Re: [asterisk-users] Asterisk only registering at one provider

[Dennis Guse]

Hey guys,

thanks for your effort.
I just replicated the typical problem: the problem sits in front of the
monitor.
(And I am so glad that my systems are not in production).

The issue was that "register =>" is *only* allowed in the [general]-section.
But since I like a precise and clean configuration I put it like this:

[general]
register => SIP1
[SIP1]
...

register => SIP2
[SIP2]
...

And then the second register is ignored as it is not in [general].
However, no error messages are thrown...

Best regards and a happy weekend!

---
Dennis Guse

---
Dennis Guse

On Wed, Mar 18, 2015 at 10:19 PM, Joshua Colp  wrote:

> Dennis Guse wrote:
>
>> Hey,
>>
>> I am running default Asterisk 11.16.0 on a FreeBSD-Machine.
>> I need to register to several other SIP-Services (actually 3):
>>
>> short sip.conf
>>
>> register => XX@a
>> register => XX@b
>> register => XX@c
>>
>> If I remember correctly this worked quite well, but I now checked the
>> system again and it is only obeying the first register statement.
>> "sip show registry" only reports the first entry and if I reorder them,
>> this effect stays the same.
>>
>> Did something changed recently in the parsing code for sip.conf or so?
>>
>
> Nope, and I'd expect we'd be seeing many bug reports if something like
> this was occurring.
>
> I just did this in my general section in 11:
> register => meh@tacos
> register => hola@bob
> register => yolo@dave
>
> And confirmed they appeared as expected:
> Hostdnsmgr Username   Refresh
> StateReg.Time
> dave:5060   N  yolo   120
> Request Sent
> bob:5060N  hola   120
> Request Sent
> tacos:5060  N  meh120
> Request Sent
>
> Does anything show up on the console when chan_sip is loaded?
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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[asterisk-users] Asterisk only registering at one provider

[Dennis Guse]

Hey,

I am running default Asterisk 11.16.0 on a FreeBSD-Machine.
I need to register to several other SIP-Services (actually 3):

short sip.conf

register => XX@a
register => XX@b
register => XX@c

If I remember correctly this worked quite well, but I now checked the
system again and it is only obeying the first register statement.
"sip show registry" only reports the first entry and if I reorder them,
this effect stays the same.

Did something changed recently in the parsing code for sip.conf or so?

---
Dennis Guse
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Re: [asterisk-users] Echo Cancellation on VoIP networks

[Dennis Guse]

On VoIP echo cancellation is basically: hope that the client is doing AND
is doing it well.
In the best case each client uses a knowledge about his hardware
(microphone, speaker, distance etc.).



---
Dennis Guse


On Tue, Aug 26, 2014 at 1:30 PM, Emiliano Vazquez  wrote:

> El 26/08/14 a las 05:33, Grant Bagdasarian escibió:
>
>  I’m new to Echo Cancellation and I was wondering how it is handled/works
>> on pure VoIP networks using Asterisk?
>>
> there is no echo problems on pure VoIP networks.
>
> echo is a common problem when you have changes from analog to digital.
>
> The only echo problem you will have is when you call another network who
> has analog circuits with wrong configuration or poor hardware. But you
> can't solve it.
>
> Best regards.
>
>
>
> --
> Emiliano Vazquez | PcCentro Informatica & CCTV
> Office: +54 (11) 4635-3218 y Rotativas
> Movil: 011-15-6253-7165
> Mail: emilianovazq...@gmail.com
> Web: http://www.pccentro.com.ar
>
>
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Re: [asterisk-users] Strange Error

[Dennis Guse]

Sound like chan_sip was not build.
Just a guess: check that openssl-dev is available


---
Dennis Guse


On Thu, Jul 3, 2014 at 12:01 PM, Andrew Colin  wrote:

> Hi Guys,
>
>
>
> Does anyone know what this error means and how to fix it?
>
>
>
> [Jul  3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/
>
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Re: [asterisk-users] How to execute an AGI script for each call.

[Dennis Guse]

You could try either the predial-handler or the dial-macro M.


---
Dennis Guse


On Wed, Jul 2, 2014 at 3:06 PM, Joshua Colp  wrote:

> Anurag Rana wrote:
>
>> Hi All,
>>
>
> Kia ora,
>
>
>  I am trying to execute some AGI script no matter what extension is called.
>> There is 'h' extension to call AGI script when any call hangs up no
>> matter what extension hangup.
>>
>> for example ->
>>
>> [some-context]
>>
>> /// something here which call AGI script no matter what extension
>> receive call.
>>
>> exten => 111,1,Dial(SIP/111)
>> exten => 112,1,Dial(SIP/112)
>>
>> exten => h,1,AGI(pt.py)   ;; executes no matter what extension hang up
>>
>
> Have your first priority be a pattern match of something like _X. which
> executes your AGI. Then have your second priority be the specialized logic
> (such as the Dial above). That should do what you want.
>
> Cheers,
>
> --
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> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] Using macros in extensions.lua?

[Dennis Guse]

Got it:

extensions = {
["macro-test"] = {
["s"] = function(c, e)
 app.verbose("This is my macro")
end;
};

default = {

["_X"] = function(c, e)
app.dial("SIP/00", nil, "mM(test)")
 end;
};
};


---
Dennis Guse


On Fri, Jun 6, 2014 at 6:49 PM, George Joseph 
wrote:

> On Fri, Jun 6, 2014 at 1:48 AM, Dennis Guse <
> dennis.g...@alumni.tu-berlin.de> wrote:
>
>> Hi,
>>
>> I have defined a dialplan in lua and now would like to use "dial" with
>> the macro M to implement some logic, when the callee-channel gets created.
>>
>> Working old style would be (extensions.conf)
>>
>> [default]
>> exten => _X,1,dial(SIP/1,,M(mymacro^parameter))
>>
>> [macro-mymacro]
>> exten => s,1,verbose(${ARG1})
>>
>> How to implement the same functionality using pbx_lua?
>>
>> Details: Asterisk 11.7 on Ubuntu 14.04
>>
>> Kind regards
>>
>> Dennis Guse
>>
>> Here's how I do it for pre-dial handlers...
>
> extensions.handlers = {
>   ["addheader"] = function(c,e)
>   channel.PJSIP_HEADER('add', "Alert-Info"):set(";info=custom1")
>   end;
> }
>
> extensions.local_default = {
>   [""] = function(c,e)
>   app.dial('PJSIP/'..e,nil,'b(handlers^addheader^1)')
>   end;
> }
>
>
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[asterisk-users] Using macros in extensions.lua?

[Dennis Guse]

Hi,

I have defined a dialplan in lua and now would like to use "dial" with the
macro M to implement some logic, when the callee-channel gets created.

Working old style would be (extensions.conf)

[default]
exten => _X,1,dial(SIP/1,,M(mymacro^parameter))

[macro-mymacro]
exten => s,1,verbose(${ARG1})

How to implement the same functionality using pbx_lua?

Details: Asterisk 11.7 on Ubuntu 14.04

Kind regards

Dennis Guse

Quality and Usability Lab
Telekom Innovation Laboratories
TU Berlin
Ernst-Reuter-Platz 7
D-10587 Berlin, Germany
Tel: +49 30 8353 58874
Fax: +49 30 8353 58409
E-mail: dennis.g...@telekom.de
Web: www.qu.tlabs.tu-berlin.de
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Re: [asterisk-users] Indications.conf: change volume

[Dennis Guse]

I let Asterisk generate the ringtone: DIAL(SIP/XX, 'r')...



---
Dennis Guse


On Tue, Apr 22, 2014 at 4:47 PM, jg  wrote:

>  The call invitation is only signaled in most cases. You need to check
> the settings of your phones.
>
> Hi,
>
>  I use Asterisk to create the dial tone (indications.conf), which works
> quite well. However the generated signal is quite loud at the client side
> (in comparison to the following speech ).
>
>  Is there an option to modify the volume?
>  ---
> Dennis Guse
>
>
>
>
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[asterisk-users] Indications.conf: change volume

[Dennis Guse]

Hi,

I use Asterisk to create the dial tone (indications.conf), which works
quite well. However the generated signal is quite loud at the client side
(in comparison to the following speech ).

Is there an option to modify the volume?
---
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Re: [asterisk-users] Upgrade from 1.0.x to AsteriskNOW 3.0

[Dennis Dryden]

AsteriskNOW 1 and 2 are both based on Centos 5 and I think the new
AsteriskNOW 3 is based on Centos 6 so upgrades are not supported by the
Linux OS distribution[1]. It's best to backup and reinstall with the new
version. It's a shame AsteriskNOW is not based on Debian so it could be
dist-upgraded between versions.

Cheers,
Dennis


[1] http://wiki.centos.org/HowTos/MigrationGuide/MigratingFiveToSix


On Mon, May 13, 2013 at 10:27 PM, Andre Goree  wrote:

> Hello all.  I was hoping someone out there might have some advice or
> suggestions regarding an upgrade from an archaic Asterisk version.
>
> I've been given the daunting task of upgrading a very old Asterisk-1.0.x
> install to a recent LTS version.  I'll also need the install to have
> high-availability and failover support.
>
> From my research, it would appear that AsteriskNOW-3.0 might be my best
> bet, as it seems to be running Asterisk-11.  I've previously installed
> Asterisk-11+FreePBX in a VM, and this appears to be very similar.  Is there
> any upside to using AsteriskNOW vs. Asterisk+FreePBX? Other than the
> obvious fact that everything is nicely placed on an iso for ease of
> installation?
>
> As for the actual upgrade, is it possible to step through each of the
> UPGRADE*.txt files under the Asterisk-11 source?  I.e, UPGRADE-1.2.txt ->
> UPGRADE-1.4.txt -> UPGRADE-1.6.txt -> UPGRADE-1.8.txt -> UPGRADE.txt? Or
> would it be prudent to recreate my current 1.0.x configuration under
> Asterisk-11 instead?
>
> Regarding HA/failover, is a hardware solution (such as Digium's R800/R850)
> my only option?  During my research I've found scripts on the internet that
> allow for failover using arp/nmap, however those appear to only work for
> hardware failures.  I would need something that can account for both
> hardware and software failures.
>
> Thanks in advance for any advice that anyone can give on the subject.  Any
> suggestions, etc. would help immensely!
>
> --
> Andre Goree
> -=-=-=-=-=-
> Email - an...@drenet.net
> Website   - http://blog.drenet.net
> PGP key   - 
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> -=-=-=-=-=-
>
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Re: [asterisk-users] Digium IP Phones

[Dennis Dryden]

Hello,
I've been using the Digium D40's for a few weeks now and I think they
are good for the price. There are a few UI problems but I hope/expect
they will be resolved in a firmware update or two.

Haven't looked at the SDK yet.

Thanks,
Dennis



On Thu, May 10, 2012 at 2:38 AM, Danny Dias  wrote:
> Hello,
>
> Im looking to buy a digium phone D70 unit just for testing on lab; to really
> understand the phone and features.
>
> I cant find any website with opinions; any here? Are they really valuable to
> the price? (D70 quite expensive)
>
> Does the SDK for building apps is usable? Can you build powerfull apps?
> Examples?
>
> Many thanks
>
>
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[asterisk-users] Digium D40 Direction map 'X' key not functioning

[Dennis Dryden]

Hello,
Is there a way to make the the 'X' key next to the direction map close
the menu system or do anything? It's kind of annoying to have to press
the "Line 1" key.

http://www1.digium.com/sites/default/files/support/d40_phoneusersheet.pdf

Thanks,
Dennis

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Re: [asterisk-users] Multi-tenant with receptionist features for managed service

[Sean Dennis]

For MT check out Thirdlane's MT PBX:

http://www.thirdlane.com/products/thirdlane-pbx-mte

I use the PBX Manager which it's based on and it works very well.
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Re: [asterisk-users] DTMF does not work

[Sean Dennis]

On Mon, Dec 29, 2008 at 1:55 PM, Brent Vrieze wrote:

> I got no resonses to this and some funny bounces so I'm trying again.
>
>
>
> First of all Merry Christmas.
>
> Second, my first problem with my provider not staying registered with
> our server was my fault.  We moved our server room and I restarted the
> test system and the production system causing them to ping-pong back and
> forth registering with our provider causing random problems, they are
> both set to register with the same account right now.  I shut Asterisk
> down on the one and now we don't drop any longer.  doh!!!
>
> Last, We are having DTMF problems with our provider (via:talk).  Does
> anyone have any experience with them and if so can you share it?
> via:talk does have a sample sip.conf and extensions.conf file to use but
> the dial plan they set up does not require any DTMF so they may never
> have tested it.  We have tried inband, auto, rfc2833 for our DTMF and
> nothing works.  I have submitted a ticket with them but the last time I
> did that they never responded so that is why I am posting here.
> I signed up with another SIP provider for a test account and the DTMF
> passes no problem from them so I must conclude there is some setting
> that via:talk has that is causing the problem.  via:talk will not
> confirm this but they must be using Asterisk as all the menus and such
> they have feel very Asteriskish.  Is there something I can tell via:talk
> to try on their end to make this work?
>
> As a side symptem every time our system registers with via:talk it seams
> to jump from server to server on their end.  They must have some sort of
> load balancing going on that is causing that.  In the past we could get
> the DTMF to pass when we were on the initial server we registered with
> but when we got pushed to another server the DTMF would fail till I did
> a sip reload or restarted Astersk.  Now we get no DTMF ever.
>
> System set up.
> Asterisk 1.4.22
> Asterisk GUI 2.0
>
> users.conf
> [trunk_1]
> context = DID_trunk_1
> host = galvatron.vtnoc.net
> username = user name
> secret = password
> trunkname = via:talk - galvatron  ; GUI metadata
> hasiax = no
> registeriax = no
> hassip = yes
> registersip = yes
> trunkstyle = voip
> hasexten = no
> fromuser = user name
> authuser = user name
> insecure = port,invite
> dtmf = rfc2833
> dtmfmode = rfc2833
> relaxdtmf = yes
> rfc2833compensate = yes
> port = 5060
> canreinvite = no
> fromdomain = galvatron.vtnoc.net
> disallow = all
> allow = ulaw,gsm
>
> If you need to see more of the setup info I can provide.
>
> Thanks
>   Brent
>



I have the same problems with Viatalk.  The problem is with their "new"
servers.  You are pointed to galvatron.vtnoc.net which is one of those.  I
currently have mine working by using their "old" servers.  Try calling
support, changing your account to rfc2833 if you haven't already and then
point to chicago-1e.vtnoc.net with your same settings .  You will have DTMF
working, but I am not sure when the "old" servers are going away.

Good Luck,

Sean
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Re: [asterisk-users] Convert CallerID name to uppercase

[Sean Dennis]

>
> In TRUNK versions of Asterisk, there is a function called TOUPPER,
> which converts strings to upper case.  I don't know when, exactly, it
> appeared but I expect if it's not in the version you're using it may
> be portable backwards without too much difficulty if the version
> you're using supports functions.
>
> JT
>
> *CLI> core show function TOUPPER
>

This looks like exactly what I need.  I see that it's available in 1.6
so I will upgrade and let you know how it goes.

Thank You.

-Sean Dennis

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[asterisk-users] Convert CallerID name to uppercase

[Sean Dennis]

Our legacy PBX will not accept the callerID name in anything but
capital letters. (Harris 20-20)  When I send a call to the legacy PBX
from asterisk I would like to have asterisk convert the callerID name
to uppercase letters.  Is there a way to do this?


