Re: [asterisk-users] Error in ubuntu dapper

[brandon kruz]

look for another, more configurable softphone if you can

but asterisk needs that for incoming sip(or maybe its iax, not sure)

you should be able to configure the differnet port
you want to use a softphone on your asterisk system
and btw netstat -a | grep 5060 worked for me

but who knows

let me know how it goes, im going to do some more research



From: don Paolo Benvenuto [EMAIL PROTECTED]
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Subject: Re: [asterisk-users] Error in ubuntu dapper
Date: Fri, 21 Jul 2006 21:10:14 -0400
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Yeah, ekiga was using port 5060, althoug netstat -a didn't say it.

I might issue netstat -alpn

The weird thing is that I had configured ekiga so that it used port
5061, but unfortunatly if ekiga is run before asterisk it catchs port
5060 too.

El vie, 21-07-2006 a las 18:13 -0500, brandon kruz escribió:
 in addition to russel
 use
 (in ubuntu)
 sudo netstat
 or man netstat for further, more precise methods
 look for your specific port
 eg
 sudo netstat -a | grep 5060
 and it shoudl tell you the process name, and what directory it is 
comming

 from
 shut it off
 and do that
 sudo netstat -a | grep 5060 again
 it should be clear
 then start asterisk :]

 keep us updated.

 the fact that it is from module chan_sip, as russel said
 i believe its a sip phone indeed(softphone, X-lite, SJ-phone, etc)

 
 On Fri, 2006-07-21 at 12:37 -0400, don Paolo Benvenuto wrote:
   Jul 21 12:31:51 WARNING[6333]: chan_sip.c:12637 reload_config: 
Failed to

 bind to 10.152.58.9:5060: Address already in use
 
 It looks like another application on your system is using port 5060.
 Did you install any new software such as a soft phone?
 
 If you are now using another application that wants to use port 5060,
 you will need to configure one of them to use a different port.
 

--
Buon Cammino!

don Paolo Benvenuto

Vuoi sapere di più su quello che succede qui?
leggi il mio diario a http://www.chiesamissionaria.it/diario

Visita l'enciclopedia libera, dove puoi contribuire anche tu:
http://it.wikipedia.org/

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RE: [asterisk-users] How to connect XLite with public IP?

[brandon kruz]

unfortunatly
its not a firewall problem(or at least thats not the main one)
the main one is forwarding the asterisk related ports to your IP(internal 
IP)

go to http://whatismyip.com
make sure u keep that information

then go to your asterisk box and type ifconfig(and get your itnernal IP)
then type your default gateway(your router) ip (usually 192.168.x.x, 
linksys=192.168.1.1, dlink=192.168.0.1, u can figure that out probably)


and type that into your itnernet browser
and forward ports to your asterisk box(the IP adress that asterisk is on)

hope this helps
`KruZ~


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   Hi Friends,

I have installed Asterisk in my house with public ip. Now, I want to 
connect to my Asterisk server from my office system with XLite softphone. I 
was given my Asterisk server public IP in my XLite Domain field. But, it is 
not connecting. It is giving an error i.e.,  Registration error: 408 - 
Request timedout. I tried using firewall and without using firewall. 
Please tell me how to configure my XLite softphone to connect with my 
Asterisk server?


Looking forward to your kind response. Thank you.

Regards,
Chandra.

-
Do you Yahoo!?
 Next-gen email? Have it all with the  all-new Yahoo! Mail Beta.




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Re: [asterisk-users] Digium TE110P IRQ

[Massimo Nuvoli]

Lincoln Zuljewic Silva ha scritto:
 Hello all. I have a Digium TE110P board and when I do a zap show
 status on CLI I get:
 DescriptionAlarms   
IRQbpviolCRC4
 Digium Wildcard TE110P T1/E1 Card 0OK00
  0
 IRQ = 0 ?

IRQ=0 means missed irq, or if you prefer, IRQ problem = 0, or NO
PROBLEM.

If this number grows (0) at this point you can think ouch! there is
a problem!.

But, after looking this, you can see your PRI going crazy.

Bye. :-)



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Re: [asterisk-users] Queue Persistence with queue.log

[lenz]

Hi Douglas,
you do not have to patch Asterisk to run QM, and you may use flat-file  
storage. And Java / JSP is because serving a very large call center  
requires quite a lot of raw number crunching and a lot of RAM to go with  
it, so PHP / Perl would not have been a viable option.


About your problem, you could always pull the stats from the manager  
interface before any reload/restart, and then add the results so you get  
your actual stats data. It should not be very hard to do.

l.


In data Fri, 21 Jul 2006 18:24:43 +0200, Douglas Garstang  
[EMAIL PROTECTED] ha scritto:


Yes, except that if the queue.log file was read by asterisk, and read at  
restart/reload, they could be pulled with the Manager interface. We are  
running three Asterisk boxes here in a cluster, and being able to pull  
the stats from the Manager interface is relatively easy.


I was just looking at QueueMetrics. I have to patch asterisk _and_ use  
MySQL? and JSP??? Good grief. Talk about overkill.


Doug.


-Original Message-
From: Lenz [mailto:[EMAIL PROTECTED]
Sent: Friday, July 21, 2006 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue Persistence with queue.log



I don't think there is. It would be rather overkill for what
the app_queue
does; there are a number of queue stats packages, commercial
and free,
that will provide a better approach to gathering stats for
the purpouse of
running a call center or an inbound queue.
l.


On Fri, 21 Jul 2006 16:49:39 +0200, Douglas Garstang
[EMAIL PROTECTED] wrote:

 All,

 Is there a way to have Asterisk read queue.log on startup,
or reload, so
 that queue stats can be retained between restarts and
reboots? It'd
 would be especially nice on the reloads, as even a 'reload
app_queue.so'
 clears all your stats. That COMPLETELY sucks, as every time
you make a
 queue configuration change, you lose your stats for ALL queues.

 Doug.



--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it

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[asterisk-users] NAT and externip problem or bug

[Robert Jenkins]

Hi,

I've recently got asterisk running on it's own pc inside my firwewall.
Mostly it's working fine, but there is one silly problem I can't figure out.
(For reference, Asterisk is the latest stable version as of last weekend
14th July. All connectivity is SIP or IAX).

I initially had 'externip' set to my public IP. I have the appropriate 5000
range ports forwarded to the asterisk PC and external calls seem OK.

The 'local' phones are a mixture of Sipura boxes and softphones.

Problem:
No or one-way audio in internal calls.

Reason: Asterisk appears to be using the 'externip' address for all SIP
devices, regardless of their NAT setting.
Once a call starts, some softphones change the address they are responding
to  use the external IP rather than the asterisk PCs local IP on the same
subnet...

I have tried all NAT options and spent quite a while reading everything I
can find about sip.conf, but I can't so far find any way of changing this
behaviour.

All the internal phones work fine if I comment out the externip line, but
then the connections outside the firewall are likely to have problems.

Is there any way of configuring externip on a per-device basis, or should it
only have effect on NATed devices?

Thanks,
Robert Jenkins.

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Re: [asterisk-users] NAT and externip problem or bug

[Julian J. M.]

Have you made sure you are also setting localnet in sip.conf?

externip=1.2.3.4
localnet=192.168.0.0/255.255.255.255

Asterisk won't use externip for devices on your local network.

Julian.

On 7/22/06, Robert Jenkins [EMAIL PROTECTED] wrote:

Hi,

I've recently got asterisk running on it's own pc inside my firwewall.
Mostly it's working fine, but there is one silly problem I can't figure out.
(For reference, Asterisk is the latest stable version as of last weekend
14th July. All connectivity is SIP or IAX).

I initially had 'externip' set to my public IP. I have the appropriate 5000
range ports forwarded to the asterisk PC and external calls seem OK.

The 'local' phones are a mixture of Sipura boxes and softphones.

Problem:
No or one-way audio in internal calls.

Reason: Asterisk appears to be using the 'externip' address for all SIP
devices, regardless of their NAT setting.
Once a call starts, some softphones change the address they are responding
to  use the external IP rather than the asterisk PCs local IP on the same
subnet...

I have tried all NAT options and spent quite a while reading everything I
can find about sip.conf, but I can't so far find any way of changing this
behaviour.

All the internal phones work fine if I comment out the externip line, but
then the connections outside the firewall are likely to have problems.

Is there any way of configuring externip on a per-device basis, or should it
only have effect on NATed devices?

Thanks,
Robert Jenkins.

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RE: [asterisk-users] NAT and externip problem or bug

[Guido Hecken]

posting the relevant parts of your config (sip.conf, extensions.conf) could
help to solve your problem. 

