[asterisk-users] Testing 911?

[voiplist]

It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
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[asterisk-users] Re: 7970 SIP configs

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 Hi All
 
 Has anyone got an annotated SEPmac.cnf.xml they are using successfully
 with the 7970 (8.0.3 Sip) and Asterisk?
 
 The SEPmac.cnf.xml files on the wiki are not annotated and although I've
 managed to upgrade the phone firmware and get a partial registration better
 info could speed it up.
 
 Is there a separate 7970 SIP forum/list anywhere?
 
 Thanks in advance.

From this page
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP
Use Another SEPmac.xml.cnf example - it works for sure (I have edited this 
part and I'm using it on my 2 7970 phone).

P.S.
If you find out how to make work something from Still need to configure: 
section, please send me e-mail

--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Re: Asterisk Database

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 Rushowr wrote:
  I'm not personally sure, but if I recall correctly, the astDB is cleared 
  whenever the Asterisk server is stopped...

 This is not correct.

Hi Doug,

Where can I find information's about maximum data that I can store in internal 
* database?


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Re: Clearing variables in the dialplan?

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 Hello all,
 
 Wanted to toss out a question that I've been looking into for some time now
 with no real results. When a variable is given a value in the dialplan, that
 obviously will take up a little memory. If you're running a rather
 large/complex dialplan, you may end up with variables you don't need after a
 while. What do you think is the best way to clear these out and free up
 memory? Currently I use:
 
   Set(varname=)
 
 To tell asterisk to set it to a null value. Any thoughts?

If those are channel variable, they should be cleared when you hang up.



--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
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[asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 If it was a .tar.gz download then you will need to reinstall.  

Hi Matt!

If I upgrade to 1.2.10 and than decide to go back to some prior version, how 
will I do that (using tar.gz)?


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] Testing 911?

[Brian Swan]

I don't think it's a stupid question at all.  Testing 911 routing is  
very important, and it would suck to find out it didn't work when you  
needed it to.  When I tested 911 at my wife's small business (we're  
on ZAP channels), I first called the non-emergency number for our  
local police department.  Depending on the size of the city your in,  
they may tell you to call a different policy department where the 911  
center for your area is located.  I then called their non-emergency  
number and explained to them that we were installing a new phone  
system and needed to test 911 functionality.  They said No problem,  
let me transfer you to the radio room, I assume when they transfered  
me I was then talking to one of the 911 supervisors or something.  I  
explained to them that I needed to make two test calls (one to 9,911  
and one to 911 as I have our system setup) in order to test 911  
functionality, and informed them that I would be calling back  
immediately after I hung up with them.   They said Sure, no problem.


When you do the actual do the deed, identify who you are (full name),  
where you are calling from (business name, etc), and that this is a  
test call on a new phone system.  They will usually read back to you  
the address they have on file for your phone number, and possibly  
some other information.  If you are using a T1, PRI they will also  
verify some E-911 information you are sending (ANI?  Help me out here  
someone...)  Also, I think it's important that you close by telling  
them that you're done testing, or that you have one (or two, or X)  
more test calls to make.


I tried to test out as much as I could in advance, so that I was  
fairly certain I wouldn't have to call them more then twice -- even  
though they know it's a test call, they may still be a little short  
with you on call #2 since I'm sure they have plenty of real  
emergencies to deal with. :)  Along those same lines, use some  
judgement as to when you perform your testing.  For instance, testing  
during severe weather, or during a hurricane probably wouldn't be a  
good idea.  Along those same lines (and some what less obvious ;),  
you may NOT want to test on a Friday or Saturday night if it could be  
avoided.   I actually used 411 while I was doing the initial setting  
up and testing to make sure I got everything right, then when I was  
99.99% sure it would work, I switched the 4 to a 9 and tested it for  
real.


Hope that helps!
Swannie



On Jul 17, 2006, at 1:05 AM, voiplist wrote:


It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
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Re: [asterisk-users] Testing 911?

[Martin Joseph]

On Jul 16, 2006, at 11:05 PM, voiplist wrote:


It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
Actually, not stupid at all.  I know in one case I have configured 911 
to dial on a 7 digit number for the local police, and I spoke to the 
police to let them know this is my setup.


In other words,  when I dial 911 from my house (shoreline, Washington 
state USA) I don't want it to dial 911 from my office (FXO is in 
Seattle) as that would be calling the wrong police to the wrong 
address.  I requested from the Shoreline police that they make a record 
of the fact that calls to them from my seattle number are actually 
coming from my shoreline address.


Hopefully I never need to test this under fire...


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RE: [asterisk-users] Testing 911?

[Alexander Lopez]

I call and immediately identify this as a test call.

I state the following. My Nane, and the fact that I am the PBX tech,
(engineer confuses them). I ask them to confirm my address and call back
number I provide to them.  If all is OK I thank them and hang up. I do
not think it is a false call if you identify it as such and give the
information. 

I once was almost charged with a false 911 call, I had added a daemon to
call my pager with a server number followed by 911 when a particular
server went down. It was a typo and not only did the server number NOT
appear but it was dialing 911 instead of my pager. I get a call from the
building security that my office door was open and that there were
firemen and police inside. I rushed over, thinking the worst and while I
was there trying to figure out why they were there they get another call
from dispatch stating that the person was calling again. After asking
the dispatcher the phone number (ANI) that was calling, I disco'ed the
modem. Thank god I was using POTS at the time.  I acted stupid and told
them it must have been a virus or something. I until this day had kept
quiet, I hope the statue of limitations has passed

Alex


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of voiplist
Sent: Monday, July 17, 2006 2:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Testing 911?

It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
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Re: [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!

[Martin Joseph]

On Jul 16, 2006, at 11:12 PM, Tomislav Parčina wrote:

In article [EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
If it was a .tar.gz download then you will need to reinstall.  

Hi Matt!

If I upgrade to 1.2.10 and than decide to go back to some prior version, how will I do that (using tar.gz)?


I think if you keep the older source in a separate directory,  you can always cd back to it and do a make clean, make,  make install.

This is only what I have gleaned from the list,  so hopefully more knowledgeable list members will chime in.

This is also the reason I have avoided building from SVN, as I like the idea of being able to revert to an earlier working build if need be...


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RE: [asterisk-users] IVR DTMF

[Khaled Chehab]

Thanks for your help but where is should put this bash script ,can
you guide me please 



Regards 



...receiving digits from IVR through
dtmf and store it on a text file 
short idea:
1 IVR start
2 set(number=)
3 playback(press_digit_or_#_to_finish)
4 (pressed) set(number=${number}${digit_pressed})
5 playback(press_another_digit_or_#_to_finish)
6 if digit pressed goto(pressed[point 44])
7 if # pressed execute
System(put_string_with_pressed_didgits_into_text_file.sh ${digit_pressed}
${calleridnum})

sh script
#!/bin/bash
digits=$1
number=$2
echo $1  $2.txt



Dear 



I want to make a billing recharge through receiving digits from IVR
through dtmf and store it on a text file ,



How can todo that ?



Regards












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Re: [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!

[trixter aka Bret McDanel]

On Sun, 2006-07-16 at 23:57 -0700, Martin Joseph wrote:
 I think if you keep the older source in a separate directory,  you can
 always cd back to it and do a make clean, make, make install.
 
or if you are lazy, make takes multiple targets so you could do:
make clean all install
all on one like that way and if one target fails the others shouldnt
proceed :)  'install' should have a dependancy on 'all' so if you just
do make clean install it should work the same.

It will use the newer zaptel if you dont do that as well, so if zaptel
is the issue that causes you to want to go back then you will have to do
a make clean all install there as well.


 This is also the reason I have avoided building from SVN, as I like
 the idea of being able to revert to an earlier working build if need
 be...

you can have different SVN repositories on your local system as well,
and still do that, or use a release tag to get a specific version,
either way.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com the VoIP provider that pays you!


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[asterisk-users] IVR DTMF

[Khaled Chehab]

Dear 



I want to make a billing recharge through receiving digits from IVR
through dtmf and store it on a text file ,



How can todo
that ?



Regards








*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.

If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person.

Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects.
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Re: [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!

[Matt Riddell (NZ)]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Martin Joseph wrote:
 
 On Jul 16, 2006, at 11:12 PM, Tomislav Parčina wrote:
 
 In article [EMAIL PROTECTED], [EMAIL PROTECTED]
 says...
 If it was a .tar.gz download then you will need to reinstall.

 Hi Matt!

 If I upgrade to 1.2.10 and than decide to go back to some prior
 version, how will I do that (using tar.gz)?


 I think if you keep the older source in a separate directory,  you can
 always cd back to it and do a make clean, make,  make install.
 
 This is only what I have gleaned from the list,  so hopefully more
 knowledgeable list members will chime in.
 
 This is also the reason I have avoided building from SVN, as I like the
 idea of being able to revert to an earlier working build if need be...

Also don't forget to pay close attention to the messages at the end of
the make process when compiling and installing Asterisk.

It will sometimes tell you that there are modules inside
/var/lib/asterisk/modules which were not compiled for the version you
are compiling.  If these are not asterisk-addons modules you will likely
need to remove them.

- --
Cheers,

Matt Riddell
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Re: [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!

[Matt Riddell (NZ)]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Julian Varanini wrote:
 So I can just install it over 1.2.9?  This is what I did and everything seems 
 to be working fine.

Yes as long as it doesn't complain there are modules which were not
compiled for the running version i.e. app_math

- --
Cheers,

Matt Riddell
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Re: [asterisk-users] SRTP enabling

[Matt Riddell (NZ)]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Martin Joseph wrote:
 
 On Jul 16, 2006, at 9:45 PM, Abdul wrote:
 
 Hello,

 In some countries i found that they are blocking SIP port 5060
 so instead of this i change to another port 1221, and its work
 well. But in one country the are not blocking SIP but they are
 playing with RTP packets, if they filtered it is VoIP RTP they
 are doing something called party cannot hear or some time caller
 cannot hear but called party can hear well.


 So i cosider to use SRTP to make encryption. and i am using
 my asterisk in VPS so i have full control to manage the server.
 If you guys have better Idea to prevent such kind of issue, it
 will be good for us.

 Why not use IAX2?  Then you only have one port to worry about
 reconfiguring

Or alternatively run the whole thing over a OpenVPN UDP encrypted
network (really simple to set up):

http://openvpn.net/

- --
Cheers,

Matt Riddell
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Re: [asterisk-users] Injecting prerecorded audio into active call

[Matt Riddell (NZ)]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Nick wrote:
 Yeah a bit messy I guess. I had been hoping for a simple solution, but
 knew there most likely wasn't!
 
 The one idea I did have would be to use some kind of SIP api on the web
 backend. Then bring the backend extension into a conference, then from
 the web api you would have to select the call to play audio in.
 
 This idea would work well I think, as it would mean the system can be
 use regardless of the training call being active on the asterisk box, as
 long as their system supported conference calls.
 
 This is where I fall down though, I'm no developer! Anyone know of an
 api that would allow this?

If you don't mind the call centre staff member pressing some buttons to
request help in the middle of the call you could use a featuremap using
features.conf and the playback application.

- --
Cheers,

Matt Riddell
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Re: [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!

[trixter aka Bret McDanel]

On Mon, 2006-07-17 at 19:21 +1200, Matt Riddell (NZ) wrote:
 It will sometimes tell you that there are modules inside
 /var/lib/asterisk/modules which were not compiled for the version you
 are compiling.  If these are not asterisk-addons modules you will likely
 need to remove them.

or modules from others that arent allowed to contribute to
asterisk-addons or the tree itself for whatever reason, of which I have
a few of those that have been specifically rejected for inclusion even
though disclaimers are on file :/

politics at its finest.  At least they work and it appears that some of
them take less ram and cpu than default asterisk stuffs :)


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com the VoIP provider that pays you!


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Re: [asterisk-users] Sphinx and Asterisk Integration, How?

[Matt Riddell (NZ)]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Zeeshan Zakaria wrote:
 After several hours of searching the Internet, couldn't understand how
 can I
 integrate Asterisk with Sphinx voice recognition system. The sphinx
 software
 itself I've installed on my server.
 
 I need help from those who have successfully done it and can guide me
 how to
 do it.
 Thanks

:-)

Top link on Google:

http://www.voip-info.org/wiki-Sphinx

- --
Cheers,

Matt Riddell
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Re: [asterisk-users] priority problem

[Kai Ober]

I use include in an other way than you do.
i use different extensions, not the same in each includet context, maybe 
that makes more sense (to you)


[apps]
include = emergency
include = cfwd
include = mailbox


[emergency]
exten = 911,1,do stuff here

[cfwd]
exten = *31,1, enable cfwd
exten = *32,1, disable cfwd

[mailbox]
exten = *41,1, enable mailbox
exten = *42,1, disable mailbox



Thanks again.  But I want to ask what is the usage of include if it is
a continue-until-matched type of contruct.


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RE: [asterisk-users] Polycom IP301 and Queues

[Dean @ INKnBITs]

Do you have a soft button on the IP301? I use the 501 and it works fine, you
do have to use the special asterisk code for it to work correctly. It lets
me login, logout, make the agent available/unavailable.

You can read about it at http://bugs.digium.com/view.php?id=6119

I found you must also use the trunk version of zaptel and libpri, and make
sure you use auth on the phones in the config.

Hope thats what you looking for, if so, any problems just ask, its just
taken me 2 weeks to get it working great.

Regards,
Dean.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Julian
Varanini
Sent: 17 July 2006 00:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom IP301 and Queues


Is there any way to use the polycom phones to log into and out of queues?
So the polycom phone could show their current status in that queue?  logged
in / logged out for example.

Thanks

Julian





 Subject: RE: [asterisk-users] PRI dropouts
 From: [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Date: Sat, 15 Jul 2006 20:47:17 +1000


 Hmm - I have had 2 bad PRI installs out of about 20, and both times it
 was faulty wiring from the Telco.
 But getting them to fix it can be a real struggle!


 Paul Hales
 Technical Manager
 www.asteriskit.com.au


 On Sat, 2006-07-15 at 12:23 +1000, James Sturges wrote:
  Have had L O T S of trouble like this, the settings zap config files
  seem to have to e exact, please send email to [EMAIL PROTECTED] and
  I will send config files.
 
 
 
  Thanks
 
 
 
  James
 
 
 
 
  __
  From:[EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Kevin
  Withnall
  Sent: Saturday, 15 July 2006 11:05 AM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] PRI dropouts
 
 
 
 
  Recently we cut over to using asterisk (trixbox 1.1.1) for our
  production system.
 
 
 
  We are using a TE110P digium card (Primary rate) with a Telstra onramp
  10.
 
 
 
  Sometimes when people call, on their end it doesn’t seem to connect.
  On our end, we get caller id, it passes ok to the sip phone but then
  no-one is there.
 
 
 
  Anyone have any similar problems and worked out how to solve it ?
 
 
 
  Thanks.
 
 
 
 
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Re: [asterisk-users] Snom 300 headset with static noise

[Adrià Vidal]

Have a look at this document:
http://www.snom.com/wiki/index.php/FAQs#Q:_Why_is_there_a_humming_noise_when_using_the_headset.3FMichielThanks Michiel, that was the second thing i do, phone was connected to 
a well powered/connected switch.I could understand a chep headset would do that, but a 30 euro headset maybe is not goingto be the best... but should perform quite better.Maybe is a design fault , because the same headset connected into another voIP phone runs fine. Snom maybe have gone too cheap and too bad for his snom 300 ?
 -- Adrià Vidal[EMAIL PROTECTED] | [EMAIL PROTECTED]
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Re: [asterisk-users] Re: Clearing variables in the dialplan?