Thanks for any input.

-Sean

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Re: [asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP

[Nathan Dennis]

Thanks for pointing that out Tony, Should have included that in my first
post.
Below is the version and the IAX config for each end

Server 1
Version : 1.4.18

IAX2.conf peer details

[brisbane]
type=friend
host=XXX.XXX.XXX.XXX
trunk=yes
context=internal
context=parkinglot
qualify=1
username=XXX
secret=
disallow=all
allow=g729

Server 2 
Version : 1.4.21.2

IAX2.conf peer details

[cairns]
type=friend
host=XXX.XXX.XXX.XXX
trunk=yes
context=internal
context=callagents
context=parkinglot
qualify=1
username=XXX
secret=
disallow=all
allow=g729



Nathan Dennis 
__ 
Integrated Solutions (QLD)P/LPhone: +61 (7) 4044 0300 
Direct: +61 (7) 4044
0302
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Offices and agents in Cairns - Brisbane - Melbourne -- Adelaide --
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Wednesday, 24 September 2008 8:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IAX Hangup floods link with repeated
VNAKand HANGUP

In article
<[EMAIL PROTECTED]
.au>,
Nathan Dennis <[EMAIL PROTECTED]> wrote:
> We have been using asterisk for a while now but have recently needed 
> to install a second server in a remote office and set up a iax trunk 
> between the 2 servers. The dial plan seems to work well when I tested 
> it on the same LAN. However this afternoon I connected the system at 
> the remote office and made some calls. All the calls connect and work 
> fine, voice quality is great no really couldn't have hoped for better.

> Hang up the call and tried to make another call and nothing, the link 
> was not responding, after much trouble shooting I have found that 
> after the call is hung up the 2 asterisk servers seem to go into some 
> kind of loop sending each other message. I have pasted a debug for 
> both servers below that include everything from the start of the call 
> to after hangup. I have cut them short at the VNAK and Hangup cycle 
> just continues for 30seconds or so flooding the link completely.
>  
> Any help you may be able to provide would be greatly appreciated

I can't help with your problem, sorry, but anyone who can help will need
to know exactly what version of Asterisk you have at each end.

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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[asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP

[Nathan Dennis]

:
ACK
   Timestamp: 09855ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
-- Hungup 'IAX2/brisbane-16384'
  == Spawn extension (iax2brisbaneout, 5510, 1) exited non-zero on
'SIP/1406-b7b2b530'
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1406-b7b2b530",
"IAX2/brisbane/[EMAIL PROTECTED]") in new stack
-- Called brisbane/[EMAIL PROTECTED]
-- Hungup 'IAX2/brisbane-16385'
Tx-Frame Retry[000] -- OSeqno: 005 ISeqno: 007 Type: IAX Subclass:
HANGUP
   Timestamp: 16939ms  SCall: 16384  DCall: 06840 [10.10.51.22:4569]
   CAUSE CODE  : 16
 
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
   Timestamp: 3ms  SCall: 16385  DCall: 0 [10.10.51.22:4569]
   VERSION : 2
   CALLED NUMBER   : h
   CODEC_PREFS : (g729)
   CALLING NUMBER  : 1406
   CALLING PRESNTN : 0
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   CALLING NAME: Nathan Dennis
   LANGUAGE: en
   CALLED CONTEXT  : internal
   USERNAME: cairns
   FORMAT  : 256
   CAPABILITY  : 57600
   ADSICPE : 2
   DATE TIME   : 2008-09-24  18:35:44
 
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
HANGUP
   Timestamp: 6ms  SCall: 16385  DCall: 0 [10.10.51.22:4569]
   CAUSE CODE  : 0
 
  == Spawn extension (iax2brisbaneout, h, 1) exited non-zero on
'SIP/1406-b7b2b530'
 Extension Changed 1406[internalhints] new state Idle for Notify User
1401 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1419 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1415 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1402 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1404 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1411 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1408 (queued)
-- Incoming call: Got SIP response 500 "Internal Server Error" back
from 10.10.11.193
Rx-Frame Retry[ No] -- OSeqno: 007 ISeqno: 006 Type: IAX Subclass:
ACK
   Timestamp: 16939ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
   Timestamp: 3ms  SCall: 08602  DCall: 16385 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
   Timestamp: 00011ms  SCall: 08602  DCall: 16385 [10.10.51.22:4569]
   AUTHMETHODS : 3
   CHALLENGE   : 842177371
   USERNAME: cairns
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass:
AUTHREP
   Timestamp: 00116ms  SCall: 16385  DCall: 08602 [10.10.51.22:4569]
   MD5 RESULT  : b9f5616e47f5f8e5605868b717d510aa
 
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
INVAL
   Timestamp: 0ms  SCall: 0  DCall: 16385 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
VNAK
   Timestamp: 00118ms  SCall: 08602  DCall: 16385 [10.10.51.22:4569]
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
HANGUP
   Timestamp: 6ms  SCall: 16385  DCall: 0 [10.10.51.22:4569]
   CAUSE CODE  : 0
 
Tx-Frame Retry[001] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass:
AUTHREP
   Timestamp: 00116ms  SCall: 16385  DCall: 08602 [10.10.51.22:4569]
   MD5 RESULT  : b9f5616e47f5f8e5605868b717d510aa
 
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
INVAL
   Timestamp: 0ms  SCall: 0  DCall: 16385 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
VNAK
   Timestamp: 00232ms  SCall: 08602  DCall: 16385 [10.10.51.22:4569]
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
HANGUP
   Timestamp: 6ms  SCall: 16385  DCall: 0 [10.10.51.22:4569]
   CAUSE CODE  : 0

 
 
IAX Debug on second Server
 
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass:
ANSWER
   Timestamp: 00091ms  SCall: 06840  DCall: 16384 [10.10.11.22:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: VOICE   Subclass:
136
   Timestamp: 00100ms  SCall: 06840  DCall: 16384 [10.10.11.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
ACK
   Timestamp: 00088ms  SCall: 16384  DCall: 06840 [10.10.11.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass:
ACK
   Timestamp: 00091ms  SCall: 16384  DCall: 06840 [10.10.11.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass:
ACK
   Timestamp: 00100ms  SCall: 16384  DCall: 06840 [10.10.11.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 004 Type: VOICE   Subclass:
136
   Timestamp: 00288ms  SCall: 16384  DCall: 06840 [10.10.11.22:4569]
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass:
ACK
   Timestamp: 00288ms  SCall: 06840  DCall: 16384 [10.

Re: [asterisk-users] Unable to run make menuselect for asterisk-addons

[Sean Dennis]

Jonn R Taylor wrote:
>
> Hi all,
>
>  
>
> I am unable to run make menuselect for asterisk-addons. Works fine for 
> zaptel and asterisk. Here is the output.
>
>  
>
> Jonn
>
>  
>
> [EMAIL PROTECTED] asterisk-addons]# make menuselect
>
> CC="gcc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect 
> CONFIGURE_SILENT="--silent" makeopts
>
> \make[1]: Entering directory `/usr/src/asterisk-addons/menuselect'
>
> Package gtk+-2.0 was not found in the pkg-config search path.
>
> Perhaps you should add the directory containing `gtk+-2.0.pc'
>
> to the PKG_CONFIG_PATH environment variable
>
> No package 'gtk+-2.0' found
>
> make[1]: Leaving directory `/usr/src/asterisk-addons/menuselect'
>
> make[1]: Entering directory `/usr/src/asterisk-addons/menuselect'
>
> make[1]: `makeopts' is up to date.
>
> make[1]: Leaving directory `/usr/src/asterisk-addons/menuselect'
>
> CC="gcc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect 
> CONFIGURE_SILENT="--silent"
>
> make[1]: Entering directory `/usr/src/asterisk-addons/menuselect'
>
> gcc -g -c -D_GNU_SOURCE -Wall   -c -o menuselect.o menuselect.c
>
> gcc -g -c -D_GNU_SOURCE -Wall   -c -o strcompat.o strcompat.c
>
> gcc -g -c -D_GNU_SOURCE -Wall-c -o menuselect_curses.o 
> menuselect_curses.c
>
> make[2]: Entering directory `/usr/src/asterisk-addons/menuselect/mxml'
>
> gcc -O -Wall   -c mxml-attr.c
>
> gcc -O -Wall   -c mxml-entity.c
>
> gcc -O -Wall   -c mxml-file.c
>
> gcc -O -Wall   -c mxml-index.c
>
> gcc -O -Wall   -c mxml-node.c
>
> gcc -O -Wall   -c mxml-search.c
>
> gcc -O -Wall   -c mxml-set.c
>
> gcc -O -Wall   -c mxml-private.c
>
> gcc -O -Wall   -c mxml-string.c
>
> /bin/rm -f libmxml.a
>
> /usr/bin/ar crvs libmxml.a mxml-attr.o mxml-entity.o mxml-file.o 
> mxml-index.o mxml-node.o mxml-search.o mxml-set.o mxml-private.o 
> mxml-string.o
>
> a - mxml-attr.o
>
> a - mxml-entity.o
>
> a - mxml-file.o
>
> a - mxml-index.o
>
> a - mxml-node.o
>
> a - mxml-search.o
>
> a - mxml-set.o
>
> a - mxml-private.o
>
> a - mxml-string.o
>
> ranlib libmxml.a
>
> make[2]: Leaving directory `/usr/src/asterisk-addons/menuselect/mxml'
>
> gcc -o cmenuselect menuselect.o strcompat.o menuselect_curses.o 
> mxml/libmxml.a -lncurses
>
> gcc -g -c -D_GNU_SOURCE -Wall   -c -o menuselect_stub.o menuselect_stub.c
>
> gcc -o menuselect menuselect.o strcompat.o menuselect_stub.o 
> mxml/libmxml.a
>
> make[1]: Leaving directory `/usr/src/asterisk-addons/menuselect'
>
> Generating input for menuselect ...
>
> **
>
> *** Install ncurses to use the menu interface! ***
>
> **
>
> menuselect changes NOT saved!
>
> [EMAIL PROTECTED] asterisk-addons]# rpm -qa | grep ncurses
>
> ncurses-5.5-24.20060715
>
> ncurses-devel-5.5-24.20060715
>
> [EMAIL PROTECTED] asterisk-addons]# rpm -qa | grep gtk
>
> gtk2-2.10.4-20.el5
>
> [EMAIL PROTECTED] asterisk-addons]#
>
> 
>
> ___
>   

I had the same problem with asterisk 1.4.18.  I switched to 1.4.21 and 
it worked great.




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Re: [asterisk-users] BLF functionality

[Sean Dennis]

>
> I find it amazing how often I find myself stuck on a problem and then 
> someone else posts a question about it to the list. I am in the same 
> boat with the OP (although I never thought to test incoming calls until 
> I read his message). If I call a phone it will show busy, however if I 
> make a call from that phone it still shows as idle. I've set call-limit 
> and limitonpeers and restarted asterisk but still no joy. What am I 
> missing? I'm running 1.4.21.2
>
> Relevant sip.conf:
>
> [lan-soundpointip](!)
> type=friend
> host=dynamic
> disallow=all
> allow=ulaw
> dtmfmode=rfc2833
> qualify=no
> call-limit=10
> limitonpeers=yes
>
> [3900](lan-soundpointip)
> username=3900
> secret=sdjghdfkjhgdf
> context=phone-operator
> callerid=Operator <3900>
>
> [3917](lan-soundpointip)
> username=3917
> secret=dfkghdjfhdkfd
> context=phone-isdept
> callerid=Dave Fullerton <3917>
> mailbox=3117
>
>   
In my general section of my sip.conf I have:

allowsubscribe=yes
notifyringing=yes
limitonpeer=yes
notifyhold=yes

and it works both ways.


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Re: [asterisk-users] Asterisk end-user GUI?

[Sean Dennis]

Check out www.thirdlane.com they have a excellent end user portal. 

Ken D'Ambrosio wrote:
> I badly want to roll out Asterisk at my job.  Unfortunately, my boss is
> dazzled by shiny objects.  We had a vendor in today who showed us their
> system which, honestly, didn't suck -- but boy, is it going to be
> expensive!  One major component of the eye candy was an end-user interface
> that allowed the user to initiate calls to a contact list, check for
> presence, create conferences, etc.  Is there anything like that, aimed at
> end-users (as opposed to admins) for Asterisk?  I'd even be willing to go
> with proprietary; I just don't want a wholly-proprietary, hobbled,
> licensed-to-Heck-and-back system, which is where it looks like my boss is
> leaning.
>
> Thanks!
>
> -Ken
>
>
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>   


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Re: [asterisk-users] SIP Testing-Tool

[Dennis Brandenburg]

> 17 jul 2008 kl. 12.08 skrev Dennis Brandenburg:
>
>   
>> Hi All,
>>
>> Does anyone know, if there is a tool, which is doing the follwing:
>>
>> - Testprogram on host A establishes a sip connection to testprogramm  
>> on
>> host B
>> - Testprogram on host A plays a tone and Testprogram B verifies, if  
>> tone
>> is playing correctly (without any interruptions)
>> 
>
> I did this test with a Spirent Test platform.
>
> /O
>   
Thank you,

but isn't there a free solution?

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[asterisk-users] SIP Testing-Tool

[Dennis Brandenburg]

Hi All,

Does anyone know, if there is a tool, which is doing the follwing:

- Testprogram on host A establishes a sip connection to testprogramm on 
host B
- Testprogram on host A plays a tone and Testprogram B verifies, if tone 
is playing correctly (without any interruptions)

Thank You.

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Re: [asterisk-users] Toll Free International Number

[Sean Dennis]

Try www.tollfreeforwarding.com, they do just that. 


Larry Costigan wrote:
> Hello All,
>  
> I am looking to find a way to provide international toll free access 
> to our Knoxville, TN (USA) office from our customers in the UK and in 
> Australia, and when I talked with AT&T I was surprised to find out how 
> expensive they are...  Surely, other businesses are not paying this 
> much - are they?!?!  
>  
> Can someone in this good group please help me with some advice as to 
> who can provide affordable and reliable international toll free 
> service for a better price than AT&T?
>  
> Thanks in advance,
> Larry Costigan
> Food Donation Connection
> (Asterisk fan and ABE user)
> 
>
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Re: [asterisk-users] forwarding pots lines

[Dennis P. Clark]

Will fax and dial-up internet work through the gateway?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Carroll
Sent: Friday, May 23, 2008 8:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List -Non-Commercial Discussion
Subject: Re: [asterisk-users] forwarding pots lines

There are a couple of companies out there that make 24 port fxo and fxs
boxes. If you have some unused  fibers you cout do this very reliably
with two channel banks...  One with fxs ports and the other with fxo
ports and t1 media converters.

 The grand stream solution mentioned in an earlier post does 8 ports,
you could get one 4 port model and one 8 port model of fxs and the same
of fxo and  accomplish your goal rather inexpensively as well.

Joe

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

Sent: 5/23/08 8:43 AM
Subject: Re: [asterisk-users] forwarding pots lines


Sorry to jump in on this but I am also interested in this topic.

In my scenario I have about 10 POTs lines brought into the front of a
facility and the only infrastructure connecting the back of the facility
is a 3000ft fiber backhaul.  I've been asked to bring the POTs lines to
the back of the facility.

