zaptel-1.2.4/zaptel.o] Error 1
make[1]: *** [_module_/usr/src/zaptel-1.2.4] Error 2
make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-smp-i686'
make: *** [linux26] Error 2
[EMAIL PROTECTED] zaptel-1.2.4]#

Thanks
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RE: [Asterisk-Users] Analog Desktop Phone

[Kerry Garrison]

You really aren't going to find an analog phone that works as well as a SIP
phone for what you are trying to do. Some people suggested the GXP2000 for
$85 which works ok in a home environment. It is not a top quality phone but
it has all the features you want plus works very nicely with Asterisk. 

This same conversation is constantly going on on numerous forums. If you
think about what you are trying to accomplish, it might put things into
perspective. You are taking a state-of-the-art phone system flush with every
business feature you may ever want and trying to install it into your home
and you want to use a cheap phone on it. Things are just not designed that
way. If you want to be happy with your system, not to mention putting some
value on your time (and heaven help you if you have a wife that will use the
system) you do NOT want to use a cheap phone on this system. At a minimum go
with a Linksys SPA941 or a Snom 360. You will have either one working in a
matter of minutes. If you don't put any value on your time, then keep
monkeying around with a lesser solution, but the few hours you will save
just dropping in a decent phone should more than make up for the extra cost.

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Thczv F. Thczv
> Sent: Monday, March 13, 2006 8:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Analog Desktop Phone
> 
> On 3/12/06, Martin Joseph <[EMAIL PROTECTED]> wrote:
> 
> > > But what the OP wanted was a sulotion that together with 
> the SAP3000 
> > > makes for something that works even when there is a 
> blackout, since 
> > > the SPA3000 allows for failover to the FXS port from the FXO port 
> > > if/when there is no power to the unit. Which makes it a very good 
> > > solution when needed because of 911 reasons or the like.
> 
> > Actually it seems to me the Sipura 3000 is overkill in that case.
> > There are many other ATA's that are less expensive that 
> also have a 1 
> > port FXS, and a PSTN failover for blackout. It seems the OP doesn't 
> > need the FXO at all?
> >
> > The PA168V based ATA I have does this and was a little more 
> then half 
> > the cost of a SPA3000.  Works well too.
> 
> Shame on me, but I already have the SPA3000.  I like it very 
> much and it works fine.  Perhaps if I need another, I will 
> look at different products.
> 
> This is for my own home, where I am keeping my POTS line, partly as a
> 911 solution.  I have found a lot of analog desktop phones 
> that have some of the features I want, but not all of them.  
> The Cortelco 2200 looks like it might fit the bill.  But it 
> costs about $80.  I'm not sure I want to pay that much for an 
> analog phone that isn't wireless. 
> Other than that, the closest I have found so far is the AT&T 959:
> 
> http://www.amazon.com/gp/product/B00067KETY/ref=wl_it_dp/104-2
> 261851-2083919?%5Fencoding=UTF8&colid=1SGHZOJ18P2FB&coliid=I23
IRSR1SF2HPG&v=glance&n=172282
> 
> The problem with that one is that (as near as I can tell from 
> the photos and the manual) it has no visual MWI.
> 
> Still looking,
> 
> Dave
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> 


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[Asterisk-Users] Regexten & Regcontext

[JR Richardson]

Hi All,

I've been trying to get regexten and regcontext going for some sip peers but 
following the examples on the wiki is not working, as far as I can tell, 
nothing is happening.  the phone registers, sip show peers is ok, but the NoOp 
priority 1 extension never gets created or added to the dialplan.  Has anyone 
got this working?

Thanks.

JR

JR Richardson
Engineering for the Masses

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[Asterisk-Users] Considering Asterisk

[Thomas Johnson]

We currently operate an MKC Communications Server for our small company. We have 4 offices across 
Canada and calls to our toll-free number are answered by our VOIP server and directed by the auto 
attendant in the server office to the 3 satellite offices. This system works well except we have 
intermittent quality of the calls, sometimes losing connection altogether, but usually garbled 
speech etc. The 3 satellite offices are behind firewalls on adsl or cable high speed connections. We 
cannot get much support from MKC and I wonder if Asterisk would be a better system for this. Is this 
a problem because of the wide geographic area being covered, and so more router hops?