[[EMAIL PROTECTED]]

On Monday 17 July 2006 2:12 am, Tomislav Parčina wrote:

 If those are channel variable, they should be cleared when you hang up.

Thanks for the input, but I was thinking more in terms of clearing the 
variable during the call. I use temporary variables in my dialplans.

SKM

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[asterisk-users] Parked calls

[harrygaillac-sip]

Hello everybody,

I is possible to manage multiple call parked per line
.
I mean a caller (agent) have to park more than two
call . It is possible to retrieve caller one ,two
,three, ... with a aplliction which one display the
calling parked to a PC screen or a screen phone .

Regards

Harry 










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[asterisk-users] DTMF in QUEUES dont work

[Rizwan Hisham]

Hi,
when im using only peer to peer call without any queues, im able to
dial any extension or send any digit thru dtmf durng a call. but
whenever i use queues then no phone dials any extension during a call
or a conference. i cant even hangup a call using * key. Any ideas how
this problem can be solved. im using H323 and SIP channels and i have
set both channels to use dtmf=rfc2033.-- RegardsRizwan HishamSoftware Engineer
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[asterisk-users] Re: call forwarding

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 Hi people. I want to know about call forwarding. I dial *72, and a message 
 say me to dial the extension , I did, then the message said is forward is 
 UNCONDITIONLA . But when I call , it doesn't work the forwarding.
 Who can help me please.

Without your dialplan - nobody!


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] priority problem

[unplug]

According to your setting, below is meaningless.  Am I right?

[apps]
include = emergency
include = cfwd
include = mailbox

[emergency]
exten = 911,1,do stuff here

[cfwd]
exten = *31,1, enable cfwd
exten = *32,1, disable cfwd
exten = 911,1, do stuff2 here
exten = 911,1, do stuff3 here

[mailbox]
exten = *41,1, enable mailbox
exten = *42,1, disable mailbox


On 7/17/06, Kai Ober [EMAIL PROTECTED] wrote:

I use include in an other way than you do.
i use different extensions, not the same in each includet context, maybe
that makes more sense (to you)

[apps]
include = emergency
include = cfwd
include = mailbox


[emergency]
exten = 911,1,do stuff here

[cfwd]
exten = *31,1, enable cfwd
exten = *32,1, disable cfwd

[mailbox]
exten = *41,1, enable mailbox
exten = *42,1, disable mailbox


 Thanks again.  But I want to ask what is the usage of include if it is
 a continue-until-matched type of contruct.

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RE: [asterisk-users] Snom 300 headset with static noise

[Christian Stredicke]

Title: FW: [asterisk-users] Snom 300 headset with static noise



There is a difference in the biasing circuit for the microphones in the 
headsets. Unfortunately there is no standard on the market. The snom phones 
190/320/360 (let’s say: type A) behave different than snom 300 (type B). So 
there is always the need to have different headsets or different cables (Quick 
Disconnect). Some headsets are just working with one type (those with extra 
amplifier) and other devices seem to work in both environments, but that’s not 
really true. The headsets are always working much better with just one type. So 
if someone has a headset designed for type A, he’ll have a bad quality while 
connecting it to type B phones although he is able to here something. 


Don’t forget to have a connection to an earth-signal (e. g. shielded 
Ethernet cable to PC/switch or earth-grounded power supply). 


Hope this helps, 
CS

  -Original Message-From: 
  [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] 
  On Behalf Of Koopmann, Jan-PeterSent: Sunday, July 16, 2006 1:31 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [asterisk-users] Snom 300 headset with static noiseOn Freitag, 14. 
  Juli 2006 10:13 Adrià Vidal wrote: Someone using these phone Snom 
  300 with his own headset ?We used to but the quality was horrifying. 
  Since we changed to Plantronics Noise Cancelling headsets everything is 
  wounderful. We got horrible static noise on them?Maybe the 
  article Michiel pointed out helps you still the overall voice quality of their 
  headsets (at least the ones they sold last year) is awful.Kind 
  regards, 
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Re: [asterisk-users] priority problem

[Steve Davies]

On 7/17/06, unplug [EMAIL PROTECTED] wrote:

According to your setting, below is meaningless.  Am I right?

[apps]
include = emergency
include = cfwd
include = mailbox

[emergency]
exten = 911,1,do stuff here

[cfwd]
exten = *31,1, enable cfwd
exten = *32,1, disable cfwd
exten = 911,1, do stuff2 here
exten = 911,1, do stuff3 here

[mailbox]
exten = *41,1, enable mailbox
exten = *42,1, disable mailbox



Why would you want (or need) to do this? As you seem to realise
already, the do stuff2/3 here lines will do nothing. If you were
writing a shell-script, and put a comment at the start of a line it
would also do nothing - Should we change that too?

You seem to be un-necessarily trying to rewrite the structure provided
in the extensions.conf file. Is there something _functional_ that you
cannot do that you need to be able to do? Or is it just that you
cannot lay out the file in the specific order that you want to?

Perhaps if you gave a real-life example with details of what you are
trying to do, it might be easier to offer a solution that suits your
needs.

Regards,
Steve
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[asterisk-users] problems to call brazil from germany

[Sebastian Reitenbach]

Hi,

I have problems to call to brazil, frome here in germany. the asterisk is 
connected to the telephone system via a pri interface. I use a preselected 
provider here to call out. 

when I try to call a number in brazil, a mobile phone here in the germany in 
the afternoon, when it is moring in brazil, then the chances to reach that 
number are next to zero. taking a mobile phone and call that number works 
fine.

when I try to call someone in brazil, taking numbers found by google, then i 
can reach a lot of these numbers.

anybody has an explanation for this? 
could it be that both carriers have different ways to route the call to brazil 
and the preselection provider has not so many lines for overseas?



kind regards
Sebastian


-- 
Sebastian ReitenbachTel.: ++49-(0)3381-8904-451
RapidEye AG Fax: ++49-(0)3381-8904-101
Molkenmarkt 30  e-mail:[EMAIL PROTECTED] 
D-14776 Brandenburg web:http://www.rapideye.de 

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Re: [asterisk-users] IVR DTMF

[Filip Drągowski]

in asterisk.conf there is 
"astagidir = /var/lib/asterisk/agi-bin"
it can be used for storing any scripts/programs fo *, it is suggested
for storiong AGI scripts there
example: /var/lib/asterisk/agi-bin/dtmf2text.file.sh


  
  
  

  
  
  Thanks for
your help but where is should put this bash script ,can
you guide me please 
  
  Regards 
   
  "...receiving digits from IVR
through
dtmf and store it on a text file "
short idea:
 1 IVR start
 2 set(number=)
 3 playback(press_digit_or_#_to_finish)
 4 (pressed) set(number=${number}${digit_pressed})
 5 playback(press_another_digit_or_#_to_finish)
 6 if digit pressed goto(pressed[point 44])
 7 if # pressed execute
System(put_string_with_pressed_didgits_into_text_file.sh
${digit_pressed}
${calleridnum})
  
sh script
#!/bin/bash
digits=$1
number=$2
echo "$1"  $2.txt
  
  
  Dear 
   
   I want to make a billing recharge through
receiving digits from IVR
through dtmf and store it on a text file ,
   
  How can todo that ?
   
  Regards
   
   
   
  




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Re: [asterisk-users] Re: Asterisk Database

[Doug Lytle]

Tomislav Parčina wrote:

Hi Doug,

Where can I find information's about maximum data that I can store in internal 
* database?

  


According to the Wiki:

The Asterisk database uses version 1 of the Berkley DB

So, you'd need to look up the information on the Berkeley website, to 
find it's limitations.


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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Re: [asterisk-users] DUNDI / regcontext

[Simon Woodhead]

Thanks for the reply Brad.The relevant section of sip.conf was posted:[general]regcontext=sipregistrationIf you mean extensions.conf, I wasn't creating the extension in there other than for testing. RegContext correctly creates the context on registration but does not create the extension. If I create the extension manually, the DUNDi lookup works just fine.
SimonOn 7/16/06, Watkins, Bradley [EMAIL PROTECTED] wrote:
Could you possibly put up the relevant section(s) of your sip.conf?It sounds like the DUNDi portion is set up properly, and obviously it's not going to find an extension that doesn't exist.Regards,- Brad
From: [EMAIL PROTECTED] on behalf of Simon WoodheadSent: Sat 7/15/2006 5:59 PMTo: 
asterisk-users@lists.digium.comSubject: [asterisk-users] DUNDI / regcontextHi folks,I've been having a go at getting DUNDI working this evening to enableusers to register to any Asterisk box and to look them up from another.
The DUNDI part works just great (very impressed), as does the subsequentjoining of calls between the two servers but I'm struggling withregcontext and would be grateful for any input.sip.conf includes:
[general]regcontext=sipregistrationWhen a user registers, I get the Added extension 'XX' priority 1 tosipregistration message. However, 'show dialplan' does not show theextension and a DUNDI lookup does not return it. The sipregistration
context has been auto-created but is empty. If I manually create thesipregistration context and add the NoOp extension, then everythingworks as expected.I've tried this across multiple boxes, each running different versions
right up to the latest stable but the behaviour is the same. It is alsothe same with both SIP and IAX registrations and doesn't make adifference if the peer is defined in the .conf file or Realtime. They do
all have identical configurations though so I suspect there might besomething in our setup which is conflicting.Any input gratefully received.All the best,SimonThe contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.
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RE: [asterisk-users] PRI dropouts - solution I hope...

[Kevin Withnall]

My files were almost exactly the same. We
only have 10 channels and the clid signaling was different.



We are however still getting the same
problems. I moved the box closer to the optomux (now we have 2m cable from the
optomux to the asterisk box.)



Any other ideas? We still are having the
same problems and also, some dropouts in the middle of calls.



Could the card be faulty ? I purchased it
from ebay second hand.



PS. What does the Transfer=yes
do ?



Thanks.













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Sturges
Sent: Saturday, 15 July 2006 10:23
PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [asterisk-users] PRI
dropouts - solution I hope...





Hi, had a few ask for this so thought may
be of interest to the list.



This is actually for the following setup:



Telstra ISDN30 - Asterisk
- BP250 PABX



The ISDN10, 20, 30s are all the
same physical link, but you may need to change the bchan and dchan settings for
ISDN 10 or 20.



We have had lot of issues over 12 months,
including physical cable issues, etc. But this config has passed Telstra
test equipment both on site and in the exchange. The calls dropping out
(for us) are timing issues do to telling Asterisk to gets it synch from the
Telstra line and providing synch to the PABX.



Anyway, Here it is, does not look like
much but have had experts working on it for a while.



The system handles 1800  2000 calls
per day.



Thanks



James



ZAPATA.conf

[channels]

context=default

musiconhold=default

switchtype=euroisdn

usecallerid=yes

cidsignalling=v23

cidstart=polarity

hidecallerid=no

callwaiting=no

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

cancallforward=yes

callreturn=yes

echocancel=yes

echocancelwhenbridged=yes

echotraining=800

rxgain=0.0

txgain=0.0



group=1

context=te405p-intelstra

pridialplan=local

signalling=pri_cpe

callerid=asreceived

channel=1-15, 17-31



group=4

context=te405p-frombp250

pridialplan=local

signalling=pri_net

overlapdial=yes

callerid=asreceived

channel=94-108, 110-124



ZAPTEL.conf

#

# Zaptel Configuration File

#

# This file is parsed by the Zaptel
Configurator, ztcfg



span=1,1,0,ccs,hdb3,crc4

bchan=1-15

dchan=16

bchan=17-31

span=2,0,0,ccs,hdb3,crc4

bchan=32-46

dchan=47

bchan=48-62

span=3,0,0,ccs,hdb3,crc4

bchan=63-77

dchan=78

bchan=79-93

span=4,0,0,ccs,hdb3,crc4

bchan=94-108

dchan=109

bchan=110-124 



loadzone=au

defaultzone=au 











From: James Sturges
[mailto:[EMAIL PROTECTED] 
Sent: Saturday, 15 July 2006 12:24
PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [asterisk-users] PRI
dropouts





Have had L O T S of trouble like this, the
settings zap config files seem to have to e exact, please send email to [EMAIL PROTECTED] and I will send
config files.



Thanks



James











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Withnall
Sent: Saturday, 15 July 2006 11:05
AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] PRI
dropouts





Recently we cut over to using asterisk (trixbox 1.1.1) for
our production system.



We are using aTE110P digium card (Primary rate) with a
Telstra onramp 10.



Sometimes when people call, on their end it doesnt
seem to connect. On our end, we get caller id, it passes ok to the sip phone
but then no-one is there.



Anyone have any similar problems and worked out how to solve
it ?



Thanks.










smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] priority problem

[unplug]

Actually, for the exten 911, it flows through do stuff, then do
stuff3 instead of do stuff2.  I want to implement it because I can
maintenance the dial plan easily.

Say,
My default context is [mycontext], and your default context is
[yrcontext].  We have some common contexts but not all.  So I can
simple remark some include statement of it is not necessary.  As I
expect, I can go thro' do stuff, 2  3 stuff and you can go thro' 2
 3 stuff. But I am wrong in the design.  Someone here suggest to use
macro to implement my design.  As I want to use ARA in my design.  If
I use macro to here, ARA will be meaningless.

[mycontext]
include = context1
include = context2
include = context3

[yrcontext]
include = context2
include = context3

[context1]
exten = 911,1,do stuff here

[context2]
exten = *31,1, enable cfwd
exten = *32,1, disable cfwd
exten = 911,1, do stuff2 here
exten = 911,2, do stuff3 here



On 7/17/06, Steve Davies [EMAIL PROTECTED] wrote:

On 7/17/06, unplug [EMAIL PROTECTED] wrote:
 According to your setting, below is meaningless.  Am I right?

 [apps]
 include = emergency
 include = cfwd
 include = mailbox

 [emergency]
 exten = 911,1,do stuff here

 [cfwd]
 exten = *31,1, enable cfwd
 exten = *32,1, disable cfwd
 exten = 911,1, do stuff2 here
 exten = 911,2, do stuff3 here  === my typo

 [mailbox]
 exten = *41,1, enable mailbox
 exten = *42,1, disable mailbox


Why would you want (or need) to do this? As you seem to realise
already, the do stuff2/3 here lines will do nothing. If you were
writing a shell-script, and put a comment at the start of a line it
would also do nothing - Should we change that too?

You seem to be un-necessarily trying to rewrite the structure provided
in the extensions.conf file. Is there something _functional_ that you
cannot do that you need to be able to do? Or is it just that you
cannot lay out the file in the specific order that you want to?

Perhaps if you gave a real-life example with details of what you are
trying to do, it might be easier to offer a solution that suits your
needs.