Are there any ATAs that trunk multiple POTs Lines?  Like a multiplexer
of some sort.

If anyone has any information can you please provide the manufacturer
and model of the device?

Thank You,
Dennis

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of randulo
Sent: Friday, May 23, 2008 4:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] forwarding pots lines

On Fri, May 23, 2008 at 10:04 AM, Eric Fort <[EMAIL PROTECTED]> wrote:
> will an ata directly connect to another remote ata thus emulating a
long
> phone cord?  also most of the ATA's I've seen drive a phone rather
than
> accepting a line from the telco.

Depending on the reliability needed (is this a way to talk to a
girlfriend in another country or a mission-critical business use?) I'd
say it's better to pay a small monthly fee to someone like OnSIP.com
and use their centrex.

If it's because you have the phone lines already installed and need to
just use them at certain times, I do think there are FXO devices but
I'm not sure they will help. You wouldn't need two asterisk servers at
any rate but only one. The phones connect (through a router if need
be) to the asterisk at the phone lines + FXO end.

/r

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Re: [asterisk-users] forwarding pots lines

[Dennis P. Clark]

Sorry to jump in on this but I am also interested in this topic.

In my scenario I have about 10 POTs lines brought into the front of a
facility and the only infrastructure connecting the back of the facility
is a 3000ft fiber backhaul.  I've been asked to bring the POTs lines to
the back of the facility.  

Are there any ATAs that trunk multiple POTs Lines?  Like a multiplexer
of some sort.  

If anyone has any information can you please provide the manufacturer
and model of the device?

Thank You,
Dennis

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of randulo
Sent: Friday, May 23, 2008 4:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] forwarding pots lines

On Fri, May 23, 2008 at 10:04 AM, Eric Fort <[EMAIL PROTECTED]> wrote:
> will an ata directly connect to another remote ata thus emulating a
long
> phone cord?  also most of the ATA's I've seen drive a phone rather
than
> accepting a line from the telco.

Depending on the reliability needed (is this a way to talk to a
girlfriend in another country or a mission-critical business use?) I'd
say it's better to pay a small monthly fee to someone like OnSIP.com
and use their centrex.

If it's because you have the phone lines already installed and need to
just use them at certain times, I do think there are FXO devices but
I'm not sure they will help. You wouldn't need two asterisk servers at
any rate but only one. The phones connect (through a router if need
be) to the asterisk at the phone lines + FXO end.

/r

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Re: [asterisk-users] Problem with cisco 7970G [EMAIL PROTECTED]

[Sean Dennis]

Jorge Munoz wrote:
>
> Hi everyone
>
> This is the first time I post something here so I’m sorry about my 
> English, I don’t know how to write properly.
>
> Well, I’ve been working with Cisco 7960 telephones and my boss bought 
> new ones , 7970G with SIP70.8-2-2SR3S firmware version, those work 
> perfectly, but one of them has the SIP70.8.3.5S version, and this one 
> doesn’t connect to the server , I wanted to install the 
> SIP70.8.2.2SR3S version, but I couldn’t, is there anyone who knows how 
> to do it?
>
> Many thanks.
>
> 
>
>   

When I updated to SIP70.8.3.5S on my 7970 I had to change 
1 to  in the XML file 
to make the phone register. I believe it is a bug in the new firmware.


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Re: [asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...

[Sean Dennis]

Steve Hickel wrote:
> I have sip set up on Callmanager 4.x. When others call my ext of 2016 on
> ccm after a busy or no answer, asterisk voice mail answers by saying,
> "Mailbox  password." I want it to put them into my mailbox so they
> can leave a message. Somehow I must be missing something... Please
> help! 
>
> I have spent 19 hours easy on trying to figure this one out. 
>
> SIP DN is  on CCM 
> VOICEMAIL on Asterisk is . 
>
> Here is my sip.conf: 
>
> [general] 
> context=default 
> allowoverlap=no 
> bindport=5060 
> bindaddr=0.0.0.0 
> srvlookup=yes 
> allowexternaldomains=yes 
> allowexternalinvites=no 
> allowguest=yes 
> allowsubscribe=no 
> allowtransfer=yes 
> alwaysauthreject=no 
> autodomain=no 
> callevents=no 
> compactheaders=no 
> dumphistory=no 
> g726nonstandard=no 
> ignoreregexpire=no 
> jbenable=no 
> jbforce=no 
> jblog=no 
> maxcallbitrate=384 
> maxexpiry=3600 
> minexpiry=60 
> nat=no 
> notifyringing=no 
> pedantic=no 
> promiscredir=no 
> recordhistory=no 
> relaxdtmf=no 
> rtcachefriends=no 
> rtsavesysname=no 
> rtupdate=no 
> sendrpid=yes 
> sipdebug=no 
> t1min=100 
> t38pt_udptl=no 
> [authentication] 
>
> [sip] 
> type=friend 
> context=incoming 
> host=172.20.1.57 
> ipaddr=172.20.1.57 
> allow=ulaw 
> allow=alaw 
> nat=no 
> canreinvite=yes 
> qualify=yes 
>
> Here is my voicemail.conf 
>
> [zonemessages] 
> eastern=America/New_York|'vm-received' Q 'digits/at' IMp 
> central=America/Chicago|'vm-received' Q 'digits/at' IMp 
> central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours' 
> military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' 
> european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM 
> [other] 
>
> [general] 
> format=wav49|gsm|wav 
> serveremail=asterisk 
> attach=yes 
> skipms=3000 
> maxsilence=10 
> silencethreshold=128 
> maxlogins=3 
> emaildateformat=%A, %B %d, %Y at %r 
> sendvoicemail=yes 
> attachfmt=wav 
> deletevoicemail=no 
> envelope=no 
> maxgreet=60 
> maxmessage=120 
> maxmsg=100 
> minmessage=1 
> operator=yes 
> review=yes 
> saycid=no 
> sayduration=yes 
> mailcmd=/usr/sbin/sendmail -t 
> externotify=/var/libasterisk/scripts/vm.sh 
> [default] 
> 2016=1234,Steve,[EMAIL PROTECTED] 
>
> Here is the relevant parts of my extensions.conf: 
>
> [macro-dialout-callmanager] 
> exten=s,1,ChanIsAvail(SIP/sip) 
> exten=s,2,Cut(AVAILCHAN=AVAILCHAN,,1) 
> exten=s,3,Dial(${AVAILCHAN}/${ARG1}) 
> exten=s,4,Hangup 
> exten=s,102,Congestion 
> [incoming] 
> exten=,1,GotoIf($[${RDNIS}]?2:400) 
> exten=,2,MailboxExists([EMAIL PROTECTED] 
> exten=,3,Congestion 
> exten=,103,Voicemail(su${RDNIS}) 
> exten=,104,Playback(vm-goodbye) 
> exten=,105,Hangup 
> exten=,400,VoicemailMain 
> [general] 
> static=yes 
> writeprotect=no 
> clearglobalvars=no 
> autofallthrough=yes 
> priorityjumping=no 
> [default] 
> exten=_230,1,SetCallerID(${EXTEN:3}) 
> exten=_230,2,Dial(SIP/[EMAIL PROTECTED]) 
> exten=_230,3,Answer 
> exten=_230,4,Wait,1 
> exten=_230,5,Hangup 
> exten=_231,1,SetCallerID(${EXTEN:3}) 
> exten=_231,2,Dial(SIP/[EMAIL PROTECTED]) 
> exten=_231,3,Answer 
> exten=_231,4,Wait,1 
> exten=_231,5,Hangup 
>
> I am using users.conf, but don't know how that ties in or whether I even
> need it...??? 
>
> thanks, 
>
> Steve
>
>
>
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>   

You didn't mention what version of asterisk, but if you are using 
version 1.4.x, in extensions.conf you need to use:

CALLERID(rdnis) instead of just RDNIS


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Re: [asterisk-users] Switch recommendation?

[Sean Dennis]

Hilary Miller wrote:
> This will be my first major asterisk experiment and I'm trying to
> choose a PoE switch for 15-24 phones. I was going to spend $400 on
> this:
>
> http://www.newegg.com/product/product.asp?item=N82E16833124053
>
> but then I see this on ebay:
>
> http://cgi.ebay.com/WS-C3524-PWR-XL-EN-Cisco-3524-24-FE-Switch-W-PoE-VoIP_W0QQitemZ370043264927QQihZ024QQcategoryZ51268QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
>
> and I'm thinking, hey, thats a lot cheaper and it is PoE. Will the
> Cisco IP phone's proprietary wizardry be a problem for my flock on
> Linksys IP phones? Because as long as it can do vlan qos and poe I
> think I can scrape by for half the price, right?
>
> Thanks for reading!
>   
The Cisco 3524 switch doesn't support 802.3af which is what your Linksys 
phones are going to want.  If you have just Cisco phones this would 
work.  To have 802.3af you have to have at least a Cisco 3560 series switch.

See:
http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00801189b5.shtml#powerover
for reference




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Re: [asterisk-users] SPA-962+ SPA-932- blf function

[Sean Dennis]

John Meksavan wrote:
> Asterisk Users,
>
>   I am running Asterisk 1.4.11 on Debian "Etch" system with the TDM03B 
> wildcard.  I recently purchased a SPA-962 and SPA-932- the sidecar for 
> our receptionist.  After reading many forum postings on how to 
> configure the side car,  I uprgraded the SPA-962 software to 
> 5.1.18(SC) version. 
>
>I got the sidecar to subscribed to an extension on the Asterisk 
> server, but the LED state on the SPA-932 never changes even when I am 
> a call with that extension on another VOIP phone- SPA-941.   I got the 
> speed dial function to work, but the "blf" function does not appear to 
> work. 
>
>   Did anybody get the "blf" function to work?  What I am doing wrong?  
> Any input would be greatly appreciated.  Thanks in advance. 
>
> Regards,
> John
> 
> How well do you know your celebrity gossip? Talk celebrity smackdowns 
> here. 
> 
>
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To make it work properly I had to add the following to sip.conf:
allowsubscribe=yes
notifyringing=yes
limitonpeer=yes
notifyhold=yes

See if that helps.

-Sean



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Re: [asterisk-users] IAXy device

[Sean Dennis]

bilal ghayyad wrote:
> Hi All;
>
> I have been chocked just when I saw some posts talking
> about how much the IAXy is bad :) - 
>
> So I would like to ask, did any one try it later and
> wether it is good or not? I am asking this because I
> need to use it as it is NAT Transparent (as I read
> also, and I did not try it to see how much it is
> transparent).
>
> What about codec? Why it is only support g711 and does
> not support compressed codec? And what about the IP
> address and the DNS usage and the DDNS usage?
>
> What main porblems contain and any advise?
>
> Regards
> Bilal
>
>
>   
> 
>   
The device has no echo cancellation and sounds horrible (lots of echo) 
on about half of the analog phones I tried it on.  I wouldn't recommend 
it unless you absolutely need IAX. It's also very expensive for a 1 port 
ATA.


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Re: [asterisk-users] does the meetme module still require anexternal timing source?

[Dennis Christopher]

Mike and Steve,

Thanks. Someone had already suggested AppConference to me. Any  
opinions on that?

Dennis
On 12-Mar-08, at 4:42 PM, Mike Fedyk wrote:

> Agreed, Callweaver and Freeswitch are both better for conferencing
> (especially if you don't have zaptel hardware).
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Steve  
> Totaro
> Sent: Wednesday, March 12, 2008 1:28 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] does the meetme module still require
> anexternal timing source?
>
>
> Try Callweaver.
>
> Thanks,
> Steve Totaro
>
> On Wed, Mar 12, 2008 at 4:12 PM, Dennis Christopher
> <[EMAIL PROTECTED]> wrote:
>> Thanks Matt,
>>
>>  However I am looking to see if Asterisk with meetme is viable on OS
>> X, and I believe that  ztdummy will not compile on that platform. If
>> so, I would need an  alternative to meetme to do conferencing...?
>>
>>  Dennis
>>
>>
>> On 12-Mar-08, at 4:02 PM, Matt Riddell wrote:
>>
>>> -BEGIN PGP SIGNED MESSAGE-
>>> Hash: SHA1
>>>
>>> Dennis Christopher wrote:
>>>> All,
>>>>
>>>> Can anyone confirm if the meetme module still requires an external
>>>> timing source, such as a card and or driver?  >
>>> Correct, but insofar as a driver, you can just use ztdummy, which  
>>> will
>>> be loaded by default when starting up zaptel if you have no hardware
>>> installed.
>>>
>>> - --
>>> Kind Regards,
>>>
>>> Matt Riddell
>>> Director
>>> ___
>>>
>>> http://www.venturevoip.com (Great new VoIP end to end solution)
>>> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
>>> http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News -  
>>> rss)
>>> -BEGIN PGP SIGNATURE-
>>> Version: GnuPG v1.4.7 (MingW32)
>>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>>>
>>> iD8DBQFH2DbpDQNt8rg0Kp4RArouAKCF0D36feiSxokdOx8UzF2gGOhonACgou4K
>>> WIAhdj/PUrOx5Z4N0fePRqM=
>>> =xfLA
>>> -END PGP SIGNATURE-
>>>
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>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
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>
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>
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Re: [asterisk-users] does the meetme module still require an external timing source?

[Dennis Christopher]

Thanks Matt,

However I am looking to see if Asterisk with meetme is viable on OS  
X, and I believe that
ztdummy will not compile on that platform. If so, I would need an  
alternative to meetme to do conferencing...?

Dennis
On 12-Mar-08, at 4:02 PM, Matt Riddell wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Dennis Christopher wrote:
>> All,
>>
>> Can anyone confirm if the meetme module still requires an external
>> timing source, such as a card and or driver?
>
> Correct, but insofar as a driver, you can just use ztdummy, which will
> be loaded by default when starting up zaptel if you have no hardware
> installed.
>
> - --
> Kind Regards,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com (Great new VoIP end to end solution)
> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
> http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.7 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFH2DbpDQNt8rg0Kp4RArouAKCF0D36feiSxokdOx8UzF2gGOhonACgou4K
> WIAhdj/PUrOx5Z4N0fePRqM=
> =xfLA
> -END PGP SIGNATURE-
>
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[asterisk-users] does the meetme module still require an external timing source?

[Dennis Christopher]

All,

Can anyone confirm if the meetme module still requires an external  
timing source, such as a card and or driver?

Dennis Christopher
Pixion Inc.

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Re: [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information

[Sean Dennis]

Sigma Networks wrote:
> I would like to get in contact with users/consultants who are or have 
> worked with the Cisco phones and Asterisk to trade information.  
>
> Cisco has reluctantly made SIP available on their phones and most of the 
> information on voip-info and other wiki's appears to be reverse 
> engineered.  There is a wealth of information out there which is 
> terrific.  
>
> I have a client with about 40 phones composed of 7970, 7960 and 7906 
> phones.   I've upgraded all of these to SIP 8-3-3SR2S and the basic 
> functions are working.
>
> My current questions are:
>
>1. How to remotely reboot 7970s.   I have both web access and SSH
>   access to the phones.  The instructions I have for SSH are to use
>   (1) user/pass (or whatever is in the confg) and then (2)
>   debug/debug.  Surprisingly  "reset" is not a valid command to
>   restart the phone.  There doesn't appear to be a reset on the web
>   page, maybe there's a hidden URL?
>2. BusyLampField? 
>
> Thanks in advance.
>
>
>
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>   
We have about 200 79x1's running SIP w/ asterisk and we are very pleased 
despite some of the non-standard things Cisco does. 
In answer to question 1 the only way we have found to reboot the phone 
remotely is shutdown the port on the POE switch.  This will drop the 
PC's network as well if it is plugged into the phone. 
Question 2 I would like to know the answer to myself.  I would be 
curious to know if it works with the SIP image in call manager.