Guido 
 
 I've recently got asterisk running on it's own pc inside my firwewall.
 Mostly it's working fine, but there is one silly problem I can't figure
out.
 (For reference, Asterisk is the latest stable version as of last weekend
 14th July. All connectivity is SIP or IAX).
 
 I initially had 'externip' set to my public IP. I have the appropriate
5000
 range ports forwarded to the asterisk PC and external calls seem OK.
 
 The 'local' phones are a mixture of Sipura boxes and softphones.
 
 Problem:
 No or one-way audio in internal calls.
 
 Reason: Asterisk appears to be using the 'externip' address for all SIP
 devices, regardless of their NAT setting.
 Once a call starts, some softphones change the address they are responding
 to  use the external IP rather than the asterisk PCs local IP on the same
 subnet...
 
 I have tried all NAT options and spent quite a while reading everything I
 can find about sip.conf, but I can't so far find any way of changing this
 behaviour.
 
 All the internal phones work fine if I comment out the externip line, but
 then the connections outside the firewall are likely to have problems.
 
 Is there any way of configuring externip on a per-device basis, or should
it
 only have effect on NATed devices?
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Re: [asterisk-users] Unicall, not HOW but WHY

[Tim Panton]

On 20 Jul 2006, at 20:27, Barzilai wrote:

Warning: This message is a valid question, and is also kind of a  
[RANT] at the end... but I'm high on caffeine and I had fun writing  
it.
The ranting part more or less reflect the state of the Asterisk  
ecosystem until the end of 2005, which has been getting a little  
better but a lot of the garbage remains.

At least someone answer my questions :-)


Ok, I feel your pain, BUT, there is no point in taking it out on the
Asterisk community.

When I first encountered my local PRI variant  (Euro ISDN) 10 years ago
I tried to configure a $10k Dialog to talk to it, I couldn't get it to
work, until some guy on the Dialogic Forum told me to
to use the GUI interface and then take a texteditor to an ini file
to change an 0x0C to 0x0A and add a line
0x0A=0ff. That was to make it work with the standard
for the whole of western europe!

So, what I'm saying is that Telephony standards are messy,
and vary from country to country and carrier to carrier, so
inevitably you get weird config files.

Now, at least with Open source you can find out what options
exist from looking at the code, even if you don't
have an exact recipe for your carrier.

Do I wish it wasn't like that - sure - can it be fixed by
developers - no.

Tim Panton

www.mexuar.com



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[asterisk-users] cannot received calls in pstn line

[Lito Lampitoc]

Hi All,

I'm having problems receiving calls in my direct lines, but it's working fine in local lines (extensions). When a direct line is connected to my fxo it can't handle the call, but when an extension is connected it's ok.


Any suggestion?

thanks.

Lito
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Re: Re: [asterisk-users] setting call-limits

[voip]

Hi,

 on 1.2.4 and 1.2.7, we have to set the 'type=peer' for call-limits to
 work effectively.
 
 type=friend doesn't seem to enforce call limits at all.
 
 if you haven't tried type=peer, try that first.

No, this doesn't work. 


  as I read there should be the functionality in 1.2.10 to set
 call-limits. As I understood this option can be set in sip.conf per user just 
 typing
 
  call-limit=1
 
  to restrict the calls to 1 incoming or 1 outgoing line for this user.
 
  I'm using mysql for sip peers, so I added the attribute call-limit to
 the appropriate database table.
 
  ... It doesn't have any effect. Asterisk permits as many calls as I
 want.

Thanks
-- 


Echte DSL-Flatrate dauerhaft für 0,- Euro*. Nur noch kurze Zeit!
Feel free mit GMX DSL: http://www.gmx.net/de/go/dsl
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RE: [asterisk-users] NAT and externip problem or bug

[Robert Jenkins]

That sounds to be what I'm missing!

Many thanks, Robert.


 From: Julian J. M. 
 Sent: 22 July 2006 10:19
 Subject: Re: [asterisk-users] NAT and externip problem or bug
 
 Have you made sure you are also setting localnet in sip.conf?
 
 externip=1.2.3.4
 localnet=192.168.0.0/255.255.255.255
 
 Asterisk won't use externip for devices on your local network.
 
 Julian.
 
 On 7/22/06, Robert Jenkins [EMAIL PROTECTED] wrote:
  Hi,
 
  I've recently got asterisk running on it's own pc inside my 
 firwewall.
  Mostly it's working fine, but there is one silly problem I 
 can't figure out.
  (For reference, Asterisk is the latest stable version as of last 
  weekend 14th July. All connectivity is SIP or IAX).
 
  I initially had 'externip' set to my public IP. I have the 
 appropriate 
  5000 range ports forwarded to the asterisk PC and external 
 calls seem OK.
 
  The 'local' phones are a mixture of Sipura boxes and softphones.
 
  Problem:
  No or one-way audio in internal calls.
 
  Reason: Asterisk appears to be using the 'externip' address for all 
  SIP devices, regardless of their NAT setting.
  Once a call starts, some softphones change the address they are 
  responding to  use the external IP rather than the 
 asterisk PCs local 
  IP on the same subnet...
 
  I have tried all NAT options and spent quite a while reading 
  everything I can find about sip.conf, but I can't so far 
 find any way 
  of changing this behaviour.
 
  All the internal phones work fine if I comment out the 
 externip line, 
  but then the connections outside the firewall are likely to 
 have problems.
 
  Is there any way of configuring externip on a per-device basis, or 
  should it only have effect on NATed devices?
 
  Thanks,
  Robert Jenkins.
 
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Re: [asterisk-users] Germany VOIP provider

[Roger Schreiter]

Thameem Ansari schrieb:

...
understand why do they give this 01801 if it is similar to local number? 
I am also exploring the options to get some landlines for flat rate.



Hi,

local numbers make the other party think to call someone
in the respective area.
Thus reaching someone at a local number, who has nothing to
do with the respective local area, is spoofing.

That's why 0180x numbers are an alternative. 0180x
number do not implique any local relationship.

But when you intend to move to Germany, you probably
will have a local address. So what's the problem getting
a local phone number?

Some of the providers at

http://voipliste.de/privat.html

have also english web pages.


Roger.

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[asterisk-users] Re: Information about Softphone support G729 ?

[Adrian]

OK but i don't have enough resource codec G729, do you have reference ? If i am 
development with speex (8k) codec can running/compatible with device or 
softphone support G729 codec ?

Many Thanks

Regards
Adrian
Quoting Asterisk Expert [EMAIL PROTECTED]:


Dear Adrian,
G729 is not open codec. Therefore you won't be able to find any opensource
softphone having g729 codec. I'm not sure but you may need to get some
opensource softphone and implement codec yourself.

On 7/22/06, Adrian [EMAIL PROTECTED] wrote:


Dear All,

Anybody know about (open source with java or C++ ) Softphone  support G729
?

Thanks for your information

Regards
Adrian





--
Regards
Asterisk Expert




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Re: [asterisk-users] Associate manager events to a previous Originate action

[Johannes Zweng]

Stefan Reuter wrote:
add an ActionId property to your Originate action and you will receive
the same ActionId as part of the OriginateSuccess or OriginateFailure
event. The OriginateSuccess event also contains a link to the channel so
you can relate them after you received the OriginateSuccess event.
For the OriginateSuccess/-Failure events to be sent you must also set
Async to true when sending the Originate action.

Thank you very much! This did the trick!

I can associate the first channel to my Originate action by reading the
Uniqueid field of the OriginateSucess event and I also can associate the
second channel (created by the Dial() command) to the first one by reading
the fields of the Dial event.

This was exactly the hint I needed. Thanks!  :-)


Best regards,
John



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[asterisk-users] Upgrading my office - Need help

[Gary Guthary]

Hi -

If I'm posting this in the wrong place, pease forgive me

Folks, I need help...

One company I consult for is upgrading their office and will need a PBX
replacement in the next two months.

I'm seriously thinking of offering them an 'Asterisk' solution versus them
getting locked in with some PBX vendor.

This means I've got to come up with some sort of demo system to show them.

I've got the hardware. - Dedicated Linux box I can re-config and/or re-load
as needed. - Have one FXS/FXO card to demo intfc to telco. - And a bunch of
Cisco 79xx phones I can use for demos.

If I can get this rolling, I'll be more than happy to pay ***MONEY*** to
anybody who can help. - Also, I'm not afraid to pay for already-developed
admin software I can use to manage my system. - I just don't know 'which is
which'.