Regards,
Steve
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[asterisk-users] asterisk sending connects when it shouldn't

[Simone Cittadini]

When asterisk receives those messages you hear when calling an 
unreacheable cellular phone it sends a 'connect' over the terminating 
PRI line (digium TE410P), making the call seen as billed from customer's 
perspective.
I don't know if this behaviour is a bug or something I can resolve with 
some fine tuning, so I'm sending to both lists.
Since the calls comes from a SIP connected GSM gateway, is there some 
SIP code which corresponds to the 'pass audio but don't connect' we want 
here ?


that's roughly the extension :


exten = _X.,1,AGI(agi://127.0.0.1:54321/SomeAgiHere?someArgumentsHere)
exten = _X.,n,GotoIf($[${CALLABLE}=TRUE]?chkmax:hangup)
exten = _X.,n(chkmax),Set(GROUP()=${TECH_PRE})
exten = _X.,n,GotoIf($[${GROUP_COUNT(${TECH_PRE})} = 
${MAX_CALLS}]?hangup:dial)

exten = _X.,n(dial),Dial(${STR_DIAL})
exten = _X.,n(hangup),Hangup

exten = h,1,Set(CDR(userfield)=${USERFIELD}-${HANGUPCAUSE})



Here the provider's trace of a call answered by asterisk :

/HDLU 4/Port
  === LAPD ===
   --- ADDRESS ---
   SAPI   : 0 = call control procedures
   CR : ..1.
   EA0: ...0
   TEI: 0 = non-automatic TEI assignment user equipment
   EA1: ...1
   --- CONTROL ---
   --- I FRAME ---
   I FORMAT   : ...0
   N(S)   : 86
   P  : ...0
   N(R)   : 31
   === ETSI ISDN ===
PROT DISC  : 08h = Q.931 user-network call control message
LEN CALL R : 2
SPARE  : 0
FLAG   : 1... = the message is sent to the side that 
originates the call reference

CALL REF   : 226
MESS TYPE  : 07h = Connect


Here the complete trace :

/HDLU 4/Port
 0  TEI:  0  CALL REF:  226  Setup  '500'  '[called number]'
 0  TEI:  0  CALL REF:  226  Setup acknowledge
 0  TEI:  0  CALL REF:  226  Call proceeding
 0  TEI:  0  CALL REF:  226  Connect  == should not
 0  TEI:  0  CALL REF:  226  Connect acknowledge
 0  TEI:  0  CALL REF:  226  Disconnect   16 normal call clearing
 0  TEI:  0  CALL REF:  226  Release
 0  TEI:  0  CALL REF:  226  Release complete


-

Here a trace from a correctly functioning non-voip system :

/HDLU 4/Port
 0  TEI:  0  CALL REF:  246  Setup  '500'
 0  TEI:  0  CALL REF:  246  Setup acknowledge
 0  TEI:  0  CALL REF:  246  Information  'c'
 0  TEI:  0  CALL REF:  246  Information  'a'
 0  TEI:  0  CALL REF:  246  Information  'l'
 0  TEI:  0  CALL REF:  246  Information  'l'
 0  TEI:  0  CALL REF:  246  Information  'e'
 0  TEI:  0  CALL REF:  246  Information  'd'
 0  TEI:  0  CALL REF:  246  Information  'n'
 0  TEI:  0  CALL REF:  246  Information  'u'
 0  TEI:  0  CALL REF:  246  Information  'm'
 0  TEI:  0  CALL REF:  246  Information  'b'
 0  TEI:  0  CALL REF:  246  Call proceeding
 0  TEI:  0  CALL REF:  246  Progress
 0  TEI:  0  CALL REF:  246  Progress
 0  TEI:  0  CALL REF:  246  Disconnect   16 normal call clearing
 0  TEI:  0  CALL REF:  246  Release
 0  TEI:  0  CALL REF:  246  Release complete

--
Simone Cittadini
2K Elektronika
Tel +39.02.26265583
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[asterisk-users] question ast db

[unplug]

Hi,
 I want to know about the content of ast db.  It is like a registry
of the asterisk to store information about register users.  The
similar user register information will be stored in DB in ARA.  I want
to verify that when user sends a register request and it is valid,
asterisk will capture the user information in ast db and also in
rdbms.  Then when the user makes a call request, asterisk will use ast
db to form a request message.  When there is a call to sip user,
asterisk will use ast db to find the location of the user.  Am I
right?

/SIP/Registry/871963596932:
221.126.25.138:5060:60:871963596932:sip:[EMAIL PROTECTED]:5060
/SIP/Registry/871966742968:
210.184.23.31:15060:180:871966742968:sip:[EMAIL PROTECTED]:5060
/SIP/Registry/871966760539:
210.184.23.31:5060:60:871966760539:sip:[EMAIL PROTECTED]:5060
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Re: [asterisk-users] priority problem

[Filip Drągowski]

Did You try CLI show dialplan ?
if You set up 911 extension in 2 diffrent context and both context are 
included
in third .. only one 911 will be available. 911 first loaded to asterisk 
dialplan

will be valid and second will be discarded.
Loading dialplan (example below)
[mycontext] should load [context1] first, 911 aplly
then [context2] is loaded .. trying to apply second 911..
but ther is one alredy !
mayby that is a problem ?

Actually, for the exten 911, it flows through do stuff, then do
stuff3 instead of do stuff2.  I want to implement it because I can
maintenance the dial plan easily.

Say,
My default context is [mycontext], and your default context is
[yrcontext].  We have some common contexts but not all.  So I can
simple remark some include statement of it is not necessary.  As I
expect, I can go thro' do stuff, 2  3 stuff and you can go thro' 2
 3 stuff. But I am wrong in the design.  Someone here suggest to use
macro to implement my design.  As I want to use ARA in my design.  If
I use macro to here, ARA will be meaningless.

[mycontext]
include = context1
include = context2
include = context3

[yrcontext]
include = context2
include = context3

[context1]
exten = 911,1,do stuff here

[context2]
exten = *31,1, enable cfwd
exten = *32,1, disable cfwd
exten = 911,1, do stuff2 here
exten = 911,2, do stuff3 here




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RE: [asterisk-users] Testing 911?

[Watkins, Bradley]

This is the tact that I take, and it's never been a problem for us.

Regards,
- Brad 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Monday, July 17, 2006 2:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Testing 911?

I call and immediately identify this as a test call.

I state the following. My Nane, and the fact that I am the PBX tech,
(engineer confuses them). I ask them to confirm my address and call back
number I provide to them.  If all is OK I thank them and hang up. I do
not think it is a false call if you identify it as such and give the
information. 

I once was almost charged with a false 911 call, I had added a daemon to
call my pager with a server number followed by 911 when a particular
server went down. It was a typo and not only did the server number NOT
appear but it was dialing 911 instead of my pager. I get a call from the
building security that my office door was open and that there were
firemen and police inside. I rushed over, thinking the worst and while I
was there trying to figure out why they were there they get another call
from dispatch stating that the person was calling again. After asking
the dispatcher the phone number (ANI) that was calling, I disco'ed the
modem. Thank god I was using POTS at the time.  I acted stupid and told
them it must have been a virus or something. I until this day had kept
quiet, I hope the statue of limitations has passed

Alex


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of voiplist
Sent: Monday, July 17, 2006 2:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Testing 911?

It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
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The contents of this e-mail are intended for the named addressee only. It 
contains information that may be confidential. Unless you are the named 
addressee or an authorized designee, you may not copy or use it, or disclose it 
to anyone else. If you received it in error please notify us immediately and 
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[asterisk-users] Bristuffed Asterisk: Hangup problems

[Jeroen Zwarts]

Hello list,

I have just tested the new Bristuff-0.3.0-Pre-1r 
(released this morning) but it seems that the hangup bug isn't resolved yet. I 
installed Bristuff the normal way (just run install.sh) but Asterisk still 
doesn't hangup properly. Investigation of the sourcecode revealed that the 
peercallstate/ourcallstate in q931.c is still wrong. But, for some reason I 
accidentally got it to work (hanging up correctly AND running the hangup 
context). Here's how:

Install Bristuff the normal way 
(install.sh)
Go into the libpri directory in the installation 
dir. 
edit q931.c with the proposed patch we discussed 
here on the board. 
run make clean all in the libpri dir.
run make install in the libpri dir. 

For what I've tested this morning Asterisk now 
seems to hang up correctly AND runs the h-context (including DeadAgi scripts 
which we need).
Maybe there is an easier way yo do this, but this 
seems to work for me. I will investigate this further, but first i'm going to 
test Asterisk 1.2.9.1 to see if it fits my needs. 


Regards,

Jeroen Zwarts

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Re: [asterisk-users] priority problem

[Steve Davies]

Someone here suggest to use
macro to implement my design.  As I want to use ARA in my design.  If
I use macro to here, ARA will be meaningless.



Yes, I suggested macros. Sorry, what is ARA?

Steve
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Re: [asterisk-users] Testing 911?

[Rich Adamson]

voiplist wrote:

It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?


As others have posted, test calls are allowed but the 911 center would 
prefer they be completed during non-peak times. The only way to know 
what their non-peak periods are is to give them a call on the 
non-emergency number and ask.


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[asterisk-users] DTMF

[Rizwan Hisham]

Hi Guys, I need a little help in using DTMF settings. Im using SIP and
H323 channels, both are set to use dtmf=rfc2833. 2 days ago it was
working fine, it still works fine when im in conference, for example
when i use the following extension:
exten=1234,1,MeetMe(1234|X|)
by using this extension im able to jump to any extension i want by dialing that extension.
The problem occurs when i use the Dial() application:
exten=1,1,Dial(SIP/200,,tT)
when i press # nothing happens
i have no idea how to solve this problem-- RegardsRizwan HishamSoftware Engineer
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Re: [asterisk-users] Connect to 'agi://blablabla' failed: Operation now in progress

[Simone Cittadini]

Moises Silva ha scritto:


AFAIK operation now in progress is a common status when you open a
socket connection. When you use blocking sockets usually you dont see
this because the connect call does not return until the connection
is done. But when using non-blocking sockets, the connect call returns
immediatly and if you try to connect again, you will get the
operation now in progress message. I have seen this in my PHP
Manager Proxy, but not sure what implications may have in FastAGI. May
be it only tells that the connection stablishment takes a little
longer, network congestion may be?



We have a 'non blocking father' which spawns a 'blocking child' for each 
connection.
So this can be the case, but I don't think it's related to network 
congestion, it's local on 127.0.0.1 and I see the messages even on low load.


Oh well, since it works ...



Regards

On 7/13/06, Simone Cittadini [EMAIL PROTECTED] wrote:


I get a lot of this warnings in my logs.

Connect to 'agi://blablabla' failed: Operation now in progress

What exactly 'operation now in progress means' ? is asterisk still
trying so the call isn't lost ?

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[asterisk-users] news

[Kris Edwards]

-- Ita erat quando hic adveni 


news.rtf
Description: RTF file
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RE: [asterisk-users] PRI dropouts - solution I hope...

[Steven Totaro]

I have had the exact opposite results.  I have hooked Asterisk up with
passthrough on many different systems and always initially had setup
problems which were fixed with tweaking. 

Maybe Sangoma boards will give you less trouble?

Thanks,
Steve Totaro
 
 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Paul Hales
 Sent: Sunday, July 16, 2006 9:23 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion;
 [EMAIL PROTECTED]
 Subject: RE: [asterisk-users] PRI dropouts - solution I hope...
 
 
 In my experience PRI pass through setups have been false economy.
 
 They seem to save a few dollars, but you still have to spend the money
 to save it and they never run as well.
 
 Paul Hales
 
 --
 Paul Hales
 Technical Manager
 AsteriskIT
 www.asteriskit.com.au
 bus: 03 8320 8100
 mob: 0434 673 529
 
 
 On Sat, 2006-07-15 at 22:22 +1000, James Sturges wrote:
  Hi, had a few ask for this so thought may be of interest to the
list.
 
 
 
  This is actually for the following setup:
 
 
 
  Telstra ISDN30 - Asterisk - BP250 PABX
 
  
 
  The ISDN10, 20, 30's are all the same physical link, but you may
need
  to change the bchan and dchan settings for ISDN 10 or 20.
 
 
 
  We have had lot of issues over 12 months, including physical cable
  issues, etc.  But this config has passed Telstra test equipment both
  on site and in the exchange.  The calls dropping out (for us) are
  timing issues do to telling Asterisk to gets it synch from the
Telstra
  line and providing synch to the PABX.
 
 
 
  Anyway, Here it is, does not look like much but have had experts
  working on it for a while.
 
 
 
  The system handles 1800 - 2000 calls per day.
 
 
 
  Thanks
 
 
 
  James
 
 
 
  ZAPATA.conf
 
  [channels]
 
  context=default
 
  musiconhold=default
 
  switchtype=euroisdn
 
  usecallerid=yes
 
  cidsignalling=v23
 
  cidstart=polarity
 
  hidecallerid=no
 
  callwaiting=no
 
  usecallingpres=yes
 
  callwaitingcallerid=yes
 
  threewaycalling=yes
 
  transfer=yes
 
  cancallforward=yes
 
  callreturn=yes
 
  echocancel=yes
 
  echocancelwhenbridged=yes
 
  echotraining=800
 
  rxgain=0.0
 
  txgain=0.0
 
 
 
  group=1
 
  context=te405p-intelstra
 
  pridialplan=local
 
  signalling=pri_cpe
 
  callerid=asreceived
 
  channel=1-15, 17-31
 
 
 
  group=4
 
  context=te405p-frombp250
 
  pridialplan=local
 
  signalling=pri_net
 
  overlapdial=yes
 
  callerid=asreceived
 
  channel=94-108, 110-124
 
 
 
  ZAPTEL.conf
 
  #
 
  # Zaptel Configuration File
 
  #
 
  # This file is parsed by the Zaptel Configurator, ztcfg
 
 
 
  span=1,1,0,ccs,hdb3,crc4
 
  bchan=1-15
 
  dchan=16
 
  bchan=17-31
 
  span=2,0,0,ccs,hdb3,crc4
 
  bchan=32-46
 
  dchan=47
 
  bchan=48-62
 
  span=3,0,0,ccs,hdb3,crc4
 
  bchan=63-77
 
  dchan=78
 
  bchan=79-93
 
  span=4,0,0,ccs,hdb3,crc4
 
  bchan=94-108
 
  dchan=109
 
  bchan=110-124
 
 
 
  loadzone=au
 
  defaultzone=au
 
 
 
 
 
 
 
 
 
 
 
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Re: [asterisk-users] zaptel on dual processor, How?

[Derek Whitten]

Zeeshan Zakaria wrote:
 How to install kernel sources?
 
 On 7/17/06, *Dennis Gilmore* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 wrote:
 
 On Monday 17 July 2006 12:05 am, Zeeshan Zakaria wrote:
  I am trying to install zaptel on dual Xeon processor but it gives
 error,
  saying 'You do not appear to have the kernel sources for your current
  kernel installed.
  make: *** [linux26] Error 1'
 
  Googled for many hours, but nothing, except to use non smp kernel.
 How can
  I build zaptel for smp.
 Install your kernel sources  the process will vary depending on your
 distro
 
 --
 Dennis Gilmore, RHCE
 Proud Australian
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 
 -- 
 Zeeshan A Zakaria
 
 
 
 
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http://www.kernel.org/pub/linux/kernel/v2.6/linux-2.6.17.6.tar.gz

There's a README in the tarball that tells all.. :-)






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Re: [asterisk-users] Polycom config file location

[Jerry Jones]

If you at least setup your ftp server, and point the phones to it,  
they will save a copy of their contact database so that will not be  
lost.