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Re: [asterisk-users] OT: Is Cisco 7960 SIP firmware same as 7940 SIP firmware?

[Sean Dennis]

Mike Dent wrote:
> Hi,
> just wondered if it was the same firmware on both devices?
> thanks
> Mike
>
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>   
Yes


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Re: [asterisk-users] What web GUI are people happy with?

[Sean Dennis]

Anciso, Roy wrote:
>
> Just wondering what web GUI people like for asterisk.  I installed 
> asterisk from source and I was looking at possibly installing web GUI 
> for system management.  So far freepbx.org looks promising anybody 
> else have any suggestions.
>
> Thanks
>
>  
>
> **Roy Anciso**
>
> Director of Technology
>
> Manistee Intermediate School District
>
> 1710 Merkey Road
>
> Manistee, MI 49660
>
> Ph: 231-723-4264
>
> Fx: 231-723-1690
>
> [EMAIL PROTECTED]
>
>  
>
> 
>
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I recommend Thirdlanes PBX manager.  We have several installations of it 
and it seems to work very well.  The best thing about it is the end user 
portal.  I believe there is a demo at thirdlane.com



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[asterisk-users] Echo Problems with IAXy

[Sean Dennis]

 From what I have found the IAXy doesn't handle echo very well.  About 
half of the analog phones I try on the adapter create an echo on the far 
end.  The person I am talking to can hear themselves.  I am using 
Asterisk 1.4 and have tried it with 1.2 as well with the same results.  
Is there is anything I can do in Asterisk to help solve the echo problem? 

Thanks,

Sean Dennis


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Re: [asterisk-users] Xorcom Bri and asterisk crashes

[Nathan Dennis]

Thanks for the input, but I still don't seem to have any luck with the
devices locking up. I've even rebuilt a new system on new hardware and a
new xorcom device but still no good. Once the device locks up that's it
the only way to get zaptel and asterisk back up is to turn them off and
restart the server. The command you have me
 
rmmod xpp_usb; /usr/share/zaptel/xpp_fxloader reset
Works great and will reload the firmware as long as the devices are
frozen. Once they lock up this command will not reload the firmware and
brings up the following errors.

'xpp_fxloader'[16065]: Resetting FPGA Firmware on /dev/bus/usb/005/016
'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (247): bulk_write
failed: error reaping URB: No such device
'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (810): Renumeration to
default failed: errno=-19
'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (203): Releasing
interface: usb: could not release intf 0: No such device
'xpp_fxloader'[16342]: fpga_load failed remoivng with status 237


I'm running Fedora 7, Kernel 2.6.21-1.3194.fc7

Will hopefully be upgrading the kernel tonight if I can get some
downtime to do so.

As for more traces, I can do that, but being reasonably new to this I
will need some help getting them for you.



 
On Thu, Jul 05, 2007 at 09:14:12AM +1000, Nathan Dennis wrote:
> We have recently install an asterisk solution with about 60 physical
> extensions. While the system is running it runs reasonably well (Still
> have a few teething problems) but twice now they have experienced a
> degradation in voice quality and dropped calls and then finally
asterisk
> completely crashes out. Restarting asterisk will work for a little
while
> and it will crash again, each time less time will pass before a crash
> out. The first time I didn't have much logging so I didn't get
anything
> to work with. I have since turned on debugging and following is the
logs
> from the time of the last crash. Can anyone point out where the
problem
> may lay, suggested updates or changes?
>  
>  
> Jul  4 11:56:51 DEBUG[20042] chan_sip.c: Auto destroying call
> 'aca7e8d7fc914018 at 192.168.12.164
<http://lists.digium.com/mailman/listinfo/asterisk-users> '
> Jul  4 11:56:54 DEBUG[20042] chan_sip.c: Stopping retransmission on
> '6eeb52b53a414a6975facbc22ca10686 at 192.168.10.12
<http://lists.digium.com/mailman/listinfo/asterisk-users> ' of Request
102: Match
> Found
> Jul  4 11:56:55 VERBOSE[20050] logger.c: -- Accepting voice call
> from '' to '40312688' on channel 0/2, span 5
> Jul  4 11:56:55 DEBUG[20050] chan_zap.c: Enabled echo cancellation on
> channel 14
> Jul  4 11:56:55 DEBUG[20295] pbx.c: Function result is 'CID withheld'
> Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
> Set("Zap/14-1", "CALLERID(name)=Old Main Line:CID withheld") in new
> stack
> Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
> Goto("Zap/14-1", "mainq|q|1") in new stack
> Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,1)
> Jul  4 11:56:55 DEBUG[20295] pbx.c: Function result is 'false'
> Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
> Set("Zap/14-1", "NightMode=false") in new stack
> Jul  4 11:56:55 DEBUG[20295] pbx.c: Expression result is '0'
> Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
> GotoIf("Zap/14-1", "0?afterhoursq|q|1") in new stack
> Jul  4 11:56:55 DEBUG[20295] pbx.c: Not taking any branch
> Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
> GotoIfTime("Zap/14-1", "8:00-17:30|mon-fri|*|*|?businesshours") in new
> stack
> Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,5)
> Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
> Set("Zap/14-1", "__ALERT_INFO=<http://www.example.com
<http://www.example.com/> >;info=MainQ") in
> new stack
> Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
> Queue("Zap/14-1", "mainq1|twr|||10") in new stack
> Jul  4 11:56:55 DEBUG[20295] chan_zap.c: Requested indication 3 on
> channel Zap/14-1
> Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Called
> Local/700 at callagents
<http://lists.digium.com/mailman/listinfo/asterisk-users> 
> Jul  4 11:56:55 VERBOSE[20298] logger.c: -- Executing
> Set("Local/700 at callagents-bc5a
<http://lists.digium.com/mailman/listinfo/asterisk-users> ,2",
"Extension=700") in new stack
> Jul  4 11:56:55 VERBOSE[20298] logger.c: -- Executing
> Set("Local/700 at callagents-bc5a
<http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Need Advice/Suggestion

[Nathan Dennis]

Hi Farooq,
  I've done just that for one of our customers. All I did was
add an exten such as *56 that set a custom database value to
nightmode=true. Then as calls come in I just check the database value to
see if it is set to true or not. Note I have asterisk patched with
Bristuff so unless you do as well the hint section will not work.


See Below

exten => *56,hint,DS/56
exten => *56,1,Set(NightMode=${DB(nightmode/active)})
exten => *56,n,playback(service)
exten => *56,n,Gotoif($["${NightMode}" = "true"]?turnoff)
exten => *56,n,Set(DB(nightmode/active)=true)
exten => *56,n,devstate(56,2)
exten => *56,n,playback(activated)
exten => *56,n,hangup()
exten => *56,n(turnoff),Set(DB(nightmode/active)=false)
exten => *56,n,playback(de-activated)
exten => *56,n,devstate(56,0)
exten => *56,n,hangup

Then as a call comes in you just check the value in the database

exten => q,1,Set(NightMode=${DB(nightmode/active)})
exten => q,n,Gotoif($["${NightMode}" = "true"]?afterhoursq,q,1)
exten => q,n,GotoIfTime(8:00-17:30|mon-fri|*|*|?businesshours)


Nathan Dennis 
__ 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Farooq
Ahmed
Sent: Tuesday, 3 July 2007 5:00 PM
To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Subject: [asterisk-users] Need Advice/Suggestion

Hi all,
As we know we can configure in astersik like before 5:00pm calls go to
reception and after 5:00 pm calls go to some mobile no. One of my client
requested that he wants to manually shift the dial plan  like above as
he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I
can not give him freepbx  access.
Any idea or solution.
Regards
Farooq
-- 

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[asterisk-users] Xorcom Bri and asterisk crashes

[Nathan Dennis]

 structure of
Local/[EMAIL PROTECTED],1
Jul  4 11:56:59 DEBUG[20295] channel.c: Got clone lock for masquerade on
'SIP/700-09530a90' at 0x952ab64
Jul  4 11:56:59 DEBUG[20298] chan_local.c: Not posting to queue since
already masked on 'Local/[EMAIL PROTECTED],2'
Jul  4 11:56:59 DEBUG[20295] channel.c: Putting channel SIP/700-09530a90
in 64/64 formats
Jul  4 11:56:59 DEBUG[20295] channel.c: Released clone lock on
'Local/[EMAIL PROTECTED],1'
Jul  4 11:56:59 DEBUG[20295] channel.c: Done Masquerading
SIP/700-09530a90 (6)
 
The last entry was just before the crash. 
 
We also have this in dmesg (Not sure if its related)
 
NOTICE-xpd_bri: XBUS-01/XPD-10: D-Chan RX Bad checksum: [FE:28=FC] (252)
NOTICE-xpd_bri: XBUS-01/XPD-10: Multibyte Drop: errno=-71
BUG: warning at kernel/softirq.c:138/local_bh_enable() (Not tainted)
 [] local_bh_enable+0x45/0x92
 [] cond_resched_softirq+0x2c/0x42
 [] release_sock+0x4f/0x9d
 [] tcp_sendmsg+0x90b/0x9f9
 [] release_sock+0x12/0x9d
 [] tcp_recvmsg+0x8d2/0x9de
 [] core_sys_select+0x218/0x2f3
 [] inet_sendmsg+0x3b/0x45
 [] sock_aio_write+0xf6/0x102
 [] get_page_from_freelist+0x23b/0x2a5
 [] do_sync_write+0xc7/0x10a
 [] autoremove_wake_function+0x0/0x35
 [] vfs_write+0xbc/0x154
 [] sys_write+0x41/0x67
 [] syscall_call+0x7/0xb
 ===
NOTICE-xpd_bri: XBUS-01/XPD-08: D-Chan RX Bad checksum: [FA:FA=FC] (252)
NOTICE-xpd_bri: XBUS-01/XPD-08: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [78:3A=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [55:55=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [55:55=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [55:55=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [AA:AA=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-10: D-Chan RX Bad checksum: [35:22=FC] (252)
NOTICE-xpd_bri: XBUS-00/XPD-10: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-10: D-Chan RX short frame (idx=3)
NOTICE-xpd_bri: XBUS-00/XPD-10: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-10: D-Chan RX Bad checksum: [35:22=FC] (252)
NOTICE-xpd_bri: XBUS-00/XPD-10: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-10: D-Chan RX Bad checksum: [22:22=FC] (252)
NOTICE-xpd_bri: XBUS-00/XPD-10: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-10: D-Chan RX Bad checksum: [55:02=FC] (252)
NOTICE-xpd_bri: XBUS-00/XPD-10: Multibyte Drop: errno=-71
zaptel Disabled echo canceller because of tone (rx) on channel 4
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [94:2D=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71

 
Once the system becomes unstable the only way to get it up again is to
shutdown (not restart) pull the power and USB on the Xorcom Bri 4
devices. Then plug them back in and start the system up.
If the power and USB is not disconnected a the devices may look like
they are working fine but zaptel will not start stating that it can not
find span ** what ever it happens to fail on.
 
 
We are running the following versions
Asterisk - Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g
Zaptel - zaptel-1.2.18
XPP - version:trunk-r3965
 srcversion: 723B8A27A7E9750BB039D00

If you need any more information please let me know.
 
 
 

Nathan Dennis 
__ 
Integrated Solutions (QLD)P/LPhone: +61 (7) 4044 0300 
Direct: +61 (7) 4044
0302
124 Spence Street   Fax:+61 (7) 4041 6600
CAIRNS QLD 4868Mobile: 0418 608609

Australia 

E-mail: [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> 
Web Site: www.i-solutions.net.au <http://www.i-solutions.net.au/> 

Offices and agents in Cairns - Brisbane - Melbourne -- Adelaide --
Sydney
__ 
The information transmitted is intended only for the person or entity to
which 
it is addressed and may contain confidential and/or privileged material.

Any review, retransmission, dissemination or other use of, or taking of
any 
action in reliance upon, this information by persons or entities other
than the 
intended recipient is prohibited. If you received this in error, please
contact 
the sender and delete the material from any computer.
 
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[asterisk-users] Problems with SIP Registration on VPN Link

[Nathan Dennis]

Hi,
We are having major problems with a remote site that links to the
head office via a VPN tunnel. The phones will register fine and work for
a few minutes to hours but then will drop their connection and will no
register to asterisk even with a restart of the phone. We have 2 other
remote sites that work exactly same and they are not having any issues
so i believe it has to be be something to do with the network rather
then asterisk but this is the sip debug for a phone trying to register.
Any idea where i should start to look as this has me totally confused as
obviously the phones can communicate with asterisk at all times just
something is causing the registration to get screwed up.
 
Jul  4 09:43:46 NOTICE[7320]: chan_sip.c:6599 check_auth: stale nonce
received from ''
Transmitting (NAT) to 192.168.12.63:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63
From: "Edmonton Boardroom 1"
;tag=65cbed22c3593805
To: ;tag=as4d6893cc
Call-ID: [EMAIL PROTECTED]
CSeq: 10005 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="48f69f92", stale=true
Content-Length: 0
 

---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000
ms
cnsmavs1*CLI>
<-- SIP read from 192.168.12.63:5060:
REGISTER sip:192.168.10.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15
From: "Edmonton Boardroom 1"
;tag=65cbed22c3593805
To: 
Contact: 
Supported: path
Authorization: Digest username="763", realm="asterisk", algorithm=MD5,
uri="sip:192.168.10.12", nonce="587da437",
response="4bd29b9213057e3e2f3a5270748fbe85"
all-ID: [EMAIL PROTECTED]
CSeq: 10005 REGISTER
Expires: 3600
User-Agent: Grandstream GXP2000 1.1.2.23
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,M
ESSAGE
Content-Length: 0
 

--- (14 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.12.63 : 5060 (NAT)
Transmitting (NAT) to 192.168.12.63:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63
From: "Edmonton Boardroom 1"
;tag=65cbed22c3593805
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 10005 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Content-Length: 0
 

---
Jul  4 09:43:48 NOTICE[7320]: chan_sip.c:6599 check_auth: stale nonce
received from ''
Transmitting (NAT) to 192.168.12.63:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63
From: "Edmonton Boardroom 1"
;tag=65cbed22c3593805
To: ;tag=as4d6893cc
Call-ID: [EMAIL PROTECTED]
CSeq: 10005 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="750fc224", stale=true
Content-Length: 0
 

 

Nathan Dennis 


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Re: [asterisk-users] Xorcom Bri 4 Port USB

[Nathan Dennis]

Thanks Tzafrir, that did the trick.
But please note the that the bristuff patch from xorcom has broken links in it. 
It can't download asterisk using the URL in the script. Easy enough to fix by 
pointing to a known good URL.
 