To prevent cluttering up this board, please send all responses to:
[EMAIL PROTECTED]

Thanks in advance  again apologize if this is not the right place to post
this.

Gary Guthary


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Re: [asterisk-users] Upgrading my office - Need help

[Mats Karlsson]

Gary Guthary wrote:

Hi -

If I'm posting this in the wrong place, pease forgive me

Folks, I need help...

One company I consult for is upgrading their office and will need a PBX
replacement in the next two months.

I'm seriously thinking of offering them an 'Asterisk' solution versus them
getting locked in with some PBX vendor.

This means I've got to come up with some sort of demo system to show them.

I've got the hardware. - Dedicated Linux box I can re-config and/or re-load
as needed. - Have one FXS/FXO card to demo intfc to telco. - And a bunch of
Cisco 79xx phones I can use for demos.

If I can get this rolling, I'll be more than happy to pay ***MONEY*** to
anybody who can help. - Also, I'm not afraid to pay for already-developed
admin software I can use to manage my system. - I just don't know 'which is
which'.

To prevent cluttering up this board, please send all responses to:
[EMAIL PROTECTED]

Thanks in advance  again apologize if this is not the right place to post
this.

Gary Guthary

Take a look at trixbox.org.

/M

--

Those that sacrifice essential liberty to obtain a little temporary 
safety deserve neither liberty nor safety. -- Ben Franklin (1759)

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Re: [asterisk-users] Associate manager events to a previous Originate action

[Richard Lyman]

Stefan Reuter wrote:


Johannes Zweng wrote:
 


Although I can associate every incoming event to a specific channel on
Asterisk (because of the Uniqueid field) I see no possibility to identify
without doubts which channels were created as a result of my Originate
action. 
   



add an ActionId property to your Originate action and you will receive
the same ActionId as part of the OriginateSuccess or OriginateFailure
event. The OriginateSuccess event also contains a link to the channel so
you can relate them after you received the OriginateSuccess event.
For the OriginateSuccess/-Failure events to be sent you must also set
Async to true when sending the Originate action.

=Stefan
 

iirc, current svn/trunk only has OriginateFailure.  (OriginateSuccess 
output was removed)


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[asterisk-users] SIP reinvite _and_ NAT

[Roger Schreiter]

Hi,

I have a sipphone behind a router doing NAT, an asterisk
box in the middle and another asterisk box, which works
as gateway to further destinations.

The asterisk box in the middle should do all call setup
and tear down, but no RTP. RTP should flow directly between
the sipphone via the router to the other asterisk box.

When calling _from_ the sipphone, everthing is fine:
The asterisk box in the middle is reinviting, and the
other asterisk box is finally exchanging RTP with the sipphone
in both directions.

When calling _to_ the sipphone, there is a problem:
The asterisk box in the middle again is reinviting, and
the RTP stream from the sipphone than goes directly to
the other asterisk box.

But the RTP stream from the other asterisk box is sent to
the private IP address of the sipphone (192.168)
The NAT workaround is not effective in this case.

Any hints?

Just for understanding: Who is responsible in this case
for the NAT workaround: The asterisk box in the middle
who is reinviting or the originating asterisk box who is
then sending the RTP traffic?


Roger.

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[asterisk-users] Re: Associate manager events to a previous Originate action

[Tony Mountifield]

In article [EMAIL PROTECTED],
Richard Lyman [EMAIL PROTECTED] wrote:
 iirc, current svn/trunk only has OriginateFailure.  (OriginateSuccess 
 output was removed)

Really??? If so, that would be a very retrograde step.

Fortunately, a look at the latest version of manager.c in trunk at
http://svn.digium.com/view/asterisk/trunk/manager.c?rev=38042view=markup
is reassuring, in the function fast_originate():

/* Tell the manager what happened with the channel */
manager_event(EVENT_FLAG_CALL,
res ? OriginateFailure : OriginateSuccess,

Comparing with 1.2, I see there were originally two calls to manager_event(),
one for OriginateFailure and another for OriginateSuccess.

They have now been combined into one, with a conditional event name,
which may have given rise to the mistaken impression if just skimming
the diffs cursorily.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Automating the registration process

[Tzafrir Cohen]

On Fri, Jul 21, 2006 at 10:26:20AM +0200, Kevin Class wrote:

 Query: is it possible to create an application that will write into
 sip_additional.conf and extensions_additional.conf, the registrations of new
 users. This application will be like a registration process for my users.
 What I'm interested on is how will the application automatically reload
 extensions and sips, when I get it working? The only way that I know how to
 reload extensions is through the cli, how will my application reload
 extensions?

Your application can issue CLI commands via the Asterisk Mnager
Interface.

BTW: as an alternative to rewriting files, consider using #exec to
generate the relevant parts of the config files on reload time by scripts


-- 
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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Re: [asterisk-users] Ztdummy

[David Cornejo]

At 09:27 PM 7/18/2006, you wrote:
Using
Zaptel-1.2.7
Asterisk 1.2.10
OS: CentOS 3.4

I am having a problem trying to get ztdummy and it wont
work. Here is what I did the following and got:
[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

FATAL: Error running install command for ztdummy
[EMAIL PROTECTED] ~]# 

What I find interesting is that timing will work. However I
don't feel comfortable letting the client use the system if this can
affect him in anyway. Thanks.

Dovid
I get this if I'm too quick to execute the commands - zaptel takes some
time to come up.

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Re: [asterisk-users] cannot received calls in pstn line

[Martin Joseph]

On Jul 22, 2006, at 3:00 AM, Lito Lampitoc wrote:


Hi All,

I'm having problems receiving calls in my direct lines, but it's  
working fine in local lines (extensions). When a direct line is  
connected to my fxo it can't handle the call, but when an extension  
is connected it's ok.


Any suggestion?

I think you might try rephrasing your question as it currently  
doesn't seem to make any sense?

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RE: [asterisk-users] NAT and externip problem or bug

[Robert Jenkins]

Oh well..

I already had localnet set:

localnet = 192.168.0.0  ; Internal NETWORK address
localmask = 255.255.255.0   ; Internal netmask 

All the involved PCs  Sipura boxes are using 192.168.0.x addresses.

The Sipura boxes work, but the fact that asterisk is sending the external IP
to any device on the local network seems to me to be a bug..

Robert Jenkins.

 Subject: RE: [asterisk-users] NAT and externip problem or bug
 
 That sounds to be what I'm missing!
 Many thanks, Robert.
 
  From: Julian J. M. 
  Sent: 22 July 2006 10:19
  Subject: Re: [asterisk-users] NAT and externip problem or bug
  
  Have you made sure you are also setting localnet in sip.conf?
  
  externip=1.2.3.4
  localnet=192.168.0.0/255.255.255.255
  
  Asterisk won't use externip for devices on your local network.
  
  Julian.
  
  On 7/22/06, Robert Jenkins [EMAIL PROTECTED] wrote:
   Hi,
  
   I've recently got asterisk running on it's own pc inside my
  firwewall.
   Mostly it's working fine, but there is one silly problem I
  can't figure out.
   (For reference, Asterisk is the latest stable version as of last 
   weekend 14th July. All connectivity is SIP or IAX).
  
   I initially had 'externip' set to my public IP. I have the
  appropriate
   5000 range ports forwarded to the asterisk PC and external
  calls seem OK.
  
   The 'local' phones are a mixture of Sipura boxes and softphones.
  
   Problem:
   No or one-way audio in internal calls.
  
   Reason: Asterisk appears to be using the 'externip' 
 address for all 
   SIP devices, regardless of their NAT setting.
   Once a call starts, some softphones change the address they are 
   responding to  use the external IP rather than the
  asterisk PCs local
   IP on the same subnet...
  
   I have tried all NAT options and spent quite a while reading 
   everything I can find about sip.conf, but I can't so far
  find any way
   of changing this behaviour.
  
   All the internal phones work fine if I comment out the
  externip line,
   but then the connections outside the firewall are likely to
  have problems.
  
   Is there any way of configuring externip on a per-device 
 basis, or 
   should it only have effect on NATed devices?
  
   Thanks,
   Robert Jenkins.

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Re: [asterisk-users] cannot received calls in pstn line

[Alex Robar]

Agreed... How are your extensions connected? What hardware are you using to connect your PSTN lines? Rephrase your question and we can probably help you out.AlexOn 7/22/06, 
Martin Joseph [EMAIL PROTECTED] wrote:
On Jul 22, 2006, at 3:00 AM, Lito Lampitoc wrote: Hi All, I'm having problems receiving calls in my direct lines, but it's working fine in local lines (extensions). When a direct line is
 connected to my fxo it can't handle the call, but when an extension is connected it's ok. Any suggestion?I think you might try rephrasing your question as it currentlydoesn't seem to make any sense?
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 http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar
[EMAIL PROTECTED]
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Re: [asterisk-users] NAT and externip problem or bug

[Martin Joseph]

On Jul 22, 2006, at 11:13 AM, Robert Jenkins wrote:


Oh well..