Just edit and save an entry after server is ready and it will create  
the file.


No too hard to use the web browser and look at each phone to get its  
current settings and manually create a config file.



On Jul 16, 2006, at 5:04 PM, Avi Miller wrote:


Stephen Murphy wrote:
My question is: How do I get the current config files the phone is  
using off the phone?


AFAIK, you can't. :( You can only provide new configuration files  
from your FTP/TFTP server. However, the Polycoms do strange things  
when they've been configured in multiple locations. You might find  
the phone overwriting the configuration files with its original  
configuration.


That is not confirmed though. I've just seen my Polycoms do weird  
stuff in the wild. :)



--
National Manager - Special Projects

 Melbourne / Sydney / Canberra / Hobart / London /
  2/340 Gore StreetT: 1 300 SQUIZ (77859)
  Fitzroy, VIC T: 03 9235 5400
  3065 F: 03 9235 5444
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Re: [asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE

[Jerry Jones]

This will typically happen over internet connections. If the qualify  
message is lost, or takes too long the * server will stop sending  
calls. This is the normal function of qualify. I find that in most  
cases it is a matter of the end user saturating his connection on his  
end, assuming you are not overloading yours.



On Jul 16, 2006, at 10:13 PM, Tong wrote:

According to your console output it looks like there is some major  
latency.  What is the average ping time from your asterisk machine  
to the polycom phone?

- Original Message -
From: Rana Dutt
To: Asterisk Users
Sent: Sunday, July 16, 2006 6:51 PM
Subject: [asterisk-users] Polycom phone cycles between UNREACHABLE  
and REACHABLE


I have a customer with a Polycom 501 phone behind a NAT. His phone  
is connected to his Netgear router at home which in turn is  
connected to his cable modem. The phone is set up to register with  
our remote Asterisk server which is on a public, static IP address,  
with no NAT.


If we set qualify=yes, our Asterisk console shows his extension  
becoming UNREACHABLE for a minute, then REACHABLE for a minute,  
then UNREACHABLE again, in an endless cycle. If we try to call the  
phone while it is UNREACHABLE, the phone never rings and the call  
goes straight to voice mail. This is very annoying.


If we set qualify=no, then if we try to call the phone, the phone  
sometimes does not ring at all, and we hear silence. The call  
eventually goes to voice mail. This is equally annoying to the  
customer.


What is the solution to this problem? We have other customers with  
Polycom phones behind NAT, and they don't have this problem. Will  
we have better luck if we replace the Polycom with a Linksys 942  
phone?


Here is some console output:

Jul 16 21:44:24 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer:  
Peer '280' is now UNREACHABLE!  Last qualify: 174
Jul 16 21:45:33 NOTICE[19981]: chan_sip.c:9697  
handle_response_peerpoke: Peer '280' is now REACHABLE! (3181ms /  
5000ms)
Jul 16 21:47:37 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer:  
Peer '280' is now UNREACHABLE!  Last qualify: 175


Here is the way the phone is set up in sip.conf:

[280]
type=peer
username=280
secret=280
host=dynamic
dtmfmode=rfc2833
callerid=John 280
context=company_x
mailbox=280
nat=yes
canreinvite=no
qualify=5000

We are using Asterisk 1.2.5 with standard .conf files. We are not  
using realtime or databases. Any help would be highly appreciated.


Rana Dutt
Softel Solutions
[EMAIL PROTECTED]



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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.10.1/389 - Release Date:  
7/14/2006

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Re: [asterisk-users] zaptel on dual processor, How?

[Olivier Picquenot]

Zeeshan Zakaria a écrit :

How to install kernel sources?


As asked before :
What distro are you using ?
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[asterisk-users] asterisk sending connects when it shouldn't (is there a q931-INFORMATION equivalent in IAX2 ?)

[Simone Cittadini]

When asterisk receives those messages you hear when calling an 
unreacheable cellular phone it sends a 'connect' over the terminating 
PRI line (digium TE410P), making the call seen as billed from customer's 
perspective.
I don't know if this behaviour is a bug or something I can resolve with 
some fine tuning, so I'm sending to both lists.


this is the layout of machines :

|gsm gateway| - sip - |asterisk client| - iax2 - |asterisk server| 
- zap -  pri lines (nortel ?)



that's roughly the extension on the server :


exten = _X.,1,AGI(agi://127.0.0.1:54321/SomeAgiHere?someArgumentsHere)
exten = _X.,n,GotoIf($[${CALLABLE}=TRUE]?chkmax:hangup)
exten = _X.,n(chkmax),Set(GROUP()=${TECH_PRE})
exten = _X.,n,GotoIf($[${GROUP_COUNT(${TECH_PRE})} = 
${MAX_CALLS}]?hangup:dial)

exten = _X.,n(dial),Dial(${STR_DIAL})
exten = _X.,n(hangup),Hangup

exten = h,1,Set(CDR(userfield)=${USERFIELD}-${HANGUPCAUSE})


Here the provider's trace of a call answered by asterisk :

/HDLU 4/Port
 === LAPD ===
  --- ADDRESS ---
  SAPI   : 0 = call control procedures
  CR : ..1.
  EA0: ...0
  TEI: 0 = non-automatic TEI assignment user equipment
  EA1: ...1
  --- CONTROL ---
  --- I FRAME ---
  I FORMAT   : ...0
  N(S)   : 86
  P  : ...0
  N(R)   : 31
  === ETSI ISDN ===
   PROT DISC  : 08h = Q.931 user-network call control message
   LEN CALL R : 2
   SPARE  : 0
   FLAG   : 1... = the message is sent to the side that 
originates the call reference

   CALL REF   : 226
   MESS TYPE  : 07h = Connect


Here the complete trace :

/HDLU 4/Port
0  TEI:  0  CALL REF:  226  Setup  '500'  '[called number]'
0  TEI:  0  CALL REF:  226  Setup acknowledge
0  TEI:  0  CALL REF:  226  Call proceeding
0  TEI:  0  CALL REF:  226  Connect  == should not
0  TEI:  0  CALL REF:  226  Connect acknowledge
0  TEI:  0  CALL REF:  226  Disconnect   16 normal call clearing
0  TEI:  0  CALL REF:  226  Release
0  TEI:  0  CALL REF:  226  Release complete


- 



Here a trace from a correctly functioning non-voip system :

/HDLU 4/Port
0  TEI:  0  CALL REF:  246  Setup  '500'
0  TEI:  0  CALL REF:  246  Setup acknowledge
0  TEI:  0  CALL REF:  246  Information  'c'
0  TEI:  0  CALL REF:  246  Information  'a'
0  TEI:  0  CALL REF:  246  Information  'l'
0  TEI:  0  CALL REF:  246  Information  'l'
0  TEI:  0  CALL REF:  246  Information  'e'
0  TEI:  0  CALL REF:  246  Information  'd'
0  TEI:  0  CALL REF:  246  Information  'n'
0  TEI:  0  CALL REF:  246  Information  'u'
0  TEI:  0  CALL REF:  246  Information  'm'
0  TEI:  0  CALL REF:  246  Information  'b'
0  TEI:  0  CALL REF:  246  Call proceeding
0  TEI:  0  CALL REF:  246  Progress
0  TEI:  0  CALL REF:  246  Progress
0  TEI:  0  CALL REF:  246  Disconnect   16 normal call clearing
0  TEI:  0  CALL REF:  246  Release
0  TEI:  0  CALL REF:  246  Release complete


On the asterisk client it seems that SIP correctly opens only a leg of 
the call :


asterisk : 102 invite
- 100 Trying
- 200 OK
asterisk : ACK (now I hear the audio)
(I hangup)
asterisk : BYE
- 200 OK

Destroying call 'blabla'@ip

(with a normally answered call I see 183 Session progress instead of the 
first 200 while ringing, and the the destroyed calls are two)


the iax debug : (still talking about the call that shouldn't send the 
connect on isdn line)


   -- Accepting AUTHENTICATED call from IP:
   requested format = alaw,
   requested prefs = (),
   actual format = alaw,
   host prefs = (alaw),
   priority = mine
   -- Executing Dial(IAX2/IP:4569-2, SIP/[EMAIL PROTECTED]) in new stack
   -- Called [EMAIL PROTECTED]
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: 
ACCEPT

  Timestamp: 00014ms  SCall: 2  DCall: 00188 [IP:4569]
  FORMAT  : 8
astegateway4*CLI
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
  Timestamp: 00014ms  SCall: 00188  DCall: 2 [IP:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: VOICE   Subclass: 8
  Timestamp: 00090ms  SCall: 00188  DCall: 2 [IP:4569]
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK
  Timestamp: 00090ms  SCall: 2  DCall: 00188 [IP:4569]
   -- SIP/gateway4-20e0 answered IAX2/82.113.204.70:4569-2
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: 
ANSWER

  Timestamp: 04698ms  SCall: 2  DCall: 00188 [IP:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK
  Timestamp: 04698ms  SCall: 00188  DCall: 2 [IP:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 003 Type: VOICE   Subclass: 8
  Timestamp: 04764ms  SCall: 2  DCall: 00188 [IP:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: 

Re: [asterisk-users] Polycom config file location

[Alex Robar]

Our 501's upload their configs to the server by themselves... Is this uncommon? Seems to me that if you had no config on the server at all but pointed the phones there anyways, they should upload their current set of files there and then default to using that set of configs until the server is updated.
AlexOn 7/17/06, Jerry Jones [EMAIL PROTECTED] wrote:
If you at least setup your ftp server, and point the phones to it,they will save a copy of their contact database so that will not belost.Just edit and save an entry after server is ready and it will create
the file.No too hard to use the web browser and look at each phone to get itscurrent settings and manually create a config file.On Jul 16, 2006, at 5:04 PM, Avi Miller wrote: Stephen Murphy wrote:
 My question is: How do I get the current config files the phone is using off the phone? AFAIK, you can't. :( You can only provide new configuration files from your FTP/TFTP server. However, the Polycoms do strange things
 when they've been configured in multiple locations. You might find the phone overwriting the configuration files with its original configuration. That is not confirmed though. I've just seen my Polycoms do weird
 stuff in the wild. :) -- National Manager - Special Projects  Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: 1 300 SQUIZ (77859)
 Fitzroy, VIC T: 03 9235 5400 3065 F: 03 9235 5444W: http://www.squiz.net/ . Open Source- Own it-
Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list
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RE: [asterisk-users] PRI dropouts - solution I hope...

[James Sturges]

Is your dial plan very simple, ie bypass FREEPBX
etc, to make sure no problems.



There are also debug command in the CLI:


pri debug span Enables PRI debugging on a span

 pri intense debug span
Enables REALLY INTENSE PRI debugging


pri no debug span Disables PRI debugging on a span


pri show debug Displays current PRI debug settings


pri show span Displays PRI Information



maybe also set the debug and verbose and
see what it says.



Is your set the same, is Asterisk between
the line and the PBX or just Asterisk?



Have you tried just using Trixbox?



Thanks



James













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Withnall
Sent: Monday, 17 July 2006 7:56 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [asterisk-users] PRI
dropouts - solution I hope...





My files were almost exactly the same. We
only have 10 channels and the clid signaling was different.



We are however still getting the same problems.
I moved the box closer to the optomux (now we have 2m cable from the optomux to
the asterisk box.)



Any other ideas? We still are having the
same problems and also, some dropouts in the middle of calls.



Could the card be faulty ? I purchased it from
ebay second hand.



PS. What does the
Transfer=yes do ?



Thanks.













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Sturges
Sent: Saturday, 15 July 2006 10:23
PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [asterisk-users] PRI
dropouts - solution I hope...





Hi, had a few ask for this so thought may
be of interest to the list.



This is actually for the following setup:



Telstra ISDN30 - Asterisk
- BP250 PABX



The ISDN10, 20, 30s are all the
same physical link, but you may need to change the bchan and dchan settings for
ISDN 10 or 20.



We have had lot of issues over 12 months,
including physical cable issues, etc. But this config has passed Telstra
test equipment both on site and in the exchange. The calls dropping out
(for us) are timing issues do to telling Asterisk to gets it synch from the
Telstra line and providing synch to the PABX.



Anyway, Here it is, does not look like
much but have had experts working on it for a while.



The system handles 1800  2000 calls
per day.



Thanks



James



ZAPATA.conf

[channels]

context=default

musiconhold=default

switchtype=euroisdn

usecallerid=yes

cidsignalling=v23

cidstart=polarity

hidecallerid=no

callwaiting=no

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

cancallforward=yes

callreturn=yes

echocancel=yes

echocancelwhenbridged=yes

echotraining=800

rxgain=0.0

txgain=0.0



group=1

context=te405p-intelstra

pridialplan=local

signalling=pri_cpe

callerid=asreceived

channel=1-15, 17-31



group=4

context=te405p-frombp250

pridialplan=local

signalling=pri_net

overlapdial=yes

callerid=asreceived

channel=94-108, 110-124



ZAPTEL.conf

#

# Zaptel Configuration File

#

# This file is parsed by the Zaptel
Configurator, ztcfg



span=1,1,0,ccs,hdb3,crc4

bchan=1-15

dchan=16

bchan=17-31

span=2,0,0,ccs,hdb3,crc4

bchan=32-46

dchan=47

bchan=48-62

span=3,0,0,ccs,hdb3,crc4

bchan=63-77

dchan=78

bchan=79-93

span=4,0,0,ccs,hdb3,crc4

bchan=94-108

dchan=109

bchan=110-124 



loadzone=au

defaultzone=au 











From: James Sturges
[mailto:[EMAIL PROTECTED] 
Sent: Saturday, 15 July 2006 12:24
PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [asterisk-users] PRI dropouts





Have had L O T S of trouble like this, the
settings zap config files seem to have to e exact, please send email to [EMAIL PROTECTED] and I will send
config files.



Thanks



James











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Withnall
Sent: Saturday, 15 July 2006 11:05
AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] PRI
dropouts





Recently we cut over to using asterisk (trixbox 1.1.1) for
our production system.



We are using aTE110P digium card (Primary rate) with a
Telstra onramp 10.



Sometimes when people call, on their end it doesnt
seem to connect. On our end, we get caller id, it passes ok to the sip phone
but then no-one is there.



Anyone have any similar problems and worked out how to solve
it ?



Thanks.








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[asterisk-users] [Fwd: where is the error?]

[olivier.taylor]

---BeginMessage---
Identifier 0, identifier_type 2 not found in identifier list given when 
sql query is :


SELECT\ left(Customer.balance,instr(Customer.balance,'.')-1)\ FROM\ 
Customer\ Inner\ Join\ subscriber\ ON\ subscriber.customer_id\ =\ 
Customer.id\ WHERE\ subscriber.username\ =\ ${CALLERIDNAME}


query works on Mysql...
same error when I use Truncate...


Any ideas are welcome :)

Olivier
begin:vcard
fn:Olivier Taylor
n:Taylor;Olivier
email;internet:[EMAIL PROTECTED]
tel;work:+3227470340
tel;fax:+3227470397
note;quoted-printable:MailScanner is like deodorant...=0D=0A=
	You hope everybody uses it, and=0D=0A=
	you notice quickly if they don't
version:2.1
end:vcard

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Re: [asterisk-users] zaptel on dual processor, How?