From: [EMAIL PROTECTED] on behalf of Tzafrir Cohen
Sent: Mon 25/06/2007 7:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Xorcom Bri 4 Port USB



Hi

On Mon, Jun 25, 2007 at 06:38:37PM +1000, Nathan Dennis wrote:
>
> Hi,
>I'm having some trouble setting up a Xorcom Bri 4 port. I have compiled 
> asterisk and zaptel using the Bristuff bristuff-0.3.0-PRE-1y-g patches.
>
> So I'm running zaptel-1.2.17.1 and asterisk-1.2.18.
>
> The problem I'm having is that for one I get no LEDs showing if the unit
> is in TE and NT mode (not a issue for me but may have some impact on
> things) I have no errors in any logs I can see but once zaptel and
> asterisk are started up I get a lots of warnings in asterisk such as
> the following

What is the output of:

modinfo xpp | grep version

if this is something of the sort of 'r3495' then you indeed have an
older version of the driver where BRI support has not been matuire
enough and specifically leds display was not as it is today. In current
version (e.g: the one in zaptel 1.2.18/1.4.3) you will always see an
orange LED for NT or green led for TE on the port.

Please get the version of bristuff from:

http://updates.xorcom.com/astribank/bristuff/
http://updates.xorcom.com/astribank/bristuff/bristuff-0.3.0-PRE-1y-g-xr1.tar.gz

At least until we see a new version of bristuff.

and also see:

http://updates.xorcom.com/astribank/bristuff/INSTALL.html

Also, for the sake of those who will see the messages in a search:

>
> Jun 25 18:18:07 WARNING[2833]: chan_zap.c:2662 pri_find_dchan: No D-channels 
> available!  Using Primary channel 3 as D-channel anyway!
>   == Primary D-Channel on span 2 down
> Jun 25 18:18:07 WARNING[2834]: chan_zap.c:2662 pri_find_dchan: No D-channels 
> available!  Using Primary channel 6 as D-channel anyway!
>   == Primary D-Channel on span 3 down
> Jun 25 18:18:07 WARNING[2835]: chan_zap.c:2662 pri_find_dchan: No D-channels 
> available!  Using Primary channel 9 as D-channel anyway!
>   == Primary D-Channel on span 1 down
>

This message comes from chan_zap when a span is down. If a span has
pri_{cpe,net} signalling or bri_{cpe,net} signalling (bristuff BRI ptp)
then you'll get those messages for spans that are down. If the
signalling is bri_{cpe,net}_ptmp they'll be debug messages.



> It errors for all for ports and makes no difference if I have the
> ISDN cables connected or not. I want to run in ptp mode and
> currently use a digium B410P card on the connections that work fine
> so I know that the lines work and ptp is the correct mode.
>

> Following are my configs. Any pointers you can give would be greatly 
> appreciated.
>
> We are running Fedora 7.
> Kernel "Linux 2.6.21-1.3194.fc7 #1 SMP i686 i686 i386 GNU/Linux" (Standard 
> Kernel with install)
> Device has jumpers all set to TE mode.
>
>
> /etc/init.d/zaptel.conf
> # Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: BRI_TE"
> span=1,1,1,ccs,ami
> # termtype: te
> bchan=1-2
> dchan=3
>
> # Span 2: XBUS-00/XPD-08 "Xorcom XPD #0/8: BRI_TE"
> span=2,2,1,ccs,ami
> # termtype: te
> bchan=4-5
> dchan=6
>
> # Span 3: XBUS-00/XPD-10 "Xorcom XPD #0/16: BRI_TE"
> span=3,3,1,ccs,ami
> # termtype: te
> bchan=7-8
> dchan=9
>
> # Span 4: XBUS-00/XPD-18 "Xorcom XPD #0/24: BRI_TE" RED
> #span=4,4,1,ccs,ami
> # termtype: te
> #bchan=10-11
> #dchan=12
>
> # Global data
>
> loadzone= au
> defaultzone = au
>
>
> /etc/asterisk/zapata.conf
> [channels]
> ;   echocancel = yes
> ;   transfer = yes
> ;   threewaycalling = yes
>
> #include zapata-channels.conf
>
>
> /etc/asterisk/zapata-channels.conf
>
> ; Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: BRI_TE"
> callerid=asreceived
> group=0
> context=from-pstn
> switchtype = euroisdn
> signalling = bri_cpe
> channel => 1-2
> callerid=
> group=
> context=default
>
> ; Span 2: XBUS-00/XPD-08 "Xorcom XPD #0/8: BRI_TE"
> callerid=asreceived
> group=0
> context=from-pstn
> switchtype = euroisdn
> signalling = bri_cpe
> channel => 4-5
> callerid=
> group=
> context=default
>
> ; Span 3: XBUS-00/XPD-10 "Xorcom XPD #0/16: BRI_TE"
> callerid=asreceived
> group=0
> context=from-pstn
> switchtype = euroisdn
> signalling = bri_cpe
> channel => 7-8
> callerid=
> group=
> context=default
>
> ; Span 4: XBUS-00/XPD-18

[asterisk-users] Zap dialling issues

[Nathan Dennis]

I'm having problems getting an Xorcom USB Bri 4 dialling out in
Australia.
 
I can receive calls into the system without an issue, but I can not for
the life of me dial out of the system. Below are my configs, I'm hoping
its something simple that I just can't see as I've been looking at it
for to long. Can any one point me in the right direction.
 
P.S. Yes it is meant to be in TE PTP mode as the current Digium B410P
works fine in that mode
 
 
 
/etc/asterisk/extensions.conf  Extract
[internal]
include=>features
include=>speeddial
 

;Extention number for main Q
exten => 700,1,Goto(mainq,q,1)
   
;-
;Calling a local extensions mailbox
exten => _*7XX,1,Set(Extension=${EXTEN:1})
exten => _*7XX,n,Goto(directtovoicemail,s,1)
 
;--
 
;Static externaly accessable Conference room with recording
exten =>
599,1,Set(MEETME_RECORDINGFILE=/var/spool/asterisk/monitor/conference-50
0-${EPOCH});
exten => 599,n,MeetMe(500,cMr,4081)
exten => 599,n,Hangup
 
;Dynamic Conference rooms for internal users to transfer callers to
exten => _5XX,1,MeetMe(${EXTEN},cMd)
exten => _5XX,n,Hangup
 
exten => 6000,1,Dial(zap/0418608609)
 
 
/etc/asterisk/zapata.conf
[channels]
;echocancel = yes
;transfer = yes
callgroup=1
pickupgroup=1
 
; Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: BRI_TE" 
group=0,11
context=zapin
switchtype = euroisdn
signalling = bri_cpe
channel => 1-2
 
; Span 2: XBUS-00/XPD-08 "Xorcom XPD #0/8: BRI_TE" 
group=0,12
context=zapin
switchtype = euroisdn
signalling = bri_cpe
channel => 4-5
 

; Span 3: XBUS-00/XPD-10 "Xorcom XPD #0/16: BRI_TE" 
;group=0,13
;context=zapin
;switchtype = euroisdn
;signalling = bri_cpe
;channel => 7-8
 
 
 
; Span 4: XBUS-00/XPD-18 "Xorcom XPD #0/24: BRI_TE" 
;group=0,14
;context=zapin
;switchtype = euroisdn
;signalling = bri_cpe
;channel => 10-11
 
/etc/zaptel.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
 
# It must be in the module loading order
 

# Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: BRI_TE"
span=1,1,1,ccs,ami
# termtype: te
bchan=1-2
dchan=3
 
# Span 2: XBUS-00/XPD-08 "Xorcom XPD #0/8: BRI_TE"
span=2,2,1,ccs,ami
# termtype: te
bchan=4-5
dchan=6
 
# Span 3: XBUS-00/XPD-10 "Xorcom XPD #0/16: BRI_TE"
span=3,3,1,ccs,ami
# termtype: te
bchan=7-8
dchan=9
 
# Span 4: XBUS-00/XPD-18 "Xorcom XPD #0/24: BRI_TE"
span=4,4,1,ccs,ami
# termtype: te
bchan=10-11
dchan=12
 
# Global data
 
loadzone= au
defaultzone = au

 
Error recieved in console without group
 
-- Executing Dial("SIP/701-09f0fc18", "zap/0418608609") in new stack
Jun 27 19:27:13 NOTICE[4011]: app_dial.c:1089 dial_exec_full: Unable to
create channel of type 'zap' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/701-09f0fc18' status is 'CONGESTION'

 
 
Error recieved in console with g0 in the dial string
 -- Executing Dial("SIP/701-08d76e98", "zap/g0/0418608609") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/0418608609
-- Zap/1-1 is proceeding passing it to SIP/701-08d76e98
-- Channel 0/1, span 1 got hangup request
-- Channel 0/1, span 1 received AOC-E charging 0 units
Jun 27 19:28:35 WARNING[4046]: app_dial.c:738 wait_for_answer: Unable to
forward voice
-- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/701-08d76e98' status is
'CHANUNAVAIL'

 

Nathan Dennis 
__ 
Integrated Solutions (QLD)P/LPhone: +61 (7) 4044 0300 
Direct: +61 (7) 4044
0302
124 Spence Street   Fax:+61 (7) 4041 6600
CAIRNS QLD 4868Mobile: 0418 608609

Australia 

E-mail: [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> 
Web Site: www.i-solutions.net.au <http://www.i-solutions.net.au/> 

Offices and agents in Cairns - Brisbane - Melbourne -- Adelaide --
Sydney
__ 
The information transmitted is intended only for the person or entity to
which 
it is addressed and may contain confidential and/or privileged material.

Any review, retransmission, dissemination or other use of, or taking of
any 
action in reliance upon, this information by persons or entities other
than the 
intended recipient is prohibited. If you received this in error, please
contact 
the sender and delete the material from any computer.
 
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[asterisk-users] Xorcom Bri 4 Port USB

[Nathan Dennis]

Hi,
   I'm having some trouble setting up a Xorcom Bri 4 port. I have compiled 
asterisk and zaptel using the Bristuff bristuff-0.3.0-PRE-1y-g patches.

So I'm running zaptel-1.2.17.1 and asterisk-1.2.18.

The problem I'm having is that for one I get no LEDs showing if the unit is in 
TE and NT mode (not a issue for me but may have some impact on things) I have 
no errors in any logs I can see but once zaptel and asterisk are started up I 
get a lots of warnings in asterisk such as the following

Jun 25 18:18:07 WARNING[2833]: chan_zap.c:2662 pri_find_dchan: No D-channels 
available!  Using Primary channel 3 as D-channel anyway!
  == Primary D-Channel on span 2 down
Jun 25 18:18:07 WARNING[2834]: chan_zap.c:2662 pri_find_dchan: No D-channels 
available!  Using Primary channel 6 as D-channel anyway!
  == Primary D-Channel on span 3 down
Jun 25 18:18:07 WARNING[2835]: chan_zap.c:2662 pri_find_dchan: No D-channels 
available!  Using Primary channel 9 as D-channel anyway!
  == Primary D-Channel on span 1 down

It errors for all for ports and makes no difference if I have the ISDN cables 
connected or not. I want to run in ptp mode and currently use a digium B410P 
card on the connections that work fine so I know that the lines work and ptp is 
the correct mode.

Following are my configs. Any pointers you can give would be greatly 
appreciated.

We are running Fedora 7.
Kernel "Linux 2.6.21-1.3194.fc7 #1 SMP i686 i686 i386 GNU/Linux" (Standard 
Kernel with install)
Device has jumpers all set to TE mode.


/etc/init.d/zaptel.conf
# Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: BRI_TE"
span=1,1,1,ccs,ami
# termtype: te
bchan=1-2
dchan=3

# Span 2: XBUS-00/XPD-08 "Xorcom XPD #0/8: BRI_TE"
span=2,2,1,ccs,ami
# termtype: te
bchan=4-5
dchan=6

# Span 3: XBUS-00/XPD-10 "Xorcom XPD #0/16: BRI_TE"
span=3,3,1,ccs,ami
# termtype: te
bchan=7-8
dchan=9

# Span 4: XBUS-00/XPD-18 "Xorcom XPD #0/24: BRI_TE" RED
#span=4,4,1,ccs,ami
# termtype: te
#bchan=10-11
#dchan=12

# Global data

loadzone= au
defaultzone = au


/etc/asterisk/zapata.conf
[channels]
;   echocancel = yes
;   transfer = yes
;   threewaycalling = yes

#include zapata-channels.conf


/etc/asterisk/zapata-channels.conf

; Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: BRI_TE"
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel => 1-2
callerid=
group=
context=default

; Span 2: XBUS-00/XPD-08 "Xorcom XPD #0/8: BRI_TE"
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel => 4-5
callerid=
group=
context=default

; Span 3: XBUS-00/XPD-10 "Xorcom XPD #0/16: BRI_TE"
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel => 7-8
callerid=
group=
context=default

; Span 4: XBUS-00/XPD-18 "Xorcom XPD #0/24: BRI_TE" RED
;callerid=asreceived
;group=0
;context=from-pstn
;switchtype = euroisdn
;signalling = bri_cpe
;channel => 10-11
;callerid=
;group=
;context=default



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Re: [asterisk-users] sip to sip ?

[Dennis Kavadas]

Hi Rob.

The local * box works fine for all local sip calls to local sip calls

i have setup 2 voip handsets and they work well, even took one home
and tried it from my private nat'ed home network, all works, the
phones register and i can call the other extension, regardless of
location.

The * server is not firewalled at all and uses a public ip address.

the only problem seems to be that i can't call other * boxes or sip
users not local to my * box.




On 2/21/07, Rob Schall <[EMAIL PROTECTED]> wrote:

If you're getting a 404, I would assume it is reacting like any other
non-connection would (http, etc). Do you know if the packets are
reaching the phone, or if the phone is registering its correct IP
Address? If it is registering, but no packets are reaching it, could it
be a routing issue?

Rob

Chris Hills wrote:
> Dennis Kavadas wrote:
>
>> hi all
>>
>> i've just setup an * box and want to test voip calling, initially from
>> sip user to sip user...
>>
>> local sip users can call each other, no issues.
>>
>> problem arises when i try and call a remote sip account, my * box
>> always returns "SIP/2.0 404 Not Found"
>>
>> any ideas ?
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>
> Dennis
>
> I use the following as my default context:-
>
> [default]
> exten => _X.,1,NoOp(Incoming Call from ${CALLERID} for
> [EMAIL PROTECTED])
> exten => _X.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
> exten => _X.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
> exten => _X.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
> exten => _X.,5,GotoIf($[${SIPDOMAIN} = ${MYIP}]?10)
> exten => _X.,6,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
> exten => _X.,7,NoOp(@${SIPDOMAIN} is remote - forwarding...)
> exten => _X.,8,Macro(uridial,[EMAIL PROTECTED])
> exten => _X.,9,HangUp()
> exten => _X.,10,Goto(default-noturi,${EXTEN},1)
> exten => h,1,HangUp()
> exten => s-BUSY,1,Congestion
> exten => s-CHANUNAVAIL,1,Congestion
> exten => s-CONGESTION,1,Congestion
>
> [macro-uridial]
> exten => s,1,NoOp(Outbound SIP URI call ${ARG1})
> exten => s,2,SetCIDNum(0123456789)
> exten => s,3,Dial(SIP/${ARG1})
> exten => s,4,Congestion()
>
>
> HTH
>

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Re: [asterisk-users] sip to sip ?

[Dennis Kavadas]

why do i need to setup a trunk ?
all i want to do is place a sip connection to a remote sip user..

e.g...