I already had localnet set:

localnet = 192.168.0.0  ; Internal NETWORK address
localmask = 255.255.255.0   ; Internal netmask

All the involved PCs  Sipura boxes are using 192.168.0.x addresses.

The Sipura boxes work, but the fact that asterisk is sending the  
external IP

to any device on the local network seems to me to be a bug..



You didn't mention whether you were also forwarding ports 1-2  
to the SIP Proxy (ie asterisk).  Thats where the actual RTP (voice  
data) is passing.  Also you need to be sure that there aren't  
multiple clients on your lan all trying to use the same ports for  
signaling (ie 5060), as this will fail.


Hope this helps.
Marty

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Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk

[Jonathan Attwood]

For the OP, do you have an entry against Display Name on the PSTN
tab, whilst logged in as admin/advanced? If I have an entry in this,
what you describe happens for me. If the field is empty, CLID is sent
correctly to my Asterisk box.



On 21/07/06, Rich Adamson [EMAIL PROTECTED] wrote:

I just ran into a problem with the spa3k and spa942's that I could not
diagnose. It appears as though the sipura boxes have a problem with
calls that include a CallerID with - in it. I can't say with 100%
certainty yet, but that's my story and I'm sticking to it (for now). ;)


Douglas Garstang wrote:
 -Original Message-
 From: Brian Capouch [mailto:[EMAIL PROTECTED]
 Sent: Friday, July 21, 2006 11:20 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to
 Asterisk


 Douglas Garstang wrote:
 I'm working with a Sipura 3000 ATA here. I'm trying to get
 incoming PSTN calls on the FXO port to go automatically to
 Asterisk. I have it working, but I had to configure the ATA
 to register with Asterisk, which means that all calls are
 being sent to Asterisk with a caller id of the username used
 to register with Asterisk.
 I want the real caller ID to be sent to Asterisk, which
 means I don't want the ATA to register. The badly written
 Sipura docs aren't clear about how to do this. Anyone set this up?
 That's not correct.

 My SPA-3000 FXO port registers with my Asterisk server, and when the
 PSTN calls come in, it uses the incoming caller's CallerID
 for the call.

 Sounds like you have something misconfigured.

 Here's my invite Brian. The From: is always going to contain the auth id the 
ATA used to register with Asterisk.

 INVITE sip:[EMAIL PROTECTED] SIP/2.0
 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK208dd2be;rport
 From: Cody XXX-527-7107 sip:[EMAIL PROTECTED];tag=as3a94778b
 To: sip:[EMAIL PROTECTED]
 Contact: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX
 Max-Forwards: 70
 Remote-Party-ID: Cody XXX-527-7107 sip:[EMAIL 
PROTECTED];privacy=off;screen=no
 Date: Fri, 21 Jul 2006 17:44:20 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
 Content-Type: application/sdp
 Content-Length: 269

 v=0
 o=root 28771 28771 IN IP4 xxx.187.142.203
 s=session
 c=IN IP4 xxx.187.142.203
 t=0 0
 m=audio 21652 RTP/AVP 0 18 101
 a=rtpmap:0 PCMU/8000
 a=rtpmap:18 G729/8000
 a=fmtp:18 annexb=no
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
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[asterisk-users] X100P clone not working

[Frank Darner]

Hi,

I have problem to set up an X100P clone card.
Installation of zaptel was successful.
Also modprobe of zaptel, ztdummy and wcfxo without problems.

kernel: wcfxo: module not supported by Novell, setting U taint flag.
kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - IRQ 193
kernel: wcfxo: DAA mode is 'FCC'
kernel: Found a Wildcard FXO: Generic Clone
kernel: Registered tone zone 0 (United States / North America)


But if I do #ztcfg -vvv no channels will be found.

the log file reports me: kernel: Registered tone zone 0 (United States / North 
America)

Iam quite sure that zaptel.conf is correct.
fxsks=1 
loadzone=us
defaultzone=us

So Iam not sure what happened here, I have found that this clone as used by 
many others, so it should work.

thanks

best regards

Frank
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Re: [asterisk-users] NAT and externip problem or bug

[brandon kruz]

specially if you only have 5060 but not the sip pass through ports as he 
mentioned
(1-2) then you can establish a call but no voice data is sent(or so 
that is in theory)


so make sure you get this right with your asterisk box and everything should 
work



From: Martin Joseph [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

Subject: Re: [asterisk-users] NAT and externip problem or bug
Date: Sat, 22 Jul 2006 11:21:08 -0700
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On Jul 22, 2006, at 11:13 AM, Robert Jenkins wrote:


Oh well..

I already had localnet set:

localnet = 192.168.0.0  ; Internal NETWORK address
localmask = 255.255.255.0   ; Internal netmask

All the involved PCs  Sipura boxes are using 192.168.0.x addresses.

The Sipura boxes work, but the fact that asterisk is sending the  external 
IP

to any device on the local network seems to me to be a bug..



You didn't mention whether you were also forwarding ports 1-2  to 
the SIP Proxy (ie asterisk).  Thats where the actual RTP (voice  data) is 
passing.  Also you need to be sure that there aren't  multiple clients on 
your lan all trying to use the same ports for  signaling (ie 5060), as this 
will fail.


Hope this helps.
Marty

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Re: [asterisk-users] Ztdummy

[brandon kruz]

im not sure of this error
but it has to be common if it is from to quick of a startup
(but the module zaptel is very quick to probe)
if worst comes to worst
this modprobeing should be in a startup script i imagine
you can just throw down a wait command(depending on what version of linux, 
and

what you are using as a language to startup the modprobe(bash? perl? etc.)

let me know if a little waiting helps

`KruZ~



From: David Cornejo [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

Subject: Re: [asterisk-users] Ztdummy
Date: Sat, 22 Jul 2006 08:07:24 -1000
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At 09:27 PM 7/18/2006, you wrote:

Using Zaptel-1.2.7
Asterisk 1.2.10
OS: CentOS 3.4

I am having a problem trying to get ztdummy and it wont work. Here is what 
I did the following and got:

[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

FATAL: Error running install command for ztdummy
[EMAIL PROTECTED] ~]#

What I find interesting is that timing will work. However I don't feel 
comfortable letting the client use the system if this can affect him in 
anyway. Thanks.


Dovid


I get this if I'm too quick to execute the commands - zaptel takes some 
time to come up.



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Re: [asterisk-users] cannot received calls in pstn line

[brandon kruz]

sounds like you have a TDM card
with an fxo port(FOR YOUR PSTN, i think)
and 1-3 fxs ports for your phone lines
make sure you have your zaptel.conf
and zapata.conf and extensions.conf right
with the context that your ports are in
and the signalling that it uses(oppisite of the port)
be careful because u can burn out your card and maybe phone line if u do 
this wrong



itll look like this
[incoming]
exten = 100,1,Answer()
exten = 100,2,Dial(Zap/1,20,rR)
exten = 101,1Answer()
exten = 101,2,Dial(Zap/2,20,rR)

etc...


let me know if this helps for i think thats what you are asking

the redmodule is the module you will plug your pstn line into
then you have to set it up in zapata.conf and zaptel.conf to show your 
signalling and context
obviously you will want to name it something like [incoming] and make sure 
you have an

[incoming] context in extensions.conf

`KruZ~



From: Alex Robar [EMAIL PROTECTED]
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Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

Subject: Re: [asterisk-users] cannot received calls in pstn line
Date: Sat, 22 Jul 2006 14:20:55 -0400
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Agreed... How are your extensions connected? What hardware are you using to
connect your PSTN lines? Rephrase your question and we can probably help 
you

out.

Alex

On 7/22/06, Martin Joseph [EMAIL PROTECTED] wrote:



On Jul 22, 2006, at 3:00 AM, Lito Lampitoc wrote:

 Hi All,

 I'm having problems receiving calls in my direct lines, but it's
 working fine in local lines (extensions). When a direct line is
 connected to my fxo it can't handle the call, but when an extension
 is connected it's ok.

 Any suggestion?