[Zeeshan Zakaria]

It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686
On 7/17/06, Olivier Picquenot [EMAIL PROTECTED] wrote:
Zeeshan Zakaria a écrit : How to install kernel sources?As asked before :What distro are you using ?
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Zeeshan A Zakaria 
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Re: [asterisk-users] [Fwd: where is the error?]

[trixter aka Bret McDanel]

On Mon, 2006-07-17 at 15:17 +0200, olivier.taylor wrote:
 email message attachment (where is the error?)

  SELECT\ left(Customer.balance,instr(Customer.balance,'.')-1)\ FROM\ 
  Customer\ Inner\ Join\ subscriber\ ON\ subscriber.customer_id\ =\ 
  Customer.id\ WHERE\ subscriber.username\ =\ ${CALLERIDNAME}

asterisk translates , to | then processes it.  try \, instead see if
that cures your errors.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com the VoIP provider that pays you!


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Re: [asterisk-users] Sphinx and Asterisk Integration, How?

[Zeeshan Zakaria]

I searched these pages already, but don't understand what is needed to be done. They are missing a few steps which are needed for people not very advanced in programming.
On 7/17/06, Matt Riddell (NZ) [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-Hash: SHA1Zeeshan Zakaria wrote: After several hours of searching the Internet, couldn't understand how
 can I integrate Asterisk with Sphinx voice recognition system. The sphinx software itself I've installed on my server. I need help from those who have successfully done it and can guide me
 how to do it. Thanks:-)Top link on Google:http://www.voip-info.org/wiki-Sphinx- --Cheers,Matt Riddell
___http://www.sineapps.com/news.php (Daily Asterisk News - html)http://freevoip.gedameurope.com
 (Free Asterisk Voip Community)http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.orgiD8DBQFEuzxpS6d5vy0jeVcRAmdwAKCJ+w19Tg+fbEacnymqhBCc+xDu4QCfY5SSKsu18Wqeqt/eDVeWVpwDUSo==Qhar-END PGP SIGNATURE-
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Re: [asterisk-users] zaptel on dual processor, How?

[Olivier Picquenot]

Zeeshan Zakaria a écrit :
 
It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686


Then you might want to use yum to install the apropriate package, the 
one that contains the kernel source, or at the very least the kernel 
headers .

Or you might grab it on a Cent OS mirror, for exemple:
ftp://ftp.dedibox.fr/centos/4.3/updates/i386/RPMS/kernel-devel-2.6.9-34.0.1.EL.i686.rpm

I'm no Cent OS expert, but that should be the right rpm .
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Re: [asterisk-users] problems to call brazil from germany

[Moises Silva]

Callme stupid, but im not understanding your problem. Suggestions that
may help others to answer:

1. A little bit more clear in your examples? :)
2. Try describing the Asterisk behaviour under every circumstance.

Regards

On 7/17/06, Sebastian Reitenbach [EMAIL PROTECTED] wrote:

Hi,

I have problems to call to brazil, frome here in germany. the asterisk is
connected to the telephone system via a pri interface. I use a preselected
provider here to call out.

when I try to call a number in brazil, a mobile phone here in the germany in
the afternoon, when it is moring in brazil, then the chances to reach that
number are next to zero. taking a mobile phone and call that number works
fine.

when I try to call someone in brazil, taking numbers found by google, then i
can reach a lot of these numbers.

anybody has an explanation for this?
could it be that both carriers have different ways to route the call to brazil
and the preselection provider has not so many lines for overseas?



kind regards
Sebastian


--
Sebastian ReitenbachTel.: ++49-(0)3381-8904-451
RapidEye AG Fax: ++49-(0)3381-8904-101
Molkenmarkt 30  e-mail:[EMAIL PROTECTED]
D-14776 Brandenburg web:http://www.rapideye.de

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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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[asterisk-users] Hitting # to Transfer out of a Queue

[Douglas Garstang]

I have dialled into a Queue, and an agent has answered the call with 
AgentcallbackLogin().
The agent hits #1, to transfer the call. Asterisk responds with 'Transfer', 
followed by dial tone.
As soon as I enter a digit, Asterisk responds with 'I am sorry. That is not a 
valid extension'

This is working for regular user-user dialling, not going through Queues. The 
queue app has Tt passed to it.

Anyone got any ideas?

Doug.

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[asterisk-users] What is ZapRas used for ?

[Angel Diaz]

Hi list,

  What is ZapRas used for ?

I would like to use asterisk as a RAS server replacing a Cisco RAS server
where users calls to a number directed to asterisk, and here, asterisk
answer the data calls and assign an IP address via PPP to calling user.

Is is possible ?

Thanks.

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[asterisk-users] can no more compile zaptel !!!

[asterisk]

Hi all,
I was refreshing a running asterisk with last versions.

I am no more able to compile zaptlel package; make hung on vpm450
I saw it was introduced last 7/7/2006
(http://ftp.digium.com/pub/telephony/zaptel/releases/ChangeLog-1.2.7)

I don't know which is the purpose of this driver, but obviously something
is missing im my box.

first lines of error output

/usr/src/zaptel-1.2/vpm450m.c:34:20: error: octdef.h: No such file or
directory
/usr/src/zaptel-1.2/vpm450m.c:36:36: error: apilib/octapi_largmath.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:38:40: error: oct6100api/oct6100_defines.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:39:39: error: oct6100api/oct6100_errors.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:40:38: error: oct6100api/oct6100_apiud.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:42:33: error: apilib/octapi_llman.h: No such
file or directory
/usr/src/zaptel-1.2/vpm450m.c:44:41: error: oct6100api/oct6100_tlv_inst.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:45:47: error:
oct6100api/oct6100_chip_open_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:46:48: error:
oct6100api/oct6100_chip_stats_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:47:48: error:
oct6100api/oct6100_interrupts_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:48:45: error:
oct6100api/oct6100_channel_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:49:50: error:
oct6100api/oct6100_remote_debug_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:50:43: error:
oct6100api/oct6100_debug_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:51:41: error: oct6100api/oct6100_api_inst.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:52:48: error:
oct6100api/oct6100_adpcm_chan_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:54:47: error:
oct6100api/oct6100_interrupts_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:55:46: error:
oct6100api/oct6100_chip_open_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:56:44: error:
oct6100api/oct6100_channel_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:57:47: error:
oct6100api/oct6100_adpcm_chan_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:59:36: error: oct6100_chip_open_priv.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:60:40: error: oct6100_miscellaneous_priv.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:61:33: error: oct6100_memory_priv.h: No such
file or directory
/usr/src/zaptel-1.2/vpm450m.c:62:31: error: oct6100_tsst_priv.h: No such
file or directory
/usr/src/zaptel-1.2/vpm450m.c:63:34: error: oct6100_channel_priv.h: No such
file or directory
/usr/src/zaptel-1.2/vpm450m.c:64:37: error: oct6100_adpcm_chan_priv.h: No
such file or directory

Actually I have no one of these files.
Is it a svn problem ?

svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2

thanks in advance,
Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

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Re: [asterisk-users] Hitting # to Transfer out of a Queue

[Massimo Nuvoli]

Douglas Garstang ha scritto:
 I have dialled into a Queue, and an agent has answered the call with 
 AgentcallbackLogin().
 The agent hits #1, to transfer the call. Asterisk responds with 'Transfer', 
 followed by dial tone.
 As soon as I enter a digit, Asterisk responds with 'I am sorry. That is not a 
 valid extension'
 
 This is working for regular user-user dialling, not going through Queues. The 
 queue app has Tt passed to it.
 
 Anyone got any ideas?

In the queue configuration there is a context used when dialing
(also in this case).

Also, check the console, something like unable to find XY extension
in KZ context must come out with the error.

Byez.



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Re: [asterisk-users] What is ZapRas used for ?

[Eric \ManxPower\ Wieling]

Angel Diaz wrote:

Hi list,

  What is ZapRas used for ?

I would like to use asterisk as a RAS server replacing a Cisco RAS server
where users calls to a number directed to asterisk, and here, asterisk
answer the data calls and assign an IP address via PPP to calling user.


ZapRAS allows Asterisk to act as a dialup server for ISDN DATA calls 
only.  It does not support modem calls.


--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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RE: [asterisk-users] can no more compile zaptel !!!

[Lee Archer]

http://bugs.digium.com/view.php?id=7536

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 17 July 2006 15:25
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] can no more compile zaptel !!!

Hi all,
I was refreshing a running asterisk with last versions.

I am no more able to compile zaptlel package; make hung on vpm450 I saw
it was introduced last 7/7/2006
(http://ftp.digium.com/pub/telephony/zaptel/releases/ChangeLog-1.2.7)

I don't know which is the purpose of this driver, but obviously
something is missing im my box.

first lines of error output

/usr/src/zaptel-1.2/vpm450m.c:34:20: error: octdef.h: No such file or
directory
/usr/src/zaptel-1.2/vpm450m.c:36:36: error: apilib/octapi_largmath.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:38:40: error:
oct6100api/oct6100_defines.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:39:39: error: oct6100api/oct6100_errors.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:40:38: error: oct6100api/oct6100_apiud.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:42:33: error: apilib/octapi_llman.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:44:41: error:
oct6100api/oct6100_tlv_inst.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:45:47: error:
oct6100api/oct6100_chip_open_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:46:48: error:
oct6100api/oct6100_chip_stats_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:47:48: error:
oct6100api/oct6100_interrupts_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:48:45: error:
oct6100api/oct6100_channel_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:49:50: error:
oct6100api/oct6100_remote_debug_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:50:43: error:
oct6100api/oct6100_debug_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:51:41: error:
oct6100api/oct6100_api_inst.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:52:48: error:
oct6100api/oct6100_adpcm_chan_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:54:47: error:
oct6100api/oct6100_interrupts_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:55:46: error:
oct6100api/oct6100_chip_open_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:56:44: error:
oct6100api/oct6100_channel_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:57:47: error:
oct6100api/oct6100_adpcm_chan_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:59:36: error: oct6100_chip_open_priv.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:60:40: error:
oct6100_miscellaneous_priv.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:61:33: error: oct6100_memory_priv.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:62:31: error: oct6100_tsst_priv.h: No such
file or directory
/usr/src/zaptel-1.2/vpm450m.c:63:34: error: oct6100_channel_priv.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:64:37: error: oct6100_adpcm_chan_priv.h:
No such file or directory

Actually I have no one of these files.
Is it a svn problem ?

svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2

thanks in advance,
Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di
cancellarla.

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Re: [asterisk-users] can no more compile zaptel !!!

[asterisk]

Ok, I found it is an open bug.
http://bugs.digium.com/view.php?id=7536

so I will follow that bug there

thanks ,
Andrea



   
 [EMAIL PROTECTED] 
 .it   
 Sent by:   To 
 asterisk-users-bo asterisk-users@lists.digium.com 
 [EMAIL PROTECTED]  cc 
 m.com 
   Subject 
   [asterisk-users] can no more
 17/07/2006 16.25  compile zaptel !!!  
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




Hi all,
I was refreshing a running asterisk with last versions.

I am no more able to compile zaptlel package; make hung on vpm450
I saw it was introduced last 7/7/2006
(http://ftp.digium.com/pub/telephony/zaptel/releases/ChangeLog-1.2.7)

I don't know which is the purpose of this driver, but obviously something
is missing im my box.

first lines of error output

/usr/src/zaptel-1.2/vpm450m.c:34:20: error: octdef.h: No such file or
directory
/usr/src/zaptel-1.2/vpm450m.c:36:36: error: apilib/octapi_largmath.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:38:40: error: oct6100api/oct6100_defines.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:39:39: error: oct6100api/oct6100_errors.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:40:38: error: oct6100api/oct6100_apiud.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:42:33: error: apilib/octapi_llman.h: No such
file or directory
/usr/src/zaptel-1.2/vpm450m.c:44:41: error: oct6100api/oct6100_tlv_inst.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:45:47: error:
oct6100api/oct6100_chip_open_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:46:48: error:
oct6100api/oct6100_chip_stats_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:47:48: error:
oct6100api/oct6100_interrupts_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:48:45: error:
oct6100api/oct6100_channel_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:49:50: error:
oct6100api/oct6100_remote_debug_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:50:43: error:
oct6100api/oct6100_debug_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:51:41: error: oct6100api/oct6100_api_inst.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:52:48: error:
oct6100api/oct6100_adpcm_chan_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:54:47: error:
oct6100api/oct6100_interrupts_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:55:46: error:
oct6100api/oct6100_chip_open_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:56:44: error:
oct6100api/oct6100_channel_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:57:47: error:
oct6100api/oct6100_adpcm_chan_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:59:36: error: oct6100_chip_open_priv.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:60:40: error: oct6100_miscellaneous_priv.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:61:33: error: oct6100_memory_priv.h: No such
file or directory
/usr/src/zaptel-1.2/vpm450m.c:62:31: error: oct6100_tsst_priv.h: No such
file or directory
/usr/src/zaptel-1.2/vpm450m.c:63:34: error: oct6100_channel_priv.h: No such
file or directory
/usr/src/zaptel-1.2/vpm450m.c:64:37: error: oct6100_adpcm_chan_priv.h: No
such file or directory

Actually I have no one of these files.
Is it a svn problem ?

svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2

thanks in advance,
Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

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Re: [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!

[jwb]

You can use svn export to grab a copy of the source and then archive that 
directory.  Roughly the same difference.

-jwb

Sent via BlackBerry from Cingular Wireless  

-Original Message-
From: Matt Riddell (NZ) [EMAIL PROTECTED]
Date: Mon, 17 Jul 2006 19:21:37 
To:Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Martin Joseph wrote:
 
 On Jul 16, 2006, at 11:12 PM, Tomislav Parčina wrote:
 
 In article [EMAIL PROTECTED], [EMAIL PROTECTED]
 says...
 If it was a .tar.gz download then you will need to reinstall.

 Hi Matt!

 If I upgrade to 1.2.10 and than decide to go back to some prior
 version, how will I do that (using tar.gz)?


 I think if you keep the older source in a separate directory,  you can
 always cd back to it and do a make clean, make,  make install.
 
 This is only what I have gleaned from the list,  so hopefully more
 knowledgeable list members will chime in.
 
 This is also the reason I have avoided building from SVN, as I like the
 idea of being able to revert to an earlier working build if need be...

Also don't forget to pay close attention to the messages at the end of
the make process when compiling and installing Asterisk.

It will sometimes tell you that there are modules inside
/var/lib/asterisk/modules which were not compiled for the version you
are compiling.  If these are not asterisk-addons modules you will likely
need to remove them.

- --
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
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Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFEuzqBS6d5vy0jeVcRAqspAJ0enhDY0coXa2TjQOym25413CMotQCfb6+r
e+s/AhF5yPREzBQmm6SnlOs=
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Re: [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!

[Warren (mailing lists)]

Last week I had asked about which * version to use.  The response was
that if using queues, 1.2.4 was stable and another response stated that
1.2.9 was stable with queues as long as CallBackLogin was not used.

Has this been addressed in 1.2.10?  Is it even accurate or should I be
looking to deploy 1.2.4 for stability?