[EMAIL PROTECTED]





On 2/20/07, Mochamad Susantok <[EMAIL PROTECTED]> wrote:

create user trunk on each box and dialplan to make call
> hi all
>
> i've just setup an * box and want to test voip calling, initially from
> sip user to sip user...
>
> local sip users can call each other, no issues.
>
> problem arises when i try and call a remote sip account, my * box
> always returns "SIP/2.0 404 Not Found"
>
> any ideas ?
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[asterisk-users] sip to sip ?

[Dennis Kavadas]

hi all

i've just setup an * box and want to test voip calling, initially from
sip user to sip user...

local sip users can call each other, no issues.

problem arises when i try and call a remote sip account, my * box
always returns "SIP/2.0 404 Not Found"

any ideas ?
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[asterisk-users] SIP/2.0 404 Not Found

[Dennis Kavadas]

anyone know what doing on here ?
i can make local to local calls but no local to remote or remote to local.
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[asterisk-users] Digium Card ?

[Dennis Kavadas]

Hi all

I'm after a Digium card that will allow me to connect an Asterisk box to..

2 x sip providers
1 x company PBX
1 x POTS provider.

Can anyone recomment a card that can do the job.
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[asterisk-users] windows SIP Softphones ?

[Dennis Kavadas]

hi all
what do must win32 clients use as a free or OSS sip softphone ?
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[asterisk-users] make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list->next != 0' failed.

[Dennis Kavadas]

hi all

i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0
any suggestions ?

make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect'
make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect'
Generating input for menuselect ...
menuselect/menuselect --check-deps   menuselect.makeopts
Generating embedded module rules ...
  [CC] astman.c -> astman.o
  [CC] md5.c -> md5.o
  [LD] astman.o md5.o -> astman
  [CC] smsq.c -> smsq.o
  [CC] strcompat.c -> strcompat.o
  [LD] smsq.o strcompat.o -> smsq
  [CC] stereorize.c -> stereorize.o
  [CC] frame.c -> frame.o
  [LD] stereorize.o frame.o -> stereorize
  [CC] streamplayer.c -> streamplayer.o
  [LD] streamplayer.o -> streamplayer
make: expand.c:489: allocated_variable_append: Assertion
`current_variable_set_list->next != 0' failed.
make: *** [utils] Aborted
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Re: [asterisk-users] T1 pricing in Oz

[Dennis Gilmore]

On Thursday 19 October 2006 14:43, Forum wrote:
> I'm looking at getting a T1 into a location in Melbourne, Australia and was
> wondering if anyone has a good source and pricing for this.
I think your looking for a E1  Australia follows the European standard  last i 
looked.  I never did any voice stuff when i lived in Oz.  but you could try 
Telstra, Optus, etc,  or look in the phone book for a reseller.   Someone  
probably has alot better experience than me.

Dennis
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RE: [asterisk-users] T1 Passthrough

[Dennis Walker]

Assuming the outgoing T1 to the Norstar is a standard T1 that accepts ANI and 
DNIS all have to do is

exten => _XXX,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)

This will redial the caller id  (ANI) and the 3 digits Dialed (DNIS)  to the 
Norstar T1 in the formst  *ANI*DNIS*

I did the same thing for a while to convert and PRI to a T1 into a Mitel system 
that could no do PRI without a
very expensive upgrade.

--
From:   Forrest Beck[SMTP:[EMAIL PROTECTED]
Reply To:   Asterisk Users Mailing List - Non-Commercial Discussion
Sent:   Monday, October 09, 2006 10:18 PM
To: Asterisk Users List
Subject:[asterisk-users] T1 Passthrough

I want to setup a asterisk server with two T1 spans (two TE110P
cards).  The server will have one card connected to the PRI and the
other will connect to our Norstar Meridian ICS system.  I want to have
a very simple dial plan for the context that the PRI card will be
assigned to something like this.  Note that our telecom provider sends
final three digits of the phone number:

SPAN 1
Channels 1-23
g1
context: pri_incoming

SPAN 2
Channels 25-48
g2
context: norstar_ics

[pri_incoming]
exten => _XXX,1,Dial,ZAP/g2/${EXTEN}

My questions are:

Will I need to set the callerid before routing to the next span, or
will the three digits remain intact.?
and
Has anyone tried this? and if so do you forsee any problems i will run into?

This is all theroey in my head right now, since I am awaiting the
second cards arrival.

Thanks.
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[asterisk-users] Cannot hear the other side of the phone call

[Dennis P. Clark]

I have had Asterisk 1.2.10 up and running for the past two months.  I
have not done anything to the system in the last month.  

I am using broadvoice.com as a sip provider.  Yesterday everything was
working fine and now when I call out or receive calls I cannot hear the
person on the other line, however they can hear me just fine.  When I
call internally to another extension both parties can hear eachother.
This only seems to be happening when I dial out.  

Additionally I setup a soft phone (X-Lite) and connected directly to the
broadvoice.com sip server and I was able to communicate perfectly.  

Any Ideas?

Thanks,
Dennis


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RE: [asterisk-users] LOUD MP3 Hold Music

[Dennis P. Clark]

If MP3s are too loud then their should be an internal function that
modifies all MP3s in the folder to be at one consistent volume
(normalization).  Anybody know of a way to do this?

Here is my musiconhold.conf just in case I missed something.

[classes]
Random => quietmp3:/var/lib/asterisk/mohmp3


Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
 
 
 
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Tuesday, August 22, 2006 2:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] LOUD MP3 Hold Music

David Freeman wrote:
> I have the opposite problem.  I can hardly hear the hold music at all.
> 
> On 8/22/06, *Dennis P. Clark* <[EMAIL PROTECTED] 
> <mailto:[EMAIL PROTECTED]>> wrote:
> 
> How do you lower the volume of MP3 hold music?

I'm certainly not an expert on MOH, but I don't believe there are any
volume control knobs to be tweaked in asterisk itself. You might want to
take a look at the configs/musiconhold.conf.sample file as there were
some parameters that impacted high/med/quiet modes, but I'm thinking
they only applied to the old mpg123 music app. Could easily be wrong.

Best guess... probably have to use something like sox to change the
volume of the mp3 (or whatever) file itself.

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[asterisk-users] LOUD MP3 Hold Music

[Dennis P. Clark]

How do you lower the volume of MP3 hold music?  

Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
 
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RE: [asterisk-users] Zaptel install - Fedora Core 5

[Dennis P. Clark]

I couldn't find 2.6.17-1 for download but this is what I used to install the 
kernel source
http://download.fedora.redhat.com/pub/fedora/linux/core/5/source/SRPMS/GFS-kernel-2.6.15.1-5.FC5.17.src.rpm

Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
 
 
 
 


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of simon elliston 
ball
Sent: Monday, August 21, 2006 7:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zaptel install - Fedora Core 5

I managed to get zaptel to compile reasonably easily on 2.6.17-1.2157_FC5smp. 
However, the yum repo sites do not provide devel packages for 2.6.17-2174 for 
some reason last time I checked, hence couldn't get it to build on that kernel. 
You could probably create the devel package without too much trouble from the 
srpm, but it's a lot easier to stick to 2157.

If anyone else have managed to get FC5 to install the correct devel packages 
for the latest kernel, please let me know!

Simon

On 21 Aug 2006, at 11:52, Tomislav Parčina wrote:

> I'm trying to install Zaptel 1.2.7 on Fedora Core 5 with
> 2.6.17-1.2174_FC5 kernel from source code. When I untar Zaptel and 
> execute this is error that I get.
>
> cc -o ztmonitor ztmonitor.o
> cc -o ztspeed.o -c ztspeed.c
> cc -o ztspeed ztspeed.o
> cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE- 
> DSTANDALONE_ZAPATA -DZAPTE
> L_CONFIG=\"/etc/zaptel.conf\"zttest.c   -o zttest
> cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE- 
> DSTANDALONE_ZAPATA -DZAPTE
> L_CONFIG=\"/etc/zaptel.conf\"   -c -o fxotune.o fxotune.c
> cc -o fxotune fxotune.o -lm
> /lib/modules/2.6.17-1.2174_FC5/build
> You do not appear to have the sources for the 2.6.17-1.2174_FC5 kernel 
> installed .
> make: *** [linux26] Error 1
>
> What could be the problem? How to solve it?
>
>
> --
> 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] Page()

[Dennis P. Clark]

That fixed it.  Thanks!

Here is what I did to fix

Asterisk -crv (enter CLI)
Stop gracefully (Shutdown Asterisk)
Cd /usr/src/asterisk-1.2.10 (Go to unpacked Asterisk installation files)
Make install (install asterisk)
Asterisk (start asterisk)

Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
 
 
 
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Wednesday, August 16, 2006 5:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Page()

I guess it might not get compiled if you don't have a timer.
Install ztdummy, recompile asterisk and try again.

On 8/16/06, Dennis P. Clark <[EMAIL PROTECTED]> wrote:
>
> I am running Fedora 5
> Cat /proc/sys/kernel/osrelease
> 2.6.15-1.2054_FC5
>
> Zaptel 1.2.7 was not installed
> Edited xpp_usb.c and wcusb.c files in Zaptel to get it to 
> compile and install by commenting out the following
> .owner = THIS_MODULE,
>
> I receive the following from CLI when I run "load module app_page.so"
> WARNING[28359]: loader.c:325 __load_resource:
> /usr/lib/asterisk/modules/app_page.so: cannot open shared object file:
> No such file or directory
>
> And yes app_page.so does not exist in /usr/lib/asterisk
>
> Dennis Clark
> DENPRO
> WRK 207.618.1998
> CEL 443.415.0527
> FAX 1.888.811.8809
> [EMAIL PROTECTED]
>
>
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of C F
> Sent: Wednesday, August 16, 2006 12:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Page()
>
> in the CLI do:
> show applications like page
> if you something there then you have it loaded, otherwise do:
> load app_page.so
> if that fails my guess is you need zaptel loaded first.
>
> On 8/16/06, Dennis P. Clark <[EMAIL PROTECTED]> wrote:
> > 1.2.10
> >
> > Dennis Clark
> > DENPRO
> > WRK 207.618.1998
> > CEL 443.415.0527
> > FAX 1.888.811.8809
> > [EMAIL PROTECTED]
> >
> >
> >
> >
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Doug 
> > Lytle
> > Sent: Wednesday, August 16, 2006 11:10 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Page()
> >
> > Dennis P. Clark wrote:
> > > I receive the following error in the Asterisk console when I try 
> > > to execute the Page() application:
> > >
> > > WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application
> 'Page'
> > > for extention (intercom, *, 1)
> > >
> >
> > What version of Asterisk are you running?
> >
> > 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] Page()

[Dennis P. Clark]

I am running Fedora 5
Cat /proc/sys/kernel/osrelease
2.6.15-1.2054_FC5

Zaptel 1.2.7 was not installed
Edited xpp_usb.c and wcusb.c files in Zaptel to get it to
compile and install by commenting out the following
.owner = THIS_MODULE,

I receive the following from CLI when I run "load module app_page.so"
WARNING[28359]: loader.c:325 __load_resource:
/usr/lib/asterisk/modules/app_page.so: cannot open shared object file:
No such file or directory

And yes app_page.so does not exist in /usr/lib/asterisk

Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
 
 
 
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Wednesday, August 16, 2006 12:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Page()

in the CLI do:
show applications like page
if you something there then you have it loaded, otherwise do:
load app_page.so
if that fails my guess is you need zaptel loaded first.

On 8/16/06, Dennis P. Clark <[EMAIL PROTECTED]> wrote:
> 1.2.10
>
> Dennis Clark
> DENPRO
> WRK 207.618.1998
> CEL 443.415.0527
> FAX 1.888.811.8809
> [EMAIL PROTECTED]
>
>
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Doug 
> Lytle
> Sent: Wednesday, August 16, 2006 11:10 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Page()
>
> Dennis P. Clark wrote:
> > I receive the following error in the Asterisk console when I try to 
> > execute the Page() application:
> >
> > WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application
'Page'
> > for extention (intercom, *, 1)
> >
>
> What version of Asterisk are you running?
>
> 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] Page()

[Dennis P. Clark]

1.2.10

Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
 
 
 
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Wednesday, August 16, 2006 11:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Page()

Dennis P. Clark wrote:
> I receive the following error in the Asterisk console when I try to 
> execute the Page() application:
>
> WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
> for extention (intercom, *, 1)
>   

What version of Asterisk are you running?

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] Page()

[Dennis P. Clark]

What is the module I should be loading and how do I load it?

Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
 
 
 
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leonardo
Kamache (Gmail)
Sent: Wednesday, August 16, 2006 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Page()

Hi there;

Did you load the respective module?


Regards;

LK



On 8/16/06, Dennis P. Clark <[EMAIL PROTECTED]> wrote:
> I receive the following error in the Asterisk console when I try to 
> execute the Page() application:
>
> WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
> for extention (intercom, *, 1)
>
> EXTENSIONS.CONF
> [Default]
> Exten => *80,1,Goto(intercom,s,1)
>
> [intercom]
> exten => s,1,Answer
> exten => s,n,SIPAddHeader(Call-Info: answer-after=0) exten => 
> s,n,Playback(beep) exten => s,n,Set(TIMEOUT(digit)=5) exten => 
> s,n,WaitExten(10)
>
> ;Page
> exten => *,1,Page(SIP/2000x1)
>
> ;Intercom
> exten => _,1,Dial(SIP/${EXTEN})
>
> Any clues?
>
> Dennis Clark
> DENPRO
> WRK 207.618.1998
> CEL 443.415.0527
> FAX 1.888.811.8809
> [EMAIL PROTECTED]
>
>
>
>
>
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RE: [asterisk-users] Page()

[Dennis P. Clark]

I receive the following error in the Asterisk console when I try to
execute the Page() application:

WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)

EXTENSIONS.CONF
[Default]
Exten => *80,1,Goto(intercom,s,1)

[intercom]
exten => s,1,Answer
exten => s,n,SIPAddHeader(Call-Info: answer-after=0)
exten => s,n,Playback(beep)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,WaitExten(10)

;Page
exten => *,1,Page(SIP/2000x1)

;Intercom
exten => _,1,Dial(SIP/${EXTEN})

Any clues?

Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
 
 
 


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[asterisk-users] Connecting to another server

[Dennis Wambugu]

Hi, 

 

I would like some advice on how to configure my asterisk such that
local users with IAX and SIP extensions can utilize a SIP account from another
server, to make international calls. 

 

In my understanding this would be possible by them dialing a certain
prefix say 888, followed by destination number. 

 

Please assist.

 

Rgds

Dennis






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[asterisk-users] creidt card processing sripts for asterisk

[Dennis Nacino]

Hi Joseph,

I think you didn't made it clear. My understanding of the script you want, is 
to make the process
of calling the bank's IVR system and responding to its prompts/recording via 
Asterisk scripts. If
my understanding is correct, the asterisk send the DTMF needed by the prompt. 
But then again, I
don't think Dial application fit in that requirement. It can only send strings 
of DTMF just before
the call bridge. Perhaps you might want to do it with the callfile, then once 
answered, run
senddDTMF, playback(silence) between prompts. But, then again I don't know to 
which channel these
application will communicate.


Regards,



 

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

[Dennis Gilmore]

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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[asterisk-users] TE110P configuration problem

[Dennis Nacino]

Read my post in this last April "[Asterisk-Users] R2 protocol error" you can 
find some guide from
the Unicall creator. Goodluck!
Also I think the subject wont attract Asterisk-MFC-R2 user, since its a kind of 
zaptel
configuration issue.