I think you might try rephrasing your question as it currently
doesn't seem to make any sense?
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--
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[EMAIL PROTECTED]




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Re: [asterisk-users] Ztdummy

[Tzafrir Cohen]

On Wed, Jul 19, 2006 at 03:27:03AM -0400, Dovid Bender wrote:
 Using Zaptel-1.2.7
 Asterisk 1.2.10
 OS: CentOS 3.4
 
 I am having a problem trying to get ztdummy and it wont work. Here is what I 
 did the following and got:
 [EMAIL PROTECTED] ~]# modprobe zaptel
 [EMAIL PROTECTED] ~]# modprobe ztdummy
 Notice: Configuration file is /etc/zaptel.conf
 line 0: Unable to open master device '/dev/zap/ctl'

The error is because of an unnecessary post-load action of the module
ztdummy.

just rem-out the 'ztdummy' line in the modprobe.conf that zaptel
installs. (or better: don't let it install anything there)

There is no need to configure ztdummy's span.

-- 
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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RE: [asterisk-users] X100P clone not working

[brandon kruz]

dmesg | grep x100p

also
im almost positive(correct me if im wrong anyone) you do NOT
need ztdummy
i repeat, do NOT need ztdummy with x100p

then rmmod zaptel  rmmod wctdm  modprobe zaptel  modprobe wctdm
or wctfxo or w/e

let me know how this works out
(post the section for your x100p from zaptel.conf and zapata.conf if you 
could m8)





Hi,

I have problem to set up an X100P clone card.
Installation of zaptel was successful.
Also modprobe of zaptel, ztdummy and wcfxo without problems.

kernel: wcfxo: module not supported by Novell, setting U taint flag.
kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - IRQ 
193

kernel: wcfxo: DAA mode is 'FCC'
kernel: Found a Wildcard FXO: Generic Clone
kernel: Registered tone zone 0 (United States / North America)


But if I do #ztcfg -vvv no channels will be found.

the log file reports me: kernel: Registered tone zone 0 (United States / 
North

America)

Iam quite sure that zaptel.conf is correct.
fxsks=1
loadzone=us
defaultzone=us

So Iam not sure what happened here, I have found that this clone as used by
many others, so it should work.

thanks

best regards

Frank
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Re: [asterisk-users] Ztdummy

[brandon kruz]

i thought so because ztdummy isnt even necessary
unistall and comment out :D

(rmmod ztdummy and look at your modprobe.conf as stated by cohen, ty cohen)

good luck, keep us updated.



From: Tzafrir Cohen [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Ztdummy
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On Wed, Jul 19, 2006 at 03:27:03AM -0400, Dovid Bender wrote:
 Using Zaptel-1.2.7
 Asterisk 1.2.10
 OS: CentOS 3.4

 I am having a problem trying to get ztdummy and it wont work. Here is 
what I did the following and got:

 [EMAIL PROTECTED] ~]# modprobe zaptel
 [EMAIL PROTECTED] ~]# modprobe ztdummy
 Notice: Configuration file is /etc/zaptel.conf
 line 0: Unable to open master device '/dev/zap/ctl'

The error is because of an unnecessary post-load action of the module
ztdummy.

just rem-out the 'ztdummy' line in the modprobe.conf that zaptel
installs. (or better: don't let it install anything there)

There is no need to configure ztdummy's span.

--
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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Re: [asterisk-users] X100P clone not working

[Frank Darner]

Hi,
i tried w/o ztdummy but nothing changed.

any ideas?


zaptel.conf
--
fxsks=1
loadzone=us
defaultzone=us

zapata.conf

[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
echocancle=yes
transfer=yes
echocanclewhenbridged=yes
channel = 1

# modprobe zaptel  modprobe wcfxo

kernel: zaptel: module not supported by Novell, setting U taint flag.
kernel: Zapata Telephony Interface Registered on major 196
kernel: Zaptel Version: SVN-tag-1.2.6-r1096 Echo Canceller: KB1
kernel: wcfxo: module not supported by Novell, setting U taint flag.
kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - IRQ 193
kernel: wcfxo: DAA mode is 'FCC'
kernel: Found a Wildcard FXO: Generic Clone
kernel: Registered tone zone 0 (United States / North America)

# lsmod | grep zaptel
zaptel201732  1 wcfxo
crc_ccitt   2176  1 zaptel


# ztcfg -v
Zaptel Configuration
==
0 channels configured.



 dmesg | grep x100p

 also
 im almost positive(correct me if im wrong anyone) you do NOT
 need ztdummy
 i repeat, do NOT need ztdummy with x100p

 then rmmod zaptel  rmmod wctdm  modprobe zaptel  modprobe wctdm
 or wctfxo or w/e

 let me know how this works out
 (post the section for your x100p from zaptel.conf and zapata.conf if you
 could m8)

 Hi,
 
 I have problem to set up an X100P clone card.
 Installation of zaptel was successful.
 Also modprobe of zaptel, ztdummy and wcfxo without problems.
 
 kernel: wcfxo: module not supported by Novell, setting U taint flag.
 kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - IRQ
 193
 kernel: wcfxo: DAA mode is 'FCC'
 kernel: Found a Wildcard FXO: Generic Clone
 kernel: Registered tone zone 0 (United States / North America)
 
 
 But if I do #ztcfg -vvv no channels will be found.
 
 the log file reports me: kernel: Registered tone zone 0 (United States /
 North
 America)
 
 Iam quite sure that zaptel.conf is correct.
 fxsks=1
 loadzone=us
 defaultzone=us
 
 So Iam not sure what happened here, I have found that this clone as used
  by many others, so it should work.
 
 thanks
 
 best regards
 
 Frank
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Re: [asterisk-users] FXS adapters and Polycom phones

[Tele Cost Price Reducer]

hi,
i would prefer a Mediant 1000 with 12 ports FXS of Audiocodes to do the Job.

further info available upon request,

Mickey
On 7/12/06, Mike [EMAIL PROTECTED] wrote:



Hi,

I`m looking for a SIP-PSTN adapter, basically to switch a customer from a cheap PBX to mine, but resuing their own Norstar PSTN phones. They have 10phones. From a price point of view, it seems that 10 individual GrandStream SIP adapters is the best way to go, but it seems so inelegant to me.


What is recommended ?

Second question: I have a GrandStream GXP-2000, that despite what everybody says I love. I am still looking for a replacement, if only because it doesn`t look as good and it does have a few quirks. I was looking at Polycoms, but some questions are unanswered by looking at their datasheet.

- Does the Polycom 501 have an integrated router (like the GXP-2000, latest firmware, does)
- Can you have more than one SIP/account on the phone, each ringing in a way that lets the user know which account is ringing? (GXP2000 does it by making it possible to have each line linked to a separate SIP account)


Thank you,

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RE: [asterisk-users] NAT and externip problem or bug

[Robert Jenkins]

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Martin Joseph
 Sent: 22 July 2006 19:21
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] NAT and externip problem or bug
 
 
 On Jul 22, 2006, at 11:13 AM, Robert Jenkins wrote:
 
  Oh well..
 
  I already had localnet set:
 
  localnet = 192.168.0.0  ; Internal NETWORK address
  localmask = 255.255.255.0   ; Internal netmask
 
  All the involved PCs  Sipura boxes are using 192.168.0.x addresses.
 
  The Sipura boxes work, but the fact that asterisk is sending the 
  external IP to any device on the local network seems to me to be a 
  bug..
 
 
 You didn't mention whether you were also forwarding ports 
 1-2 to the SIP Proxy (ie asterisk).  Thats where the 
 actual RTP (voice
 data) is passing.  Also you need to be sure that there aren't 
 multiple clients on your lan all trying to use the same ports 
 for signaling (ie 5060), as this will fail.
 
 Hope this helps.
 Marty
 

The simple thing is that if I have 'externip' set, I can see on a soft phone
(running on a PC on the same local subnet as asterisk) that it's seeing a
call from another local device as coming from [EMAIL PROTECTED] - which is
the external IP and as everything is inside the firewall there is no audio
from the soft phone when the call answered.

If I comment out the 'externip' line  restart asterisk, the soft phone then
correctly sees the local call as being from [EMAIL PROTECTED] and I get
two-way speech.


Re. multiple clients using port 5060, I have seen comments both ways..
This is how I have it at present and it works (without externip, which
appears to be down to asterisk sending the wrong info  nothing to do with
ports).
As has been said elsewhere, if online VoIP services with thousands of
connections work with a single port, why should there be a problem smaller
numbers of clients?

Robert Jenkins.

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[asterisk-users] Asterisk Dial Plan to Play Message

[broadbandvoice]

Anyone know how to use dial plan to play messages as soon as a phone is picked up. Like when a user picks up a phone, get a message to contact administrator instead of a dial tone?