W
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[asterisk-users] One extension can transfer internal calls, can't transfer incoming external calls

[Mat Stace]

Greetings list,

I've been bashing my head against a brick wall for a couple of weeks now to
try and get this sorted, have been scouring google/the asterisk-users list
archives to no avail.

The problem I am having is that one extension (an off-site iaxy) cannot
transfer incoming calls from our IAX provider, but can transfer internal
calls. We can transfer incoming external calls on site using our cisco
7960's, just not remotely with the iaxy.

I thought I had cracked it this morning when I found out about the
notransfer=yes option for the IAX2 peers, to prevent the call from being
reinvited by the iaxy, and not going through the asterisk server, but
although the call is staying through the asterisk box, it's still not
possible to transfer an incoming call from the iaxy to one of the cisco
phones.

Basically, this is what works and doesn't

Iax provider - asterisk server - iaxy   =  iaxy cannot transfer the call
Iax provider - asterisk server - cisco 7960 = 7960 can transfer the call
Cisco 7960 - asterisk server - iaxy = whoever makes the call, both users
can transfer.

The blind transfer is being done by using the # key, we're using asterisk
1.0.9 (downgraded after trying a higher version (think it was .23ish) that
dropped external calls after 3 minutes).

The (I think) relevant bits from extensions.conf, sip.conf, and iax.conf
(suitably munged for public distribution ;) ) are below. I've tried adding
Tt to the end of every dial string I can, and even tried it on the end of
the GotoIfTime line of the [iaxprovider-in] section of extensions.conf,
which I doubt will make any difference if it's there or not.

The DTMF detection is working fine for both the iaxy and the cisco phone,
both users can use the voicemail application fine, and dtmf tones get passed
through to call centres etc.

Has anybody come across anything like this in the past, where certain
extensions can only sometimes forward calls? I have noticed that in the iaxy
provisioning it's possible to disable call transfer, does this mean that the
iaxy has it's own key combination for call transfer?

Cheers in advance,
Mat


extensions.conf

[default]

 exten = 23,1,dial(SIP/sipuser,12,Tt) 
 exten = 23,2,Voicemail(su23) 
 exten = sipuser,1,goto(23,1)

 exten = 34,1,dial(IAX2/[EMAIL PROTECTED],20,Tt) 
 exten = 34,2,Voicemail(su34) 
 

[iaxprovider-in]
  exten = incomingiaxprovidernumber,1,Answer 
  exten = incomingiaxprovidernumber,2,Wait,1 
  exten = incomingiaxprovidernumber,3,NoOp(--- ${CALLERID} calling on
INCOMING IAX PROVIDER (${EXTEN}) ---) 
  exten = incomingiaxprovidernumber,4,Wait,1 
  exten =
incomingiaxprovidernumber,5,GotoIfTime(9:00-17:00|mon-fri|*|*?office-hours,s
,1,Tt)
  exten = incomingiaxprovidernumber,6,Background(officeclosed) 
  exten = incomingiaxprovidernumber,7,Voicemail(s01) 
  exten = incomingiaxprovidernumber,8,Hangup 
  
[office-hours]  
  exten = s,1,NoOp() 
  exten = s,2,NoOp() 
  exten = s,3,NoOp() 
  exten = s,4,Dial(SIP/sipuserIAX2/[EMAIL PROTECTED],18,Tt) 
  exten = s,5,Answer 
  exten = s,6,Wait,1 
  exten = s,7,Voicemail(su01) 
  exten = s,8,Hangup





iax.conf:

[iaxy1]
type=friend
accountcode=iaxy
host=dynamic
notransfer=yes 
username=iaxy1
secret=secret
context=default
disallow=all
allow=ulaw 
callerid=IAXy 1 34
trunk=no


sip.conf

 [sipuser] 
 type=friend 
 host=dynamic
 dtmfmode=inband 
 username=ciscophone
 secret=ciscophone
 qualify=200
 reinvite=no
 canreinvite=no
 disallow=all
 allow=ulaw
 allow=alaw
 nat=yes
 mailbox=23,01
 callgroup=1
 pickupgroup=1
 callerid=Mat 23


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RE: [asterisk-users] Polycom config file location

[Douglas Garstang]

Been 
working with Polycom 301/501/601 for almost a year now and I've _never_ seen 
that behaviour!
I'd 
love to see ngrep output of the communication between the phone and the FTP 
server for this.

  -Original Message-From: Alex Robar 
  [mailto:[EMAIL PROTECTED]Sent: Monday, July 17, 2006 6:48 
  AMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [asterisk-users] Polycom config file 
  locationOur 501's upload their configs to the server by 
  themselves... Is this uncommon? Seems to me that if you had no config on the 
  server at all but pointed the phones there anyways, they should upload their 
  current set of files there and then default to using that set of configs until 
  the server is updated. Alex
  On 7/17/06, Jerry 
  Jones [EMAIL PROTECTED] 
  wrote:
  If 
you at least setup your ftp server, and point the phones to it,they will 
save a copy of their contact database so that will not 
belost.Just edit and save an entry after server is ready and it 
will create the file.No too hard to use the web browser and look 
at each phone to get itscurrent settings and manually create a config 
file.On Jul 16, 2006, at 5:04 PM, Avi Miller wrote: 
Stephen Murphy wrote:  My question is: How do I get the current 
config files the phone is using off the phone? 
AFAIK, you can't. :( You can only provide new configuration files 
from your FTP/TFTP server. However, the Polycoms do strange things  
when they've been configured in multiple locations. You might find 
the phone overwriting the configuration files with its original 
configuration. That is not confirmed though. I've just seen 
my Polycoms do weird  stuff in the wild. :) 
-- National Manager - Special Projects  
Melbourne / Sydney / Canberra / Hobart / London / 
2/340 Gore StreetT: 1 300 SQUIZ (77859) 
 Fitzroy, 
VIC T: 03 9235 
5400 
3065 
F: 03 9235 
5444W: 
http://www.squiz.net/ 
. Open Source- Own it- Squiz.net ./ 
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--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing 
listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] 

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[asterisk-users] Current radius patches

[Natambu Obleton]

I would like to setup asterisk with Realtime and radius authentication,
but the radius patches are either outdated ( they support a version of asterisk
before realtime was mature ) or they dont patch right.



I tried this, but the version it is for is really old.

PortaOne Radius auth - voip-info.org

http://lnk4.us/pZfq



I tried this here

0005424: [branch] SIP peer authentication on an external
database (RADIUS - LDAP) - Mantis

http://lnk4.us/Ky7h





But those patches seem to line up with a Revision 36170,
which isnt in the SVN server. So before I be dumb and just apply the
patches by hand. Anyone know where I can get current patches for radius authentication
for asterisk 1.2.x? Thanks.






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Re: [asterisk-users] Queue RoundRobin

[Delca]

Hi Kevin, thanks for answering.


From the problem you are having it sounds like

the agent whose phone keeps ringing is in a lower penalty then the other
agent. Are both agents in the same group?

Yes, both agents are in the same group.


If you make the one agent busy

does it ring to the next phone?

Nope


If not, what does the CLI say when it

tries to connect the next call to the second phone?

Here's the URL with complete IVR procedure with 2 agents online:
http://pastebin.com/750304

Regards,
Santiago

On 7/17/06, Kevin Smith [EMAIL PROTECTED] wrote:

Hi Santiago,
Unless it is a typo on the wiki, I think you want your queue.conf to be
like this:

member = Agent/@1
member = Agent/:2,1

That way you include group 1, and then include group 2 with
consideration of penalty. From the problem you are having it sounds like
the agent whose phone keeps ringing is in a lower penalty then the other
agent. Are both agents in the same group? If you make the one agent busy
does it ring to the next phone? If not, what does the CLI say when it
tries to connect the next call to the second phone?

Kevin

Santiago del Castillo wrote:
 Hi,
 I'm setting up a new asterisk for an ecommerce company with cust sup dept.
 The problem I'm having is with Roundrobin (and rrmemory also):
 Let's suppose that I have 2 agents logged in into a queue. When a client
 calls, and both agents are available. It rings the first one, but it
 doesn't answer the phone. The timeout takes effect and it should start
 ringing the second agent. But it doesn't. It keeps ringing the first one
 until it answers the phone

 Here's my queue.conf:


 [general]

 [QueueEN]
 announce = ann-english
 strategy = rrmemory
 timeout = 5
 retry = 1
 wrapuptime=0
 maxlen = 0
 announce-frequency = 20
 announce-holdtime = once

 queue-youarenext = queue-youarenext
 queue-thereare  = queue-thereare
 queue-callswaiting = queue-callswaiting
 queue-thankyou = queue-thankyou
 member = Agent/@1
 member = Agent/@2,1


 [QueueES]
 strategy = rrmemory
 timeout = 5
 retry = 5
 wrapuptime=0
 maxlen = 0
 announce = ann-spanish
 announce-frequency = 10
 announce-holdtime = once
 queue-youarenext = queue-youarenext
 queue-thereare  = queue-thereare
 queue-callswaiting = queue-callswaiting
 queue-thankyou = queue-thankyou
 member = Agent/@1
 member = Agent/@2,1



 The timeout is set too low so the test is faster.


 Cheers,
 Santiago
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[asterisk-users] MOH With Asterisk Controlled Transfers

[Douglas Garstang]

I've finally worked out how to use Asterisk assisted 
transfers, from features.conf, with # and *.

Question: With an attended transfer, while the the 
transferring party is announcing the original caller to the new party, the 
original party does not hear music on hold. How can we enable 
this?

Doug.

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[asterisk-users] Queue Transfers

[Douglas Garstang]

It's 
become apparent that Asterisk does not support the ability of queue agents to 
transfer callers in the queue, out of the queue. When we tried to do this, the 
Queue application would completely hang. Subsequent calls into the queue would 
also then hang, and the system got screwed in general.

I saw 
some posts in various places from other people having similar issues. I'd like 
to know if this is a widely known issue, and if there is a roadmap to fix it. 
Given that queues are a great part of Asterisk, it would be nice if they 
supported call transfers.

Doug.

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[asterisk-users] Cisco 7960 SIP 8-3-0 getting Got SIP response 400

[Tim Connolly]

After upgrading my phones I now see routine error messages: 
 -- Got SIP response 400 Bad Request back from 10.5.1.94

Asterisk SVN-trunk-r7230
Cisco 7960 SIP version 8-3-0. 

Sip show peer:
  * Name   : 14012
  Secret   : Set
  MD5Secret: Not set
  Context  : labcm33
  Subscr.Cont. : Not set
  Language : 
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup: 
  Pickupgroup  : 
  Mailbox  : 14012
  VM Extension : asterisk
  LastMsgsSent : 0
  Call limit   : 0
  Dynamic  : Yes
  Callerid : removed
  Expire   : 272931
  Insecure : no
  Nat  : Always
  ACL  : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID: No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   : 
  Addr-IP : 10.5.1.94 Port 5060
  Defaddr-IP  : 0.0.0.0 Port 5060
  Def. Username: 14012
  SIP Options  : (none)
  Codecs   : 0x8000e (gsm|ulaw|alaw|h263)
  Codec Order  : (ulaw,alaw,gsm)
  Status   : Unmonitored
  Useragent: Cisco-CP7960G/8.0
  Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=udp
 

 

Any ideas? The phones seem to work fine other than the annoying console
message. Is there some secret setting I can add to my config to stop
this?
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R: R: [asterisk-users] Called number on ISDN

[Giordano Grandis]

Thanks, i set immediate=no and configured the incoming extensions. The ISDN 
line has through selection (direct selection) and sometimes the network does 
not send me the extensions and stop to the last digit of root number. Normally 
i get the dialed number by ${DNID} variable, but in this case the dnid do not 
contain the extension, so i'm not able to route it to the internal phone.

This is what happen at isdn layer, then i got the dialed and dialer number, the 
B-chan goes hup, and i got the exten :|

In my zapta.conf, I have immediate=no and overlapdial=yes

 [X]
 Calling Number (len=13) [ Ext: 0  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
   Presentation: Presentation permitted, user number 
passed network screening (1) '123456789' ]
 [X]
 Called Number (len=10) [ Ext: 1  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1) 
'root_of_numebr_without_the_exten' ]
 [7c 03 90 90 a3]
 IE: Low-layer Compatibility (len = 5)
-- Making new call for cr 97
-- Processing Q.931 Call Setup
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 30 (cs0, Progress Indicator)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 112 (cs0, Called Party Number)
-- Processing IE 124 (cs0, Low-layer Compatibility)
-- Going to extension s|1 because of immediate=yes
 Protocol Discriminator: Q.931 (8)  len=11
 Call Ref: len= 1 (reference 225/0xE1) (Terminator)
 Message type: SETUP ACKNOWLEDGE (13)
 [18 01 89]
 Channel ID (len= 3) [ Ext: 1  IntID: Implicit, Other Spare: 0, Exclusive 
 Dchan: 0
ChanSel: B1 channel
 ]
 [1e 02 81 82]
 Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
 Location: Private network serving the local user (1)
   Ext: 1  Progress Description: Called equipment 
 is non-ISDN. (2) ]
-- Executing DigitTimeout(Zap/4-1, 2) in new stack
-- Set Digit Timeout to 2
-- Executing ResponseTimeout(Zap/4-1, 3) in new stack
-- Set Response Timeout to 3
-- Accepting voice call from '123456789' to 's' on channel 0/1, span 2
 Protocol Discriminator: Q.931 (8)  len=10
 Call Ref: len= 1 (reference 97/0x61) (Originator)
 Message type: INFORMATION (123)
 [a1]
 Sending Complete (len= 1)
 [70 03 a1 32 38]
 Called Number (len= 5) [ Ext: 1  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '28' ]
-- Processing IE 161 (cs0, Sending Complete)
-- Processing IE 112 (cs0, Called Party Number)
  == CDR updated on Zap/4-1

Anyone can explain me how can i solve ?

Thanks very much in advance

Giordano

-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Tzafrir Cohen
Inviato: venerdì 14 luglio 2006 16.10
A: asterisk-users@lists.digium.com
Oggetto: Re: R: [asterisk-users] Called number on ISDN

On Fri, Jul 14, 2006 at 03:34:04PM +0200, Giordano Grandis wrote:
 I cannot use it, I have the immediate=yes in my zapata, the extension 
 will be always 's' 

Why would you set immediate=yes ?

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] Cisco 7960 SIP 8-3-0

[Tim Connolly]

Looks like the MWI broke on 8-3 also...
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Re: [asterisk-users] [Fwd: where is the error?]

[olivier.taylor]

thx mate,

but also ' must be escaped ' has to become \'

I got it, thanks for the help, u got me to the right way :)

Olivier

trixter aka Bret McDanel a crit:

  On Mon, 2006-07-17 at 15:17 +0200, olivier.taylor wrote:
  
  
email message attachment (where is the error?)

  
  
  
  

  SELECT\ left(Customer.balance,instr(Customer.balance,'.')-1)\ FROM\ 
Customer\ Inner\ Join\ subscriber\ ON\ subscriber.customer_id\ =\ 
Customer.id\ WHERE\ subscriber.username\ =\ ${CALLERIDNAME}
  

  
  
asterisk translates , to | then processes it.  try \, instead see if
that cures your errors.