Dennis

  


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[asterisk-users] MFC/R2 country and carrier specific protocol variants

[Dennis Nacino]

Hi Paulo,

I'm from Philippines and here's the protocol variant line I use for our R2 
provider (Nextel
Philippines) 

protocolvariant=ph,12,18,1

But it never reach production stage pending resolution of the problem I post in 
this list last May
"[Asterisk-Users] Unicall MFC/R2 B3,B4 and clear back". Anyway, I presumed 
you've been using
UNICALL/R2 channel in production. May, I know how did you deal with that 
problem. Should I
presumed that since R2 is so variant, somehow you've been spare.


Regards,


Dennis
  


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[Asterisk-Users] password on radius authentication

[Dennis Nacino]

Hi,


It's kind of off-topic , but still within Asterisk. I developed an asterisk 
module that send an
authentication to a radius server for call authorization and process its reply 
(limited to
User-Name and Cisco or Quintum VSA h323 attribute). My question, is when it 
make sense to use or
include the attribute Password/User-Password? Looking on PDF's of Quintum and 
Cisco none of it
really make use of this attribute. Any comment?



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[Asterisk-Users] Unicall MFC/R2 B3,B4 and clear back

[Dennis Nacino]

Hi All,


I have an R2 installation still undergoing testings, during the test I notice 
that the Unicall
always respond B6 to a II-1 (from a forward switch). Except, for a DNIS that 
can't be found in the
dial plan, in this case it respond with B5. My real problem is, the call will 
be terminate on a
Cisco 7206 with ISDN/PRI thru SIP. If the Called number is busy or the Cisco 
7206 is busy or
congested, it seems there's no way for Unicall to issue B3 or B4 since its 
already on accepted
state. Please see the log below;

May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17  <- 1 on 
[2/   2/Group B   /Go to grp II ]
May  3 12:51:11 WARNING[11325]: chan_unicall.c:2644 handle_uc_event: Unicall/17 
event Offered
May  3 12:51:11 WARNING[11325]: chan_unicall.c:2699 handle_uc_event: CRN 32782 
- Offered on
channel 0 (ANI: 09797280105, DNIS: 0015107973287, Cat: 0)
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Call
control(4)
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Accept call
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 6 on  -> 
[2/   4/Group B   /Go to grp II ]
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17  <- 1 off
[2/   4/Group B   /Accepted Paid]
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 6 off -> 
[2/   4/Group B   /Accepted Paid]
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Answer guard
expired
May  3 12:51:11 WARNING[11325]: chan_unicall.c:2644 handle_uc_event: Unicall/17 
event Accepted
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Channel gains
-- Executing Dial("UniCall/17-1", "SIP/aaa.bbb.ccc.ddd/15107973287|45||") 
in new stack
-- Called aaa.bbb.ccc.ddd/15107973287
-- Got SIP response 486 "Busy here" back from aaa.bbb.ccc.ddd
-- SIP/aaa.bbb.ccc.ddd-7bad is busy
  == Everyone is busy/congested at this time (1:1/0/0)
  == Auto fallthrough, channel 'UniCall/17-1' status is 'BUSY'
May  3 12:51:20 WARNING[12011]: chan_unicall.c:2441 unicall_indicate: 
unicall_indicate 5
May  3 12:51:31 WARNING[12011]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Channel gains
May  3 12:51:31 WARNING[12011]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Channel
switching
May  3 12:51:31 WARNING[12011]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Call
control(6)
May  3 12:51:31 WARNING[12011]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Drop
call(cause=User busy [17])
May  3 12:51:31 WARNING[12011]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 1101  -> 
[1/  20/Group B   /Accepted Paid]
-- Hungup 'UniCall/17-1'

 The worst part of it, the forward switch, look lost and never respond to that 
clearback thus
never release the channel. 
 As a another test I called an extension with Busy as an asterisk application, 
it still respond
with B6.
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17  <- 1 on 
[2/   2/Group B   /Go to grp II ]
May  3 13:21:50 WARNING[11325]: chan_unicall.c:2644 handle_uc_event: Unicall/17 
event Offered
May  3 13:21:50 WARNING[11325]: chan_unicall.c:2699 handle_uc_event: CRN 32783 
- Offered on
channel 0 (ANI: 09797280105, DNIS: 006321234569, Cat: 0)
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Call
control(4)
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Accept call
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 6 on  -> 
[2/   4/Group B   /Go to grp II ]
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17  <- 1 off
[2/   4/Group B   /Accepted Paid]
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 6 off -> 
[2/   4/Group B   /Accepted Paid]
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Answer guard
expired
May  3 13:21:50 WARNING[11325]: chan_unicall.c:2644 handle_uc_event: Unicall/17 
event Accepted
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Channel gains
-- Executing Busy("UniCall/17-1", "8") in new stack
May  3 13:21:50 WARNING[12259]: chan_unicall.c:2441 unicall_indicate: 
unicall_indicate 5
  == Spawn extension (nextel-r2, 006321234569, 1) exited non-zero on 
'UniCall/17-1'
May  3 13:21:59 WARNING[12259]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Channel gains
May  3 13:21:59 WARNING[12259]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Channel
switching
May  3 13:21:59 WARNING[12259]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Call
control(6)
May  3 13:21:59 WARNING[12259

RE: [Asterisk-Users] E1 PRI problem with TE205P

[Dennis Walker]

I had a simular problem, but my configuration would last a couple of days and 
then
go crazy.  I had to restart the computer to get the PRI to work right.  

I turned out the problem was shared interrupts.  I didn't have as many
problems until I went to 1.2.4 and it was worse with 1.2.5.

I had to rearrange my PCI cards and disable onboard sound to get free 
interrupts; but since
then it's run 21 days with out any problems.

--
From:   Phone Dev[SMTP:[EMAIL PROTECTED]
Reply To:   Asterisk Users Mailing List - Non-Commercial Discussion
Sent:   Monday, April 10, 2006 1:56 PM
To: asterisk-users@lists.digium.com
Subject:[Asterisk-Users] E1 PRI problem with TE205P

Hi all,
I've setup a PBX in production environment last week but we have immediatly
find out drop conversations and lot of errors (in asterisk logs) like:

Apr 10 17:30:48 NOTICE[25154] chan_zap.c: PRI got event: HDLC Bad
FCS (8) on Primary D-channel of span 2


My configuration is:

Motherboard: SuperMicro P8SCT (Intel E7221 chipset)
Hypertrading and APIC enabled
CPU: P4 3GHz
Digium TE205P
Digium TDM400P with 4 FXS module
Linux Centos 4.1 (kernel 2.6.9-22.Elsmp ) 
Asterisk 1.2.5 (using [EMAIL PROTECTED] 2.7)

On span 1 of TE205P (configured as T1 PRI) is connected to a Rhino Channel
Bank 24 FXS
On span 2 of TE205P (configured as E1 PRI) is connected to Telco E1.

Both digium cards (TE205P and TDM400P) uses IRQ 9 (APIC is enable and
"virtual" IRQ is 161) but TDM400 is not used for the moment and I'm going to
disable it if could be usefull. IRQ 9 is not shared to other PCI card
installed. I've disable USB.

I've read a new comment on
http://www.voip-info.org/wiki/view/Asterisk+hardware about a similar problem
using TE110P and same MB that was solved using Sangoma A101 card. 

So, does TE205P may not work correctly on a SuperMicro P8SCT MB ? Have I to
buy a new A102 Sangoma card ?
There is any configurations may solve this problem ?
I'm going to try to disable hyperthreading but I'm not sure it may help.
Any help will be welcome. 
Thanks,


Simone Caretta





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[Asterisk-Users] R2 protocol error

[Dennis Nacino]

Hi,


Thanks a lot, guys! The problem is now fixed by updating the libmfcr2-0.0.3 to 
pre9 and setting
the span timing correctly.


Dennis




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[Asterisk-Users] R2 protocol error

[Dennis Nacino]

Hi MM and Steve,

I still got the same problem when I changed the span configuration setting into

span=1,1,0,cas,hdb3

Where can I get the pre9? Is there something wrong with www.soft-switch.org 
site? It seems
unreachable.

Thanks again.


Dennis






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[Asterisk-Users] Re: How to use Sendtxt?

[Dennis Nacino]

Here's a good link.  

http://www.asteriskguru.com/tutorials/sendtext.html

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[Asterisk-Users] R2 protocol error

[Dennis Nacino]

Hi,

I have three R2 installation on different carriers, all shows the same 
inconsistency at varying
degree. But, on most test calls we made, it reaches T3. The worst part of 
these, the carrier
claims that it's my R2 box that is not responding in time. Please, check the 
attached file and
take note of the timestamp, you'll find that in some call, it already 
contradict what the carrier
claims but they too have logs to counter my claim. So, I hope people, please 
give me a good
insight and direction to resolve this problem.

I have the following for my R2 box:
unicall-0.0.3pre8
 libmfcr2-0.0.3
 libsupertone-0.0.2
 libunicall-0.0.3
spandsp-0.0.2pre25

asterisk-1.2.6
zaptel-1.2.5
wanpipe-2.3.3-2

2.6.11-1.1369_FC4smp
sangoma A101

in my zaptel.conf I got the following:

span=1,0,0,cas,hdb3
loadzone = us
defaultzone=us
cas=1-15:1101
cas=17-31:1101

in my unicall.conf I got these lines:

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
supertones=ph
loglevel=255
protocolclass=mfcr2
protocolvariant=ph,10,3,12
protocolend=co
group = 1
channel => 1-15
channel => 17-31


Thanks a lot.

Dennis






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http://mail.yahoo.com Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 0001  [1/   1/Idle  /Idle ]
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 Detected
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 Making a new call with CRN 32769
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1101  ->  [2/   2/Idle  /Idle ]
Apr  3 11:34:54 WARNING[17334]: chan_unicall.c:2644 handle_uc_event: Unicall/1 
event Detected
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 on  [2/   2/Seize ack /Seize ack]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 on  ->  [2/   2/Seize ack /Seize ack]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 off [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 on  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 on  ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 off [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 3 on  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 5 on  ->  [2/   2/Group A   /DNIS request ]
Apr  3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 1 on  [2/   2/Group A   /Category req ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 1 off [2/   2/Group A   /ANI request  ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 5 off ->  [2/   2/Group A   /ANI request  ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 R2 prot. err. [2/   2/Group A   /ANI request  ] cause 32771 - 
T3 timed out
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:2644 handle_uc_event: Unicall/1 
event Protocol failure
-- Unicall/1 protocol error. Cause 32771
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 Channel echo cancel
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1  <- 1001  [1/   1/Idle  /Idle ]
Apr  3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
==
Apr  3 11:41:49 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCa

RE: [Asterisk-Users] Config TE110P and TDM400 with 2 FXS modules

[Dennis]

Hi,

Thanks for that Paul, it has solved that problem at least.
In doing so however it broke the incoming calls on the other card.
To fix this we need both immediate=yes set before the TE110P channels, and
then immediate=no after that but before the TDM channel assignment.

It all works for now, but is this how the configuration for these files is
meant to be?

Either way, thanks for your help. This has had me completely confused for
several days now.

Regards,
 
Dennis Corp
Uniware Pty Ltd.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Liew
Sent: Friday, 31 March 2006 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Config TE110P and TDM400 with 2 FXS modules

Hi Dennis,

You need to add "immediate=no" before your channel assignment; asterisk 
will then give you a dialtone when you pick up the handset. Also the 
context "uniware_sendfax" must have the pattern match and associated zap 
dial that you need.

HTH.

Cheers,
Paul

www.austechpartnerships.com
t) +61 (0)3 9221 0888
SIP) [EMAIL PROTECTED]
IAX) [EMAIL PROTECTED]
IAXtel) 1700-482-8273
ATP Centrex) 


Dennis wrote:
>
> signalling=fxs_ks
>
> context=uniware_sendfax
>
> channel =>33-34
>
> -=-=-=-
>
>  
>
> What we want to be able to do is plug a standard telephone into an fxs 
> port on the tdm card, and have it get a line from the E1 when the 
> handset is picked up.
>
>  
>
> exten => _NXXX,1,Dial(Zap/g1/${EXTEN}|20,t)
>
>  
>
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[Asterisk-Users] Config TE110P and TDM400 with 2 FXS modules

[Dennis]

Hi,

 

Let me explain a little about our system here first.

 

We have a Digium TE110P card hooked up to an isdn 30 line. (Australia
– EuroISDN) This works fine.

We also have a Digium TDM400P with 2 FXS modules installed.
(The green modules)

 

The /etc/zaptel.conf file has nothing but the following in
it.

-=-=-=-

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

dchan=16

bchan=1-15,17-31

defaultzone=au

loadzone=au

##tdm card

fxoks=33-34

-=-=-=-

 

The /etc/asterisk/zapata.conf has been stripped down to
include only the following.

-=-=-=-

[trunkgroups]

 

[channels]

language=en

rxwink=300

usecallerid=yes

hidecallerid=no

callwaiting=yes

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

canpark=yes

cancallforward=yes

callreturn=yes

rxgain=0.0

txgain=0.0

group=1

callgroup=1

pickupgroup=1

immediate=yes

faxdetect=both

jitterbuffers=24

signalling=pri_cpe

switchtype=euroisdn

echocancel=yes

echocancelwhenbridged=yes

echotraining=400

context=default

channel => 1-15,17-31 ; Set this to 1-15,17-31 for E1

signalling=fxs_ks

context=uniware_sendfax

channel =>33-34

-=-=-=-

 

What we want to be able to do is plug a standard telephone
into an fxs port on the tdm card, and have it get a line from the E1 when the handset
is picked up.

 

The problem is that when the handset is picked up, the phone
automatically gets picked up by asterisk as an incoming call, and the call is managed
by the incoming call part of the extensions.conf.

IE: the part of the extensions.conf with the following line.

Exten => s,1,MakeReceptionPhoneRing.

This is fantastic if I needed an instant dial to chat with
our receptionist.

What I was expecting was that it would go to the part in the
extensions.conf where it detects the numbers dialed.

Ie: 

exten => _NXXX,1,Dial(Zap/g1/${EXTEN}|20,t)

 

Thus allowing us to dial a number and have asterisk direct
it to the appropriate outgoing line.

 

I could possibly be going about this completely the wrong
way. Ideally I would have thought that a bridge between cards was possible
outside of asterisk.

 

Any ideas or thoughts on this would be appreciated.

 

 

--Dennis

 

 

 






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[Asterisk-Users] Random Zap port going crazy When channel released after a flash.

[Dennis Walker]

On 1.2.x I have a random problem when a Zap/x channel flashes to transfer 
or make a three way call.

The Zap/x-2 channel is created and the transfer or three way proceeds, but 
on hangup the Zap/x-1 channel fails to destroy the old bridge and asterisk 
goes crazy logging the problem.  Here is an example debug log.

This happens only once a day or so, with 100 or so users transfering and 
three way calling all the time.

Anyone having a simular problem.