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Re: [asterisk-users] NAT and externip problem or configuration problem

[Martin Joseph]

On Jul 22, 2006, at 12:54 PM, Robert Jenkins wrote:
snip


On Jul 22, 2006, at 11:13 AM, Robert Jenkins wrote:


Oh well..

I already had localnet set:

localnet = 192.168.0.0  ; Internal NETWORK address
localmask = 255.255.255.0   ; Internal netmask

All the involved PCs  Sipura boxes are using 192.168.0.x addresses.

The Sipura boxes work, but the fact that asterisk is sending the
external IP to any device on the local network seems to me to be a
bug..



You didn't mention whether you were also forwarding ports
1-2 to the SIP Proxy (ie asterisk).  Thats where the
actual RTP (voice
data) is passing.  Also you need to be sure that there aren't
multiple clients on your lan all trying to use the same ports
for signaling (ie 5060), as this will fail.

Hope this helps.
Marty



The simple thing is that if I have 'externip' set, I can see on a  
soft phone
(running on a PC on the same local subnet as asterisk) that it's  
seeing a
call from another local device as coming from [EMAIL PROTECTED] -  
which is
the external IP and as everything is inside the firewall there is  
no audio

from the soft phone when the call answered.

If I comment out the 'externip' line  restart asterisk, the soft  
phone then
correctly sees the local call as being from [EMAIL PROTECTED] and I  
get

two-way speech.


Re. multiple clients using port 5060, I have seen comments both ways..
This is how I have it at present and it works (without externip, which
appears to be down to asterisk sending the wrong info  nothing to  
do with

ports).
As has been said elsewhere, if online VoIP services with thousands of
connections work with a single port, why should there be a problem  
smaller

numbers of clients?


They are exposed as a single IP address.  A single port 5060 is fine  
for your asterisk box.  BUT if you expect calls from  the outside of  
your LAN to pass to SIP phones on the inside of your LAN, you need to  
do one of two things. 1) Use separate ports for the softphones  so  
the NAT isn't confused, or 2) make sure canreinvite is set to no in  
your extensions for the softphone.


If you don't do one of those two things, then what will happen is  
that the caller from outside will connect to the softphone inside,  
and then attempt to talk directly to the softphone.  BUT since your  
router is forwarding all port 5060 traffic to your asterisk box you  
are no longer talking to each other.


You don't mention whether your test calls are coming from inside your  
lan or outside?  You aren't by chance running on a softphone on the  
asterisk box directly?


Marty

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Re: [asterisk-users] Re: Associate manager events to a previous Originate action

[Richard Lyman]

Tony Mountifield wrote:


*snipped

Comparing with 1.2, I see there were originally two calls to manager_event(),
one for OriginateFailure and another for OriginateSuccess.

They have now been combined into one, with a conditional event name,
which may have given rise to the mistaken impression if just skimming
the diffs cursorily.

Cheers
Tony
 


that is exactly what happened.  sorry for the false alarm.

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Re: [asterisk-users] Unicall, not HOW but WHY

[Tzafrir Cohen]

On Thu, Jul 20, 2006 at 05:29:21PM -0500, Moises Silva wrote:
 On 7/20/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
 On Thu, Jul 20, 2006 at 03:51:25PM -0500, Moises Silva wrote:

  2) Why the Unicall package? Not *how* but *why*.
 
  You are not forced to use Unicall, but AFAIK, libmfcr2 is the only
  MFCR2 open source library and since libmfcr2 was wrote by the same guy
  that wrote Unicall telephony abstraction (Steve Underwood), makes
  sense to use it. However you can use Asterisk Zapata channels with
  MFCR2 support, that way you skip the Unicall thing, however, Zapata
  would still be using libmfcr2, and I think there is even less
  documentation about using Zapata with libmfcr2 than using Unicall with
  libmfcr2.
 
 AFAIK, the MFCR2 support in chan_zap doesn't actually work and was
 Steve Underwood's first shot at the problem. I believe it is to be
 removed from 1.4.
 
 I really dont know, if it works, but digging into the source for other
 purposes, I just looked a lot of code in chan_zap with ifdefs
 regarding MFCR2.

Some reference: http://bugs.debian.org/342139

MFCR2 support has been patched out of Debian's chan_zap for that reason.
chan_unicall is availble in a separate package.

-- 
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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Re: [asterisk-users] X100P clone not working

[brandon kruz]

configs look fine to me
it has to be a module problem im guessing


try rmmod for zaptel and wctfxo or w/e then modprobing again(must be root)

if that does not work(which is does on my system, so i added it into a 
startup conf)



kernel: wcfxo: module not supported by Novell

are you sure that wcfxo is the correct module for that device??

hoping to help
`KruZ~






From: Frank Darner [EMAIL PROTECTED]
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Subject: Re: [asterisk-users] X100P clone not working
Date: Sat, 22 Jul 2006 21:47:03 +0200
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Hi,
i tried w/o ztdummy but nothing changed.

any ideas?


zaptel.conf
--
fxsks=1
loadzone=us
defaultzone=us

zapata.conf

[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
echocancle=yes
transfer=yes
echocanclewhenbridged=yes
channel = 1

# modprobe zaptel  modprobe wcfxo

kernel: zaptel: module not supported by Novell, setting U taint flag.
kernel: Zapata Telephony Interface Registered on major 196
kernel: Zaptel Version: SVN-tag-1.2.6-r1096 Echo Canceller: KB1
kernel: wcfxo: module not supported by Novell, setting U taint flag.
kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - IRQ 
193

kernel: wcfxo: DAA mode is 'FCC'
kernel: Found a Wildcard FXO: Generic Clone
kernel: Registered tone zone 0 (United States / North America)

# lsmod | grep zaptel
zaptel201732  1 wcfxo
crc_ccitt   2176  1 zaptel


# ztcfg -v
Zaptel Configuration
==
0 channels configured.



 dmesg | grep x100p

 also
 im almost positive(correct me if im wrong anyone) you do NOT
 need ztdummy
 i repeat, do NOT need ztdummy with x100p

 then rmmod zaptel  rmmod wctdm  modprobe zaptel  modprobe wctdm
 or wctfxo or w/e

 let me know how this works out
 (post the section for your x100p from zaptel.conf and zapata.conf if you
 could m8)

 Hi,
 
 I have problem to set up an X100P clone card.
 Installation of zaptel was successful.
 Also modprobe of zaptel, ztdummy and wcfxo without problems.
 
 kernel: wcfxo: module not supported by Novell, setting U taint flag.
 kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - 
IRQ

 193
 kernel: wcfxo: DAA mode is 'FCC'
 kernel: Found a Wildcard FXO: Generic Clone
 

[asterisk-users] Operator Console(s)/Shared Call Appearances

[Mr. Jones]

Hi Folks,

We're migrating from a conventional KSU/PBX to Asterisk and I'm trying
to determine the best way to allow our receptionist to answer certain
executives telephone lines.

It seems there are probably two routes, but I'm not sure of the
limitations of each.

1. Shared call appearances. This would seem to be the most similar to
what we currently have where we have stations/DNs for 3 executives on
3 assistants phones. Of course with the existing system we have lots
of programmable buttons.  We're leaning towards the SPA-942s, so I'd
be interested to know how this might work (do we need the 4 line
license, and are we limited to 4 call appearances)?

2. Some form of PC application such as the one at Asternic.org, or
something else. This would seem to have the most flexibility, but may
require the operator to pay too much attention to the window, unless
there's some audible notification.

A couple of other alternatives maybe to create a queue, or possibly go
with a side car type device.

I'm open to any and all input.

Best,

Mr. Jones.
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[asterisk-users] newbbie question

[Pablo L. Arturi]

Hello, I am playing with my new * install, and there are a couple of things
that I don't understand, if someone could point me in the right direction it
will be appreciated.

I am trying to configure a voipstunt.com account to place outgoing calls,
and this is my config.

sip.conf:

[voipstunt]
type=friend ; (or peer if we don't need incoming calls, or if there is a
separate section with type=user)
host=sip.voipstunt.com
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
username=voipstuntuser_replacement
fromuser=voipstuntuser_replacement
secret=hiddenpassword
qualify=1000 ; optional
canreinvite=no ; new SIP servers don't like reINVITEs
dtmfmode=inband ; only inband currently works, and not that well

extensions.conf:

[internal]
exten = 787793,1,Dial(SIP/john)
exten = 700099,1,Dial(SIP/maribel)
exten = 10,1,Dial(SIP/dieguez)
exten = 11,1,Dial(SIP/chparson)
exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])


As stated in asteriskTFOT _NXXX will match 541152184829 which is the
phone number of my place, which I am trying to place a call.