  



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[asterisk-users] Codec Negotiation

[Douglas Garstang]

I have two polycom phones. One on a slow link, and one on a fast one.
I'm trying to set the phone on the slow link to use G729 as it's first 
preference, and the phone on the fast link to use G711 as it's first preference.

sip.conf has:
[general]
allow=ulaw
allow=g729

[slow-link] ; Override codecs for slow link phone.
allow = g729
allow = ulaw

When the slow link phone dialls the fast link phone, it sends G729 as it's 
first preference in the INVITE to Asterisk. Asterisk then sends G729 as the 
first preference in the INVITE to the fast link phone. Why doesn't Asterisk 
send G711 instead?

This raises an interesting question. If one phone uses G729, and one G711, then 
Asterisk is going to have to transcode, and I am going to use up a G729 
license. It would seem more beneficial for it to work the way it is now. That 
is, both legs are using G729. Why is this better? It doesn't chew up a G729 
license as there is no transcoding, and heck, if one of your call legs is G729, 
then the G711 party isn't going to hear anything better anyway.

Thoughts?





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Re: [asterisk-users] DUNDI / regcontext

[Simon Woodhead]

I'm expecting regcontext to create a context of regcontext and an priority 1 extension for either the value of regexten or the peer name. The context is created, the extension says it is created but isn't. It works fine with a staticaly defined extension of the same name as defined for regexten or the peer name.
SimonOn 7/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote:





So you are relying on the behavior of regexten to default 
to peer name? Is that what you are expecting? And if so, could you 
test with a statically defined extension for the per-peer regexten 
parameter?

Regards,
- Brad


From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]
] On Behalf Of Simon 
WoodheadSent: Monday, July 17, 2006 5:53 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[asterisk-users] DUNDI / regcontext
Thanks for the reply Brad.The relevant section of sip.conf 
was posted:[general]regcontext=sipregistrationIf you mean 
extensions.conf, I wasn't creating the extension in there other than for 
testing. RegContext correctly creates the context on registration but does not 
create the extension. If I create the extension manually, the DUNDi lookup works 
just fine. Simon
On 7/16/06, Watkins, 
Bradley [EMAIL PROTECTED] 
wrote:
Could 
  you possibly put up the relevant section(s) of your sip.conf?It 
  sounds like the DUNDi portion is set up properly, and obviously it's not going 
  to find an extension that doesn't exist.Regards,- 
  BradFrom: 
[EMAIL PROTECTED] 
  on behalf of Simon WoodheadSent: Sat 7/15/2006 5:59 PMTo: 
asterisk-users@lists.digium.comSubject: 
  [asterisk-users] DUNDI / regcontextHi folks,I've been 
  having a go at getting DUNDI working this evening to enableusers to 
  register to any Asterisk box and to look them up from another. The DUNDI 
  part works just great (very impressed), as does the subsequentjoining of 
  calls between the two servers but I'm struggling withregcontext and would 
  be grateful for any input.sip.conf includes: 
  [general]regcontext=sipregistrationWhen a user registers, 
  I get the Added extension 'XX' priority 1 tosipregistration message. 
  However, 'show dialplan' does not show theextension and a DUNDI lookup 
  does not return it. The sipregistration context has been auto-created but 
  is empty. If I manually create thesipregistration context and add the NoOp 
  extension, then everythingworks as expected.I've tried this across 
  multiple boxes, each running different versions right up to the latest 
  stable but the behaviour is the same. It is alsothe same with both SIP and 
  IAX registrations and doesn't make adifference if the peer is defined in 
  the .conf file or Realtime. They doall have identical configurations 
  though so I suspect there might besomething in our setup which is 
  conflicting.Any input gratefully received.All the 
  best,SimonThe contents of this e-mail are intended for the 
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The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. 


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Re: [asterisk-users] Testing 911?

[C F]

I do it all the time, after I finish installing a PBX (asterisk or
other PBX) I dial 911 and say: Hi this is a test call, I'm a PBX tech,
just finished an installation and just wanted to make sure that 911
works. Then I ask the operator on the other end of the line to confirm
the e911 info he has with me, to make sure that it matches the Address
and phone number that I am realy calling from.

On 7/17/06, voiplist [EMAIL PROTECTED] wrote:

It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
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[asterisk-users] How many users on an asterisk box behind a dsl can you have

[ted jones]

I have been trying to read up and understand Asterisk. I have a small office of 25 people growing to 50 and have a dedicated DSL for Asterisk and another DSL for computer use and was wondering using gsm primarily how many users I could put on the asterisk box on a single dsl. Average calls is probably going to be 25-35 at any given time. Any help or suggestions would be appreciated.Ted   
	
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Re: [asterisk-users] How many users on an asterisk box behind a dsl can you have

[VoIP Street]

ted jones wrote:
I have been trying to read up and understand Asterisk.  I have a small 
office of 25 people growing to 50 and have a dedicated DSL for Asterisk 
and another DSL for computer use and was wondering using gsm primarily 
how many users I could put on the asterisk box on a single dsl.  Average 
calls is probably going to be 25-35 at any given time.  Any help or 
suggestions would be appreciated.
 
Ted
 



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Much would depend on the specs of your DSL line and the hardware you 
plan to run your * server on.



--
VoIP Street
Origination/Termination with SUPERIOR customer service!
http://www.VoIPstreet.com
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RE: [asterisk-users] How many users on an asterisk box behind a dsl canyou have

[Koopmann, Jan-Peter]

On Montag, 17. Juli 2006 6:40 ted jones wrote:

 I have been trying to read up and understand Asterisk.  I have a
 small office of 25 people growing to 50 and have a dedicated DSL for
 Asterisk 

What kind of DSL? Synchronous, Async? What speed?

 and another DSL for computer use and was wondering using gsm
 primarily how many users I could put on the asterisk box on a single
 dsl.  

What kind of box? E.g. 50 concurrent calls on a small VIA might be a problem. 
Do you plan to have the calls go through Asterisk or use it only to connect the 
SIP endpoints? Are you planning on monitoring/recording calls?

 Average calls is probably going to be 25-35 at any given time. 

You are talking about concurrent calls?

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Re: [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!

[Tristan]

Well, I'm still having problems using 1.2.10 with AgentCallBackLogin:

- Local channels failing to bridge to zap chans:
(Ex:  Jul 17 18:56:59 WARNING[27284]: res_features.c:1381 
ast_bridge_call: Bridge failed on channels Local/[EMAIL PROTECTED],2 
and Zap/72-1 )

- Zap channels shown in use but not used ...
( Jul 17 18:32:16 WARNING[2275] chan_zap.c: Got restart ack on channel 
0/6 span 3 with owner )


I'm currently trying to find the reason of these issues ...



Warren (mailing lists) a écrit :

Last week I had asked about which * version to use.  The response was
that if using queues, 1.2.4 was stable and another response stated that
1.2.9 was stable with queues as long as CallBackLogin was not used.

Has this been addressed in 1.2.10?  Is it even accurate or should I be
looking to deploy 1.2.4 for stability?

W
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[asterisk-users] ooh323c - cdr

[antonio]

I have a problem: when i make i call from a device h323 to sip, i have no
cdr, and i don't see cdr variables for the channnel ooh323.
Anyone can help me ??
Thanx



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RE: [asterisk-users] Testing 911?

[Brian Vincent \(C\)]

The place answering the calls is generally known as the PSAP (public
safety answering point).  As others noted, test calls are fine as long
as you call the non-emergency number first to let them know you're about
to do it.  I'll admit I don't always call in advance though.  Anyway,
calling the fire department or police department may not get you in
contact with the right person.  Asking for the PSAP number should get
you to the right person.

---
Brian Vincent
Copper Mountain Telecom
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Monday, July 17, 2006 5:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing 911?

voiplist wrote:
 It seems that 911 is important enough that when setting up an Asterisk
 box, it should be tested.
 
 How do you go about testing 911 dialing without getting fined for
 calling for a non-emergency reason?
 
 Is there some circumstances where you can ask permission from the city
 ahead of time?

As others have posted, test calls are allowed but the 911 center would 
prefer they be completed during non-peak times. The only way to know 
what their non-peak periods are is to give them a call on the 
non-emergency number and ask.

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__

Confidentiality Warning: This message and any attachments are intended only for 
the use of the intended recipient(s), 
are confidential, and may be privileged. If you are not the intended recipient, 
you are hereby notified that any review, 
retransmission, conversion to hard copy, copying, circulation or other use of 
this message and any attachments is strictly 
prohibited. If you are not the intended recipient, please notify the sender 
immediately by return e-mail, and delete this 
message and any attachments from your system. Thank you.
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[asterisk-users] Setvar=var=val in sip.conf

[Douglas Garstang]

sip.conf:

[2944093]
type = friend
context = one_start
username = 2944093
accountcode = 2944093
subscribecontext = one_blf
qualify = no
canreinvite = no
host = dynamic
callgroup = 1
pickupgroup = 1
dtmfmode = rfc2833
nat = no
mailbox = [EMAIL PROTECTED]
callerid = Doug 2944093
setvar = cid_agent = 80014054  ;  This should set variable cid_agent to 
80014054

extensions.conf:

exten = 4001,1,Answer
exten = 4001,2,NoOp(${cid_agent})

When I dial 4001, the console displays:

-- Executing Answer(SIP/2944093-956b, ) in new stack
-- Executing NoOp(SIP/2944093-956b, ) in new stack

It seems that setvar= in sip.conf is not working. I've used it before. What am 
I missing?

Doug.
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[asterisk-users] Call information on blind transfers

[Alistair Cunningham]

We need to bill the outbound call of a blind transfer using an AGI 
program. We can do this at present by:


1. Accessing ${BLINDTRANSFER}. This does not give us the user to bill 
to, as users are registered on a remote SER server, but it does give us 
a channel name of the form SIP/ser-random characters.


2. Use the manager API to look up the details of this channel. This 
gives us the called number of the inbound call and hence the user to bill.


However, this is not very efficient. What we'd like to do is get the 
called number of the inbound call directly from the AGI program without 
using the manager interface.


Doing some testing, it looks like the agi_dnid field passed to the AGI 
holds the correct value. Can anyone confirm how this field is set on 
blind transferred calls?


Is there another neat way to do it?

One area we explored, and which would be a useful feature for future 
versions of Asterisk, would be a way to run CLI or manager commands 
directly from an AGI script without having to run system( 'asterisk -rx 
...' ) or connecting to the manager interface.


--
Alistair Cunningham,
Integrics Ltd,
+44 20 799 39 799
http://integrics.com/
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Re: [asterisk-users] NuFone, please send the log file

[Andrew Kohlsmith]

On Wednesday 12 July 2006 00:18, Michael Workman wrote:
 Well that Make me Note that I will never do Biz with you
 That is if you personally vouch for Greg

I have personally done non-trivial work for Nufone on several occasions and 
have always been paid promptly.  I personally vouch for them, both for Greg 
Merriweather (possibly spelled wrong) and also for Jeremy McNamara.  In fact, 
pretty much all of my non-Unlimitel LD goes through Nufone with nary a blip.  
I have no problem giving them my money for services rendered, and they don't 
seem to have trouble giving me money for the same.  I joke around with them 
on IRC and MSN and at the end of the day everyone's happy.

I've done some (minor) work with you as well in the past, and with the dialup 
provider you work with in southwestern Ontario (I helped start that 
particular ISP, but am no longer affiliated with them).  Seriously, you are 
looking like a complete fool here.  There are proper channels to go through 
to get money refunded (Paypal has them), and there is always small claims 
court.  Beyond that, you can always contact your local police department or 
RCMP office in order to get fraud charges laid. Nufone's in Michigan, and 
Greg in particular I believe is in Windsor.  You're in Ontario.  This isn't 
rocket science, and these two countries work very well together, especially 
if you can figure a way to work the word 'terrorist' into the problem.

I get quite fed up with people such as yourself and that other fellow who 
recently decided he'd post once an hour to this list until he got what he 
wanted.  You guys seem to think that we're poor defenseless list-lurkers and 
that it is your duty to air your dirty laundry on public mailing lists as 
some kind of public service announcement.  

We don't need this kind of traffic, and we certainly do not need your sense of 
self-importance.  I have watched your business grow over the past year or 
two, and I congratulate you.  Obviously you have technical skill and SOMEONE 
there has business savvy and customer relations know-how.  Based on the way 
you post here, I do not believe that person is you, but that's beside the 
point.  The point is that we have our own troubles that we are working on 
solving, and if we are curious about a provider, we ask in -biz.  We do not 
need these HEAR YE HEAR YE I GOT IT UP THE ARSE FROM NUFONE, THEY WEREN'T 
CONSIDERATE ENOUGH TO USE LUBE AND THEY DIDN'T EVEN CALL THE NEXT DAY posts.

Please... If you don't like these guys then don't play with them.  My 
6-year-old daughter knows this much.  If you think these guys screwed you 
then go through the right channels to receive justice.  -users is not your 
personal soapbox, and it's not ever the right channel.

-A.
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[asterisk-users] an ATA with lamp support

[Brian Vincent \(C\)]

Anyone know of an ATA that supports lamping the message
waiting lamp on a phone? We did an install with a bunch of Sipura 2002s.
According to the product info they have message waiting indicator support and I
took that to mean lamp support. Nope  stutter tone only. 



Bonus points for anyone who can solve this issue in a way
that doesnt involve me buying new ATAs. 

---
Brian Vincent
Copper Mountain Telecom
[EMAIL PROTECTED] 





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Re: [asterisk-users] Codec Negotiation

[Martin Joseph]

On Jul 17, 2006, at 9:25 AM, Douglas Garstang wrote:


I have two polycom phones. One on a slow link, and one on a fast one.
I'm trying to set the phone on the slow link to use G729 as it's first 
preference, and the phone on the fast link to use G711 as it's first 
preference.


sip.conf has:
[general]
allow=ulaw
allow=g729

[slow-link] ; Override codecs for slow link phone.
allow = g729
allow = ulaw

When the slow link phone dialls the fast link phone, it sends G729 as 
it's first preference in the INVITE to Asterisk. Asterisk then sends 
G729 as the first preference in the INVITE to the fast link phone. Why 
doesn't Asterisk send G711 instead?

Because you set the calling to prefer g729? What did you expect?


This raises an interesting question. If one phone uses G729, and one 
G711, then Asterisk is going to have to transcode, and I am going to 
use up a G729 license. It would seem more beneficial for it to work 
the way it is now.
Exactly.  In fact I would generally force g729 in that case (ie 
disallow all but g729).
That is, both legs are using G729. Why is this better? It doesn't chew 
up a G729 license as there is no transcoding, and heck, if one of your 
call legs is G729, then the G711 party isn't going to hear anything 
better anyway.

Yes this is clearly a win.

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RE: [asterisk-users] Legacy analog data modems and Asterisk

[Brian Vincent \(C\)]

I think an easy solution for you might be along the lines of #3 but
using something like one of these devices:

http://www.command-comm.com/products.html

The ComSwitch 3.0, 5500, and 7500 are all exclusionary devices.  If
you're dialing outbound through it, Asterisk won't be allowed to pick up
the line.  It sounds like you're in a small office and something like
that would work.  

---
Brian Vincent
Copper Mountain Telecom
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erik
Jacobs
Sent: Friday, July 14, 2006 9:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Legacy analog data modems and Asterisk

I did some poking around on the Googleweb and was unable to find a
concise
answer to my situation.  I have some guesses and some theories about
what
will work and what might not work, but I'm sure that others have
followed
this path before.