--
Thomas Johnson
Pacwill Environmental
527 Beaverbrook Court, Suite 420
Fredericton NB, CANADA, E3B 1X6
Tel. 506-462-0014
Fax: 506-462-0015
Email: [EMAIL PROTECTED]
Internet: http://www.pacwill.ca
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Re: [Asterisk-Users] Diff between X100M and X100P?

[John Daragon]

Phil Freed wrote:
> I have noticed a lot of folks mentioning the x100P, and very few
> mentioning x100M (which is what I have).  Are there important
> differences between them?

The X100P was a PCI card with a single FXO port (actually a WinModem,
more or less).

The X100M is a daugterboard for the TDM400P card.

jd
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[Asterisk-Users] Failed installing zaptel

[Hall, Eric M.]

Group
 Having trouble installing zaptel. Below is my server specs

Intel Motherboard D101GGC
TE405P
CentOS-4.2-i386



Here is the output trying to do a 'make'
===

make clean
rm -f torisatool makefw tor2fw.h radfw.h
rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so *.lo
rm -f *.ko *.mod.c .*o.cmd
rm -rf .tmp_versions
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f fxotune
rm -f core
rm -f ztcfg-shared fxstest
[EMAIL PROTECTED] zaptel]# make
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c   -o makefw
./makefw tormenta2.rbt tor2fw > tor2fw.h
Loaded 69900 bytes from file
./makefw pciradio.rbt radfw > radfw.h
Loaded 42096 bytes from file
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o zonedata.lo
zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o tonezone.lo
tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o libtonezone.a -lm
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o torisatool.o torisatool.c
cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztmonitor.o ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o zttool.o zttool.c
cc -o zttool zttool.o -lnewt
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c   -o zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o fxotune.o fxotune.c
cc -o fxotune fxotune.o -lm
/lib/modules/2.6.9-34.ELsmp/build
make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel
modules
make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686'
  CC [M]  /usr/src/zaptel/zaptel.o
/usr/src/zaptel/zaptel.c:372: error: syntax error before "zone_lock"
/usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in
declaration of `zone_lock'
/usr/src/zaptel/zaptel.c:372: error: incompatible types in
initialization
/usr/src/zaptel/zaptel.c:372: error: initializer element is not constant
/usr/src/zaptel/zaptel.c:372: warning: data definition has no type or
storage class
/usr/src/zaptel/zaptel.c:373: error: syntax error before "chan_lock"
/usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in
declaration of `chan_lock'
/usr/src/zaptel/zaptel.c:373: error: incompatible types in
initialization
/usr/src/zaptel/zaptel.c:373: error: initializer element is not constant
/usr/src/zaptel/zaptel.c:373: warning: data definition has no type or
storage class
/usr/src/zaptel/zaptel.c: In function `free_tone_zone':
/usr/src/zaptel/zaptel.c:1022: warning: passing arg 1 of `_write_lock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c:1025: warning: passing arg 1 of `_write_unlock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone':
/usr/src/zaptel/zaptel.c:1035: warning: passing arg 1 of `_write_lock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c:1042: warning: passing arg 1 of `_write_unlock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c: In function `set_tone_zone':
/usr/src/zaptel/zaptel.c:1083: warning: passing arg 1 of `_read_lock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c:1095: warning: passing arg 1 of `_read_unlock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c: In function `zt_chan_reg':
/usr/src/zaptel/zaptel.c:1176: warning: passing arg 1 of
`_write_lock_irqsave' from incompatible pointer type
/usr/src/zaptel/zaptel.c:1199: warning: passing arg 1 of
`_write_unlock_irqrestore' from incompatible pointer type
/usr/src/zaptel/zaptel.c: In function `zt_chan_unreg':
/usr/src/zaptel/zaptel.c:1572: warning: passing arg 1 of
`_write_lock_irqsave' from incompatible pointer type
/usr/src/zaptel/zaptel.c:1608: warning: passing arg 1 of
`_write_unlock_irqrestore' from incompatible pointer type
/usr/src/zaptel/zaptel.c: In function `zt_ctl_ioctl':
/usr/src/zaptel/zaptel.c:3331: warning: passing arg 1 of `_write_lock'
from incompatible pointer type
/usr/src/zaptel/zaptel.c:: warning: passing arg 1 of `_write_unlock'
from incompatible pointer type
/usr/src/zaptel/