I am asuming that the sign + in (SIP/[EMAIL PROTECTED]) will be appended to
what I press in my softphone.

All I get when I call to 541152184829 is:

-- Executing Dial(SIP/john-0819a010, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
Jul 22 20:15:45 NOTICE[1396]: chan_sip.c:1997 auto_congest: Auto-congesting
SIP/voipstunt-0819f520
-- SIP/voipstunt-0819f520 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/john-0819a010' status is 'CONGESTION'


Any idea? suggestion?

Thanks in advance for any comment/help.

Pablo

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Re: [asterisk-users] X100P clone not working

[brandon kruz]

here is some more information
http://www.voip-info.org/wiki/view/X100P+clone



From: Frank Darner [EMAIL PROTECTED]
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Subject: Re: [asterisk-users] X100P clone not working
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Hi,
i tried w/o ztdummy but nothing changed.

any ideas?


zaptel.conf
--
fxsks=1
loadzone=us
defaultzone=us

zapata.conf

[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
echocancle=yes
transfer=yes
echocanclewhenbridged=yes
channel = 1

# modprobe zaptel  modprobe wcfxo

kernel: zaptel: module not supported by Novell, setting U taint flag.
kernel: Zapata Telephony Interface Registered on major 196
kernel: Zaptel Version: SVN-tag-1.2.6-r1096 Echo Canceller: KB1
kernel: wcfxo: module not supported by Novell, setting U taint flag.
kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - IRQ 
193

kernel: wcfxo: DAA mode is 'FCC'
kernel: Found a Wildcard FXO: Generic Clone
kernel: Registered tone zone 0 (United States / North America)

# lsmod | grep zaptel
zaptel201732  1 wcfxo
crc_ccitt   2176  1 zaptel


# ztcfg -v
Zaptel Configuration
==
0 channels configured.



 dmesg | grep x100p

 also
 im almost positive(correct me if im wrong anyone) you do NOT
 need ztdummy
 i repeat, do NOT need ztdummy with x100p

 then rmmod zaptel  rmmod wctdm  modprobe zaptel  modprobe wctdm
 or wctfxo or w/e

 let me know how this works out
 (post the section for your x100p from zaptel.conf and zapata.conf if you
 could m8)

 Hi,
 
 I have problem to set up an X100P clone card.
 Installation of zaptel was successful.
 Also modprobe of zaptel, ztdummy and wcfxo without problems.
 
 kernel: wcfxo: module not supported by Novell, setting U taint flag.
 kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - 
IRQ

 193
 kernel: wcfxo: DAA mode is 'FCC'
 kernel: Found a Wildcard FXO: Generic Clone
 kernel: Registered tone zone 0 (United States / North America)
 
 
 But if I do #ztcfg -vvv no channels will be found.
 
 the log file reports me: kernel: Registered tone zone 0 (United States 
/

 North
 America)
 
 Iam quite sure that zaptel.conf is correct.
 fxsks=1
 loadzone=us
 

RE: [asterisk-users] newbbie question

[brandon kruz]

hey pablo
i havent messed aorund with stun that much except as a repeator
but maybe this is your problem ;]


exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])
should be
exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])
notice the $ sign in front of {EXTEN} that declares it a variable(haha like 
my oxymoron?)

tells asteirsk to lookup the extensions dialed by the user




Hello, I am playing with my new * install, and there are a couple of things
that I don't understand, if someone could point me in the right direction 
it

will be appreciated.

I am trying to configure a voipstunt.com account to place outgoing calls,
and this is my config.

sip.conf:

[voipstunt]
type=friend ; (or peer if we don't need incoming calls, or if there is a
separate section with type=user)
host=sip.voipstunt.com
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
username=voipstuntuser_replacement
fromuser=voipstuntuser_replacement
secret=hiddenpassword
qualify=1000 ; optional
canreinvite=no ; new SIP servers don't like reINVITEs
dtmfmode=inband ; only inband currently works, and not that well

extensions.conf:

[internal]
exten = 787793,1,Dial(SIP/john)
exten = 700099,1,Dial(SIP/maribel)
exten = 10,1,Dial(SIP/dieguez)
exten = 11,1,Dial(SIP/chparson)
exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])


As stated in asteriskTFOT _NXXX will match 541152184829 which is 
the

phone number of my place, which I am trying to place a call.

I am asuming that the sign + in (SIP/[EMAIL PROTECTED]) will be appended 
to

what I press in my softphone.

All I get when I call to 541152184829 is:

-- Executing Dial(SIP/john-0819a010, SIP/[EMAIL PROTECTED]) in 
new

stack
-- Called [EMAIL PROTECTED]
Jul 22 20:15:45 NOTICE[1396]: chan_sip.c:1997 auto_congest: Auto-congesting
SIP/voipstunt-0819f520
-- SIP/voipstunt-0819f520 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/john-0819a010' status is 'CONGESTION'


Any idea? suggestion?

Thanks in advance for any comment/help.

Pablo

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RE: [asterisk-users] Asterisk Dial Plan to Play Message

[brandon kruz]

youll have to decide what context this goes in
either
[internal]
or [incoming]
but i hope you can figure this out yourself
here is an idea

[internal]
exten = s,1,Answer()
exten = s,n,Playback(pbx-invalid)
exten = s,n,Hangup()

so now when a user from [internal] picks up the phone the pbx-invalid.gsm is 
played from

the asterisk sounds directory
so now you are saying to yourself i want my own voice, to record
check this out

[internal]
exten = s,1,Answer()
exten = s,n,Playback(custom)
exten = s,n,Hangup()

(dont tell anyone this part, or just add it temporarily)

exten = 999,1,Answer()
exten = 999,n,Record(custom.gsm)
exten = 999,n,Wait(1)
exten = 999,n,Playback(custom)

you will hear a beep after u dial 999, start recording
then hangup or i think u can type # to stop and move to the next option



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Subject: [asterisk-users] Asterisk Dial Plan to Play Message
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Anyone know how to use dial plan to play messages as soon as a phone is 
picked up. Like when a user picks up a phone, get a message to contact 
administrator instead of a dial tone?




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Re: [asterisk-users] newbbie question

[Pablo L. Arturi]

 exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])
 should be
 exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])
 notice the $ sign in front of {EXTEN} that declares it a variable(haha
like
 my oxymoron?)
 tells asteirsk to lookup the extensions dialed by the user

Brandon, that whas exactly the problem.

Thank you very much!!! now I am calling :)


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Re: [asterisk-users] Asterisk Dial Plan to Play Message

[Eric \ManxPower\ Wieling]

brandon kruz wrote:

youll have to decide what context this goes in
either
[internal]
or [incoming]
but i hope you can figure this out yourself
here is an idea

[internal]
exten = s,1,Answer()
exten = s,n,Playback(pbx-invalid)
exten = s,n,Hangup()


*sigh*

Playback will BY DEFAULT answer the line.  The only time you need an 
Answer() before a Playback() is if you want a Wait() between them.


Doesn't anyone read the docs for the applications they use?  For a good 
time type: show application playback


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RE: Re: [asterisk-users] setting call-limits

[Alejandro Kauffmann]

 on 1.2.4 and 1.2.7, we have to set the 'type=peer' for call-limits to 
 work effectively.
 
 type=friend doesn't seem to enforce call limits at all.
 
 if you haven't tried type=peer, try that first.

No, this doesn't work. 


I believe you need to setup hints for call-limit to work.
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RE: [asterisk-users] Asterisk Dial Plan to Play Message

[broadbandvoice]

Thanks, I'll try that in a few hours and share the experience.