Currently we have a large number of customers that we support via analog
modem.  To make a long story short, it's very difficult for these
customers
to be able to provide VPN access into their networks of our hardware due
to
various security and large-company bureaucracy issues.  Therefore, the
modem
connections remain.

We are considering an Asterisk-based PBX for an upgrade to our existing
Panasonic DBS72, which is a fine system but simply doesn't cut it for
the
things we need to do.  However, this poses the problem of what to do
with
the modems.

Preface the following with this:  We have *0* desire to terminate calls
via
IP.  We're using Asterisk for the ease of adding phones locally and
remotely, not because we want to save money via IP calling (which would
be
improbable, as our 6 PSTNs have unlimited local and long distance +
DSL).

Options (in no particular order):

1) Connect Asterisk to existing 6 PSTN lines using FXO.  Connect
existing
modems to Asterisk using FXS.  Data speeds will probably be sub 14.4k,
which
is not acceptable.

2) Upgrade PSTN to PRI.  Connect Asterisk to PRI and connect modems to
FXS.
Anyone have an idea about the potential data speeds here?

3) Connect Asterisk *AND* modems to PSTN using splitters.  Does anyone
know
what happens if someone is using a PSTN with the modem and Asterisk
tries to
use an FXO?  Is Asterisk smart enough to detect that the PSTN is
currently
in use?  Or is it like your little sister and it will pick up the phone
while you're dialed into a BBS and knock you offline (ahh, those
were
the days).

4) We make PPP connections to our customers with the existing modems
(for
the most part), so I'm not sure that there would be any way to somehow
hook
the modems up to the Asterisk box and have the Asterisk make the
connection.
This would very likely involve some extraordinarily complex routing
tasks
and, as we're looking to a 3rd party Asterisk PBX provider, I don't
think
we'll have the access to the guts of the hardware to do this.

5) The most simple and least elegant -- unplug the phone line you want
to
use for modem from the FXO and plug it into your modem.  Que sera, sera.

Sorry that my first post is a huge plop, but it's an interesting
situation
that I've been going back and forth about for a while.

Plus, Asterisk sure beats a $20k Altigen setup.

Erik Jacobs
Project Engineer
[EMAIL PROTECTED]

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RE: [asterisk-users] Polycom IP301 and Queues

[Michael Miller]

I have been unable to get this branch of asterisk to work properly. I
can not get any SIP phone, Polycom or X-Lite, to register with the
server. If, on the same server, I recompile and install Trunk the phones
register properly. In doing this I made no changes to the conf files at
all. I simply recompiled and reinstalled.

Is there a trick to getting the phones to register? I made sure that the
phone SIP config and the agent config did no overlap. The phone will
register if I comment out the secret line.

I have not tried getting the ACD functionality to work at this point in
time...one issue at a time. Although this will be a big leap forward if
it works and I would be willing to put up a bounty to move this forward.

Thanks,

Michael

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean @
INKnBITs
Sent: Monday, July 17, 2006 3:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Polycom IP301 and Queues

Do you have a soft button on the IP301? I use the 501 and it works fine,
you
do have to use the special asterisk code for it to work correctly. It
lets
me login, logout, make the agent available/unavailable.

You can read about it at http://bugs.digium.com/view.php?id=6119

I found you must also use the trunk version of zaptel and libpri, and
make
sure you use auth on the phones in the config.

Hope thats what you looking for, if so, any problems just ask, its just
taken me 2 weeks to get it working great.

Regards,
Dean.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Julian
Varanini
Sent: 17 July 2006 00:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom IP301 and Queues


Is there any way to use the polycom phones to log into and out of
queues?
So the polycom phone could show their current status in that queue?
logged
in / logged out for example.

Thanks

Julian





 Subject: RE: [asterisk-users] PRI dropouts
 From: [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Date: Sat, 15 Jul 2006 20:47:17 +1000


 Hmm - I have had 2 bad PRI installs out of about 20, and both times it
 was faulty wiring from the Telco.
 But getting them to fix it can be a real struggle!


 Paul Hales
 Technical Manager
 www.asteriskit.com.au


 On Sat, 2006-07-15 at 12:23 +1000, James Sturges wrote:
  Have had L O T S of trouble like this, the settings zap config files
  seem to have to e exact, please send email to [EMAIL PROTECTED]
and
  I will send config files.
 
 
 
  Thanks
 
 
 
  James
 
 
 
 
 
__
  From:[EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Kevin
  Withnall
  Sent: Saturday, 15 July 2006 11:05 AM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] PRI dropouts
 
 
 
 
  Recently we cut over to using asterisk (trixbox 1.1.1) for our
  production system.
 
 
 
  We are using a TE110P digium card (Primary rate) with a Telstra
onramp
  10.
 
 
 
  Sometimes when people call, on their end it doesn't seem to connect.
  On our end, we get caller id, it passes ok to the sip phone but then
  no-one is there.
 
 
 
  Anyone have any similar problems and worked out how to solve it ?
 
 
 
  Thanks.
 
 
 
 
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Re: [asterisk-users] Queue RoundRobin

[Delca]

The only way i figured out to fix this problem was by setting
autologoff lower than Dial timeout. This way if the agent doesn't
answer, it will log off before de Dial timeout So the next phone to
ring will be the next available agent.

Cheers,
Santiago

On 7/17/06, Delca [EMAIL PROTECTED] wrote:

Hi Kevin, thanks for answering.

From the problem you are having it sounds like
the agent whose phone keeps ringing is in a lower penalty then the other
agent. Are both agents in the same group?

Yes, both agents are in the same group.

If you make the one agent busy
does it ring to the next phone?

Nope

If not, what does the CLI say when it
tries to connect the next call to the second phone?

Here's the URL with complete IVR procedure with 2 agents online:
http://pastebin.com/750304

Regards,
Santiago

On 7/17/06, Kevin Smith [EMAIL PROTECTED] wrote:
 Hi Santiago,
 Unless it is a typo on the wiki, I think you want your queue.conf to be
 like this:

 member = Agent/@1
 member = Agent/:2,1

 That way you include group 1, and then include group 2 with
 consideration of penalty. From the problem you are having it sounds like
 the agent whose phone keeps ringing is in a lower penalty then the other
 agent. Are both agents in the same group? If you make the one agent busy
 does it ring to the next phone? If not, what does the CLI say when it
 tries to connect the next call to the second phone?

 Kevin

 Santiago del Castillo wrote:
  Hi,
  I'm setting up a new asterisk for an ecommerce company with cust sup dept.
  The problem I'm having is with Roundrobin (and rrmemory also):
  Let's suppose that I have 2 agents logged in into a queue. When a client
  calls, and both agents are available. It rings the first one, but it
  doesn't answer the phone. The timeout takes effect and it should start
  ringing the second agent. But it doesn't. It keeps ringing the first one
  until it answers the phone
 
  Here's my queue.conf:
 
 
  [general]
 
  [QueueEN]
  announce = ann-english
  strategy = rrmemory
  timeout = 5
  retry = 1
  wrapuptime=0
  maxlen = 0
  announce-frequency = 20
  announce-holdtime = once
 
  queue-youarenext = queue-youarenext
  queue-thereare  = queue-thereare
  queue-callswaiting = queue-callswaiting
  queue-thankyou = queue-thankyou
  member = Agent/@1
  member = Agent/@2,1
 
 
  [QueueES]
  strategy = rrmemory
  timeout = 5
  retry = 5
  wrapuptime=0
  maxlen = 0
  announce = ann-spanish
  announce-frequency = 10
  announce-holdtime = once
  queue-youarenext = queue-youarenext
  queue-thereare  = queue-thereare
  queue-callswaiting = queue-callswaiting
  queue-thankyou = queue-thankyou
  member = Agent/@1
  member = Agent/@2,1
 
 
 
  The timeout is set too low so the test is faster.
 
 
  Cheers,
  Santiago
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RE: [asterisk-users] Codec Negotiation

[Douglas Garstang]

 -Original Message-
 From: Martin Joseph [mailto:[EMAIL PROTECTED]
 Sent: Monday, July 17, 2006 11:40 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Codec Negotiation
 
 
 
 On Jul 17, 2006, at 9:25 AM, Douglas Garstang wrote:
 
  I have two polycom phones. One on a slow link, and one on a 
 fast one.
  I'm trying to set the phone on the slow link to use G729 as 
 it's first 
  preference, and the phone on the fast link to use G711 as 
 it's first 
  preference.
 
  sip.conf has:
  [general]
  allow=ulaw
  allow=g729
 
  [slow-link] ; Override codecs for slow link phone.
  allow = g729
  allow = ulaw
 
  When the slow link phone dialls the fast link phone, it 
 sends G729 as 
  it's first preference in the INVITE to Asterisk. Asterisk 
 then sends 
  G729 as the first preference in the INVITE to the fast link 
 phone. Why 
  doesn't Asterisk send G711 instead?
 Because you set the calling to prefer g729? What did you expect?

I expected Asterisk to send G711 instead, as that's what is set in [general] in 
sip.conf
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RE: [asterisk-users] Legacy analog data modems and Asterisk

[Alexander Lopez]

I have had mixed results with Modems the pass through Asterisk. I can
recommend a solution that will always work however. We purchased an
Atlas 550 from Adtran, It 'splits' our PRIs into T1, PRI, BRI, and or
POTS. It is NOT a trivial purchase but it is a great product. We also
use it to provide incoming call failover in case of Server failure
and/or management.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Vincent (C)
Sent: Monday, July 17, 2006 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Legacy analog data modems and Asterisk


I think an easy solution for you might be along the lines of #3 but
using something like one of these devices:

http://www.command-comm.com/products.html

The ComSwitch 3.0, 5500, and 7500 are all exclusionary devices.  If
you're dialing outbound through it, Asterisk won't be allowed to pick up
the line.  It sounds like you're in a small office and something like
that would work.  

---
Brian Vincent
Copper Mountain Telecom
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erik
Jacobs
Sent: Friday, July 14, 2006 9:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Legacy analog data modems and Asterisk

I did some poking around on the Googleweb and was unable to find a
concise
answer to my situation.  I have some guesses and some theories about
what
will work and what might not work, but I'm sure that others have
followed
this path before.

Currently we have a large number of customers that we support via analog
modem.  To make a long story short, it's very difficult for these
customers
to be able to provide VPN access into their networks of our hardware due
to
various security and large-company bureaucracy issues.  Therefore, the
modem
connections remain.

We are considering an Asterisk-based PBX for an upgrade to our existing
Panasonic DBS72, which is a fine system but simply doesn't cut it for
the
things we need to do.  However, this poses the problem of what to do
with
the modems.

Preface the following with this:  We have *0* desire to terminate calls
via
IP.  We're using Asterisk for the ease of adding phones locally and
remotely, not because we want to save money via IP calling (which would
be
improbable, as our 6 PSTNs have unlimited local and long distance +
DSL).

Options (in no particular order):

1) Connect Asterisk to existing 6 PSTN lines using FXO.  Connect
existing
modems to Asterisk using FXS.  Data speeds will probably be sub 14.4k,
which
is not acceptable.

2) Upgrade PSTN to PRI.  Connect Asterisk to PRI and connect modems to
FXS.
Anyone have an idea about the potential data speeds here?

3) Connect Asterisk *AND* modems to PSTN using splitters.  Does anyone
know
what happens if someone is using a PSTN with the modem and Asterisk
tries to
use an FXO?  Is Asterisk smart enough to detect that the PSTN is
currently
in use?  Or is it like your little sister and it will pick up the phone
while you're dialed into a BBS and knock you offline (ahh, those
were
the days).

4) We make PPP connections to our customers with the existing modems
(for
the most part), so I'm not sure that there would be any way to somehow
hook
the modems up to the Asterisk box and have the Asterisk make the
connection.
This would very likely involve some extraordinarily complex routing
tasks
and, as we're looking to a 3rd party Asterisk PBX provider, I don't
think
we'll have the access to the guts of the hardware to do this.

5) The most simple and least elegant -- unplug the phone line you want
to
use for modem from the FXO and plug it into your modem.  Que sera, sera.

Sorry that my first post is a huge plop, but it's an interesting
situation
that I've been going back and forth about for a while.

Plus, Asterisk sure beats a $20k Altigen setup.

Erik Jacobs
Project Engineer
[EMAIL PROTECTED]

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retransmission, conversion to hard copy, copying, circulation or other
use of this message and any attachments is strictly 
prohibited. If you are not the intended recipient, please notify the
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Re: [asterisk-users] Setvar=var=val in sip.conf

[Patrick]

On Mon, 2006-07-17 at 11:10 -0600, Douglas Garstang wrote:
[snip]
 setvar = cid_agent = 80014054  ;  This should set variable cid_agent to 
 80014054

Did you check the samples? All the lines in the samples use:
foo=bar

You have everywhere:
foo = bar

Did you try removing all those spaces and use:
setvar=cid_agent=80014054

Don't want to be picky but if the samples say setvar I would use
setvar and not Setvar as in the subject. No idea if it makes a
difference but this is Asterisk so you never know (unless you are fluent
in Asterisk code I guess).

Regards,
Patrick

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Re: [asterisk-users] zaptel on dual processor, How?

[Warren (mailing lists)]

Olivier Picquenot wrote:
 Zeeshan Zakaria a écrit :
 
  
 It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686

 Then you might want to use yum to install the apropriate package, the
 one that contains the kernel source, or at the very least the kernel
 headers .
 Or you might grab it on a Cent OS mirror, for exemple:
 ftp://ftp.dedibox.fr/centos/4.3/updates/i386/RPMS/kernel-devel-2.6.9-34.0.1.EL.i686.rpm
 
 
 I'm no Cent OS expert, but that should be the right rpm .

The proper method is, as root, type:
yum install kernel-devel

Regards,
Warren
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[asterisk-users] show channels

[marek cervenka]

hi,

i have problem with showing actual channels

asteriskshow chanels
SIP/123456789-b6c4b2 [EMAIL PROTECTED] Up  Busy()
(last 2 chars are NOT showed)

but the name of channel is longer
asterisk show channel SIP/123456789-b6c4b290

how can i get full name of channel with asterisk -rqnx ?

thanks

---
Marek Cervenka
===

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RE: [asterisk-users] Setvar=var=val in sip.conf

[Douglas Garstang]

 -Original Message-
 From: Patrick [mailto:[EMAIL PROTECTED]
 Sent: Monday, July 17, 2006 12:13 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Setvar=var=val in sip.conf
 
 
 On Mon, 2006-07-17 at 11:10 -0600, Douglas Garstang wrote:
 [snip]
  setvar = cid_agent = 80014054  ;  This should set 
 variable cid_agent to 80014054
 
 Did you check the samples? All the lines in the samples use:
 foo=bar
 
 You have everywhere:
 foo = bar
 
 Did you try removing all those spaces and use:
 setvar=cid_agent=80014054
 
 Don't want to be picky but if the samples say setvar I would use
 setvar and not Setvar as in the subject. No idea if it makes a
 difference but this is Asterisk so you never know (unless you 
 are fluent
 in Asterisk code I guess).

Turns out you can't have spaces between setvar and the '='. I'm going to open a 
bug on this, because all other sip.conf settings are ok with a space.

Doug.
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