-- Original message -- From: "brandon kruz" [EMAIL PROTECTED]  youll have to decide what context this goes in  either  [internal]  or [incoming]  but i hope you can figure this out yourself  here is an idea   [internal]  exten = s,1,Answer()  exten = s,n,Playback(pbx-invalid)  exten = s,n,Hangup()   so now when a user from [internal] picks up the phone the pbx-invalid.gsm is  played from  the asterisk sounds directory  so now you are saying to yourself i want my own voice, to record  check this out   [internal]  exten = s,1,Answer()  exten = s,n,Playback(custom)  exten = s,n,Hangup()   (dont tell anyone this part, or just add it temp
 oraril
y)   exten = 999,1,Answer()  exten = 999,n,Record(custom.gsm)  exten = 999,n,Wait(1)  exten = 999,n,Playback(custom)   you will hear a beep after u dial 999, start recording  then hangup or i think u can type # to stop and move to the next optionFrom: [EMAIL PROTECTED]  Reply-To: Asterisk Users Mailing List - Non-Commercial  Discussion  To: asterisk-users@lists.digium.com  Subject: [asterisk-users] Asterisk Dial Plan to Play Message  Date: Sat, 22 Jul 2006 20:00:37 +  MIME-Version: 1.0  Received: from lists.digium.com ([69.16.138.164]) by  bay0-mc12-f15.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat,  22 Jul 2006 13:02:18 -0700  Received: from digium-69-16-138-164.phx1.puregig.net (localhos
 t 
 [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 81F652FC9F5;Sat, 22  Jul 2006 13:00:42 -0700 (MST)  Received: from psmtp.com (exprod8mx22.postini.com [64.18.3.122])by  lists.digium.com (Postfix) with SMTP id DD3DF2FC9BFfor  ;Sat, 22 Jul 2006 13:00:37 -0700 (MST)  Received: from source ([216.148.227.153]) by  exprod8mx22.postini.com([64.18.7.10]) with SMTP; Sat, 22 Jul 2006 13:00:38  PDT  Received: from rmailcenter18.comcast.net ([204.127.197.128])by comcast.net  (rwcrmhc13) with SMTPid 20060722200038m1300da2kae; Sat, 22 Jul 2006  20:00:38 +  Received: from [208.17.34.25] by rmailcenter18.comcast.net;Sat, 22 Jul 2006  20:00:37 +  X-Message-Info: txF49lGdW41NVVYVTXpoQhqJpeo4y5ph0jUyL25Wnpk=  X-Original-To: asterisk-users@lists.digium.com  &
 gt;Del
ivered-To: asterisk-users@lists.digium.com  X-Mailer: ATT Message Center Version 1 (Apr 11 2006)  X-Authenticated-Sender: bnRhbmRvaEBjb21jYXN0Lm5ldA==  X-pstn-levels: (S:99.9/99.9 FC:95.5390 LC:95.5390 R:95.9108  P:95.9108M:96.8350 C:98.4741 )  X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c  X-pstn-addresses: from <[EMAIL PROTECTED]>[db-null] X-BeenThere:  asterisk-users@lists.digium.com  X-Mailman-Version: 2.1.5  Precedence: list  List-Id: Asterisk Users Mailing List - Non-Commercial  Discussion  List-Unsubscribe:  , [EMAIL PROTECTED]  List-Archive:  List-Post:  List-Help:  List-Subscribe:  , [EMAIL PROTECTED]  Errors-To: [EMAIL PROTECTED]  Return-Path: [EMAIL PROTECTED]  X-OriginalArrivalTime: 22 Jul 2006 20:02:19.0387 (UTC)  FILETIME=[BCC7A4B0:01C6ADC9]Anyone know how to use dial plan to play messages as soon as a phone is  picked up. Like when a user picks up a phone, get a message to contact  administrator instead of a dial tone?___  --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list  To UN
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RE: [asterisk-users] Asterisk Dial Plan to Play Message

[Russell Bryant]

On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote:
 [internal]
 exten = s,1,Answer()
 exten = s,n,Playback(custom)
 exten = s,n,Hangup()

This, by itself, does not solve the problem where you want the message
to be played back when the phone is picked up without any user
intervention.  If you're using zap phones, you can simply set this
option:

immediate=yes

Then, as soon as the phone goes off hook, the call will begin at the 's'
extension in the configured context instead of providing dialtone.

-- 
Russell Bryant
Software Developer
Digium, Inc.

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RE: [asterisk-users] Asterisk Dial Plan to Play Message

[broadbandvoice]

I'm using SIP channel in Sip.conf and hand the calls over to a termination vendor. 

-- Original message -- From: Russell Bryant [EMAIL PROTECTED]  On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote:   [internal]   exten = s,1,Answer()   exten = s,n,Playback(custom)   exten = s,n,Hangup()   This, by itself, does not solve the problem where you want the message  to be played back when the phone is picked up without any user  intervention. If you're using zap phones, you can simply set this  option:   immediate=yes   Then, as soon as the phone goes off hook, the call will begin at the 's'  extension in the configured context instead of providing dialtone.   --  Russell Bryant  Software Developer  Digium, Inc.   
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Re: [asterisk-users] newbbie question

[brandon kruz]

no problem
most of the time
it is these annoying little problems

stay in touch
glad i could be of assistance :D



From: Pablo L. Arturi [EMAIL PROTECTED]
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 exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])
 should be
 exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])
 notice the $ sign in front of {EXTEN} that declares it a variable(haha
like
 my oxymoron?)
 tells asteirsk to lookup the extensions dialed by the user

Brandon, that whas exactly the problem.

Thank you very much!!! now I am calling :)


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Re: [asterisk-users] Asterisk Dial Plan to Play Message

[brandon kruz]

it will either way work will it not
just because there is more way to do something doesnt mean it wrongs sir

bette ot have answer
then to not have answer on applications that will no
its a good practice




From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
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Subject: Re: [asterisk-users] Asterisk Dial Plan to Play Message
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brandon kruz wrote:

youll have to decide what context this goes in
either
[internal]
or [incoming]
but i hope you can figure this out yourself
here is an idea

[internal]
exten = s,1,Answer()
exten = s,n,Playback(pbx-invalid)
exten = s,n,Hangup()


*sigh*

Playback will BY DEFAULT answer the line.  The only time you need an 
Answer() before a Playback() is if you want a Wait() between them.


Doesn't anyone read the docs for the applications they use?  For a good 
time type: show application playback


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RE: [asterisk-users] Asterisk Dial Plan to Play Message

[brandon kruz]

thank you russel
forgot to mention this.



From: Russell Bryant [EMAIL PROTECTED]
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Subject: RE: [asterisk-users] Asterisk Dial Plan to Play Message
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On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote:
 [internal]
 exten = s,1,Answer()
 exten = s,n,Playback(custom)
 exten = s,n,Hangup()

This, by itself, does not solve the problem where you want the message
to be played back when the phone is picked up without any user
intervention.  If you're using zap phones, you can simply set this
option:

immediate=yes

Then, as soon as the phone goes off hook, the call will begin at the 's'
extension in the configured context instead of providing dialtone.

--
Russell Bryant
Software Developer
Digium, Inc.

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RE: [asterisk-users] SNOM missed call.

[Christian Stredicke]

If the * server sends the following header in the CANCEL request, then
then snom phone does not count the call as missed:

Reason: SIP;cause=200;text=Call completed elsewhere

See http://www.ietf.org/rfc/rfc3326.txt. Maybe someone can post an
example on how to insert this header.

Christian

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Franklin Webb
 Sent: Friday, July 21, 2006 4:01 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] SNOM missed call.
 
 Hi Thomas,
 
 That setting is controlled by line.  Maybe you could setup 
 two seperate lines on the phones and direct the two different 
 call types accordingly.
 
 Franklin Webb
 Assistant IT Project Leader
 Inter Medi@ Marketing Solutions
 610-701-9670
 [EMAIL PROTECTED]
 
 - Original Message -
 From: Thomas Laurids Pedersen [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Thursday, July 20, 2006 2:19:14 PM GMT-0500
 Subject: [asterisk-users] SNOM missed call.
 
 Hi All,
 
 Using AAH 2.8.
 
 I have configured a group to handle a common number for a 
 remote office.
 All phones in the office is in the group and they are ringing 
 with a seperate ringtone. All this is very well.
 
 However all phones other than the one how answered the call 
 is recording a missed call. I know this is an option in the 
 SNOM phone, but is there some way to avoid this for this type 
 of calls ? or is there another way of doing this ?
 
 Best regards
 
 Thomas Laurids Pedersen
 
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Re: [asterisk-users] X100P clone not working

[Tzafrir Cohen]

On Sat, Jul 22, 2006 at 09:02:48PM +0200, Frank Darner wrote:
 Hi,
 
 I have problem to set up an X100P clone card.
 Installation of zaptel was successful.
 Also modprobe of zaptel, ztdummy and wcfxo without problems.
 
 kernel: wcfxo: module not supported by Novell, setting U taint flag.
 kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - IRQ 193
 kernel: wcfxo: DAA mode is 'FCC'
 kernel: Found a Wildcard FXO: Generic Clone
 kernel: Registered tone zone 0 (United States / North America)
 
 
 But if I do #ztcfg -vvv no channels will be found.

What is the exact output from that command?

What is the output from 'cat /proc/zaptel/*'

-- 
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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