Thanks for you help

Mar  7 11:21:29 VERBOSE[8204] logger.c: -- Starting simple switch on 
'Zap/99-1'
Mar  7 11:21:31 DEBUG[8204] chan_zap.c: DTMF digit: 8 on Zap/99-1
Mar  7 11:21:32 DEBUG[8204] chan_zap.c: DTMF digit: 3 on Zap/99-1
Mar  7 11:21:32 DEBUG[8204] chan_zap.c: DTMF digit: 3 on Zap/99-1
Mar  7 11:21:33 DEBUG[8204] chan_zap.c: DTMF digit: 5 on Zap/99-1
Mar  7 11:21:33 DEBUG[8204] chan_zap.c: DTMF digit: 6 on Zap/99-1
Mar  7 11:21:33 DEBUG[8204] chan_zap.c: DTMF digit: 8 on Zap/99-1
Mar  7 11:21:33 DEBUG[8204] chan_zap.c: DTMF digit: 6 on Zap/99-1
Mar  7 11:21:34 DEBUG[8204] chan_zap.c: DTMF digit: 8 on Zap/99-1
Mar  7 11:21:34 DEBUG[8204] chan_zap.c: Enabled echo cancellation on 
channel 99
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Executing 
SetCallerID("Zap/99-1", "9377738550") in new stack
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Executing 
SetCallerPres("Zap/99-1", "allowed") in new stack
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Executing Dial("Zap/99-1", 
"Zap/G1/9373356868||Wg") in new stack
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Requested transfer 
capability: 0x00 - SPEECH
Mar  7 11:21:34 DEBUG[25354] channel.c: Avoiding initial deadlock for 
'Zap/22-1'
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Called G1/9373356868
Mar  7 11:21:34 DEBUG[25368] chan_zap.c: Queuing frame from 
PRI_EVENT_PROCEEDING on channel 0/22 span 1
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Zap/22-1 is proceeding 
passing it to Zap/99-1
Mar  7 11:21:34 DEBUG[8204] chan_zap.c: Requested indication 15 on channel 
Zap/99-1
Mar  7 11:21:34 DEBUG[8204] chan_zap.c: Received AST_CONTROL_PROCEEDING on 
Zap/99-1
Mar  7 11:21:36 DEBUG[25368] chan_zap.c: Enabled echo cancellation on 
channel 22
Mar  7 11:21:36 DEBUG[25354] channel.c: Avoiding initial deadlock for 
'Zap/22-1'
Mar  7 11:21:36 VERBOSE[8204] logger.c: -- Zap/22-1 is ringing
Mar  7 11:21:36 DEBUG[8204] chan_zap.c: Requested indication 3 on channel 
Zap/99-1


Mar  7 11:21:56 VERBOSE[8204] logger.c: -- Zap/22-1 answered Zap/99-1
Mar  7 11:21:56 DEBUG[8204] chan_zap.c: Requested indication -1 on channel 
Zap/99-1
Mar  7 11:21:56 DEBUG[8204] chan_zap.c: Took Zap/99-1 off hook
Mar  7 11:21:56 VERBOSE[8204] logger.c: -- Attempting native bridge of 
Zap/99-1 and Zap/22-1

Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Exception on 145, channel 99
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Got event Wink/Flash(3) on channel 
99 (index 0)
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Winkflash, index: 0, normal: 145, 
callwait: -1, thirdcall: -1
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Already have a dsp on Zap/99-2?
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Swapping 2 and 0
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: disabled echo cancellation on 
channel 99

>   Mar  7 11:22:03 VERBOSE[8229] logger.c: -- Starting simple switch 
on 'Zap/99-2'

Mar  7 11:22:03 VERBOSE[8204] logger.c: -- Started three way call on 
channel 99
Mar  7 11:22:03 VERBOSE[8204] logger.c: -- Started music on hold, class 
'default', on channel 'Zap/22-1'
Mar  7 11:22:03 DEBUG[8204] channel.c: Scheduling timer at 160 sample 
intervals
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Updated conferencing on 99, with 0 
conference users
Mar  7 11:22:03 DEBUG[8204] channel.c: Generator got voice, switching to 
phase locked mode
Mar  7 11:22:03 DEBUG[8204] channel.c: Scheduling timer at 0 sample 
intervals
Mar  7 11:22:03 DEBUG[8204] channel.c: Auto-deactivating generator
Mar  7 11:22:03 VERBOSE[8204] logger.c: -- Stopped music on hold on 
Zap/22-1
Mar  7 11:22:03 DEBUG[8204] channel.c: Scheduling timer at 0 sample 
intervals


>   Mar  7 11:22:04 DEBUG[8229] chan_zap.c: DTMF digit: 8 on Zap/99-2
Mar  7 11:22:05 DEBUG[8229] chan_zap.c: DTMF digit: 3 on Zap/99-2
Mar  7 11:22:05 DEBUG[8229] chan_zap.c: DTMF digit: 3 on Zap/99-2
Mar  7 11:22:05 DEBUG[8229] chan_zap.c: DTMF digit: 5 on Zap/99-2
Mar  7 11:22:05 DEBUG[8229] chan_zap.c: DTMF digit: 6 on Zap/99-2
Mar  7 11:22:06 DEBUG[8229] chan_zap.c: DTMF digit: 8 on Zap/99-2
Mar  7 11:22:06 DEBUG[8229] chan_zap.c: DTMF digit: 6 on Zap/99-2
Mar  7 11:22:06 DEBUG[8229] chan_zap.c: DTMF digit: 8 on Zap/99-2
Mar  7 11:22:06 DEBUG[8229] chan_zap.c: Enabled echo cancellation on 
channel 99
Mar  7 11:22:06 VERBOSE[8229] logger.c: -- Executing 
SetCallerID("Zap/99-2", "9377738550") in new stack
Mar  7 11:22:06 VERBOSE[8229] logger.c: -- Executing 
SetCallerPres("Zap/99-2", "allowed") in new

Re: [Asterisk-Users] Voipsupply - my experience

[Dennis Gilmore]

On Thursday 15 December 2005 11:55, Terry H. Gilsenan wrote:
> Hi,
>
> I would just like to let everyone know about the support I have rec'd
> from voipsupply.
>
> I travelled from .au to the US to do some urgent server upgrades for
> $EMPLOYER. During this trip I has impressed upon me (with absolutely
> zero notice) the requirement for a IP Phone link to the .au office.
>
> With only a couple of day left in the US I was stumped.
>
> I called Voipsupply and left a message (it was 9:30PM CST - Houston) at
> 10:30PM Mark emailed me and sent me his Cell phone number.
>
> The upshot is:
>
> I placed an order for some Digium kit at Midnight, on the 14th and I
> have just rec'd the goods in Houston now.
>
> Mark: Fantastic service! I appreciate the help.
>
> Regards,
Having moved from AU  to US last year I know how important a good reliable and 
cheap link back home is.   for this i use digium equipment and asterisk  with 
an AU iax provider.  i have a local number in Brisbane.  I have also 
purchased some VoIP equipment from Voipsupply for my employer and have 
received fantastic service and support from them for that also.

Dennis
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Re: [Asterisk-Users] RE:IConnecthere dial out problems

[Dennis Gilmore]

Once upon a time Wednesday 07 December 2005 8:42 pm, John Voss wrote:
> Your SIP.conf file looks much different than mine. I'll give it a try.

Hope mine helped

> [iconnect]
> type=friend
> secret=
> username=
> host=213.137.73.140 ;sipauth.deltathree.com
> permit=213.137.73.140/255.255.255.0
> permit=208.170.168.0/255.255.255.0
> disallow=all
> context=incoming
> allow=gsm
> allow=ulaw
> allow=alaw
> allow=G726
> insecure=very
> nat=Yes
> canreinvite=no
>
> I don't know what your register line looks like in your SIP.conf. This is
> mine.
>
> register => ::@213.137.73.140:5060
>
> I was unable to receive calls until I added the insecure=very line.
mine is register => ::@natrelay.deltathree.com

i can receive incomming calls for a little while after a reload  but after 
some timeouts incomming calls stop
-- 
Dennis Gilmore,  RHCE  
 http://www.ausil.us


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Re: [Asterisk-Users] IConnecthere dial out problems

[Dennis Gilmore]

On Wednesday 07 December 2005 11:10, John Voss wrote:
> I can't seem to get my outgoing connections to work with IConnecthere. At
> one time it did with v1.0
>
> I can register and receive calls just fine. But can't make them.
>
> Ultimately, the trace ends with a "400 Bad Request" error when you do a SIP
> debug.
>
> Has anyone got it to work with v1.2? Don't know if it is related to the
> version or not since I haven't worked with it in a while and didn't test
> outgoing before upgrading.
Im having no issues with my outgoing calls  but my incoming call registration 
keeps locking up.  it seems after awhile it stops sending reregistration 
packets.
my extentions.conf has 
; ** Dial Out iconnecthere ***
exten => _1NXXNXX,1,SetCallerID(ph number)
exten => _1NXXNXX,2,Dial(SIP/[EMAIL PROTECTED],60,r)
exten => _1NXXNXX,3,Congestion

exten => _NXX,1,SetCallerID("Dennis Gilmore" )
exten => _NXX,2,Dial(SIP/[EMAIL PROTECTED],60,r)
exten => _NXX,3,Congestion

exten => _61.,1,SetCallerID(ph number)
exten => _61.,2,Dial(SIP/[EMAIL PROTECTED],60,r)
exten => _61.,3,Congestion

in sip.conf
[iconnect]
; for routing calls outbound to the PSTN numbers via iconnecthere
; (aka deltathree)
type=friend
secret=
username=
CallerID="Dennis Gilmore "
authname=
host=natrelay.deltathree.com
nat=yes
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
allow=alaw
#allow=G726

Dennis
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[Asterisk-Users] DUNDi

[Dennis Boylan]

After reading all that I could find on DUNDi, I'd like to play with it, but
before I put it in production, would like to be able to see what DUNDi would
publish.  Is there a way to query DUNDi to see what it thinks it should
send?

- Dennis
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Re: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late

[Dennis Gilmore]

Once upon a time Sunday 20 November 2005 8:39 pm, Matt Riddell wrote:
> Eric Bishop wrote:
> > Hi All,
> >
> > I am running Asterisk (1.0.9.) on CentOS 4 with CDR recording being
> > output to MySQL. However whenever the system boots up after a reboot I
> > am needing to manually restart Asterisk because MySQL is after Asterisk
> > in the service startup sequence and I get
> >
> > ERROR[3367]: Failed to connect to mysql database cdr on localhost.
> >
> > Anyone know of a simple and elegant way to fix this?
> >
> > I'd prefer not to have to hack either MySQL or Asterisk init scripts
>
> If it's running using services, you could set MySQL to start on level 2 and
> Asterisk on level 3.
>
> chkconfig --list
umm.  you obviously dont understand how the different run levels work.   run 
level 2 has nothing to do with run level 3 the easiest way would be to put 
in /etc/rc.local 
/etc/init.d/asterisk restart   then asterisk will be restarted very last thing 
before you get a login prompt.   that is the only way to do it without 
changing the priorites in the init scripts  to make sure asterisk starts 
later.

though on my setup my init scripts are set to run asterisk almost last  and 
way after mysql  has started.

-- 
Dennis Gilmore,  RHCE  
 http://www.ausil.us


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Re: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late

[Dennis Gilmore]

Once upon a time Sunday 20 November 2005 10:38 pm, JP Carballo wrote:
> JP Carballo wrote:
> > Eric Bishop wrote:
> >> I have:
> >>
> >> [EMAIL PROTECTED] ~]# chkconfig --list | grep mysql
> >> mysqld  0:off   1:off   2:off   3:on4:off   5:off   6:off
> >> [EMAIL PROTECTED] ~]# chkconfig --list | grep asterisk
> >> asterisk0:off   1:off   2:on3:on4:on5:on6:off
> >>
> >> What would you suggest I do?
> >
> > 
> > 
> > Holy crap, this kind of replying is getting me dizzy! Up, down, what
> > next? Left and right?
> > Why can't we just agree to delete all previous text, anyway we all
> > have threaded readers...don't we?
> > 
> >
> > chkconfig --level 3 mysqld off
> > chkconfig --level 2 mysqld on
> > chkconfig --level 2 asterisk off
>
> I forgot to add that you should get this:
>
> ([EMAIL PROTECTED]:asterisk)# chkconfig --list | grep "asterisk\|mysqld"
> mysqld 0:off1:off2:on3:off4:off5:off
> 6:off
> asterisk   0:off1:off2:off3:on4:off5:off
> 6:off

ok a little back round on runlevels.  

Linux allows for up to 10 runlevels, 0-9, but usually only some of these are 
defined by default. Runlevel 0 is defined as ``system halt''. Runlevel 1 is 
defined as ``single user mode''. Runlevel 6 is defined as ``system reboot''. 
Other runlevels are dependent on how your particular distribution has defined 
them, and they vary significantly between distributions. Looking at the 
contents of /etc/inittab usually will give some hint what the predefined 
runlevels are and what they have been defined as.

ok so  when you turn mysqld off on run level 3 and thats what you system runs 
as mysqld  will never start. the services selected for that run level are ran 
when you enter that run level.

the order that they are run at is defined by a priority system.  so you need 
to make sure the priority of asterisk is such that is it started after 
mysqld.

on my system  mysqld  has a priority of 64 and asterisk is 99   look 
in /etc/rc3.d   the files starting with a S are for startup and K for 
shutdown.  they start with lowest number  up through highest number.  that 
last thing ran is /etc/rc.local  so you could always put in 
there /etc/init.d/asterisk restart  to make sure its the last thing done.


-- 
Dennis Gilmore,  RHCE  
 http://www.ausil.us


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Re: [Asterisk-Users] Aussie Call home!

[Dennis Gilmore]

Once upon a time Wednesday 26 October 2005 11:17 pm, James Sturges wrote:
> Hi,
>
> We have some people going to the US, namely LA & Las Vegas.
>
> And I would like to organize a local normal phone number for them to dial.
>
> No problem on the tech stuff, but don't know the local providers in the US
> of A!!
maybe look at broadvoice   being from oz i know all about trans continental 
setups
> The idea is the number would be with a VOIP SIP service which I would like
> to link to our Asterisk box over here in Australia.
>
> So then they can dial the number and get to our switch board.
>
> Also, is there a mobile provider we could use that would allow us to call
> that VOIP number as a reasonable rate.
Mobile providers here pretty much suck  I personnaly use tmobile as they are 
on a GSM network  so when i go home for hoildays i can take my phone.  they 
do offer prepaid services  so your guys could bring there phones and get sim 
cards here .  they would need to be Tri or Quad band phones  as they decided 
to use different a spectrum here.

> We are doing this with places in Australia, but was asking the group who
> would be the best US VOIP provider to use for these localities.
Cheers 

Dennis


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RE: [Asterisk-Users] wm_w DTMF solution for T1 tie line losing deigits.

[Dennis Walker]

I had a similar problem with a wink tie t1
try setting the emdigitwait=[ms]   in zapata.conf

on my system  I set  emdigitwait=600

--
From:   Steven[SMTP:[EMAIL PROTECTED]
Reply To:   Asterisk Users Mailing List - Non-Commercial Discussion
Sent:   Thursday, October 20, 2005 8:17 AM
To: asterisk-users@lists.digium.com
Subject:[Asterisk-Users] wm_w DTMF solution for T1 tie line losing 
deigits.

I assume the real fix is to alter some DTMF setting in my Panasonic DBS576, 
but I have yet to find it.

I was using a PRI card in my panasonic, but it broke, so I switched to a 
spare T1 card.
I set it up for em_w, but asterisk was dialing before it recieved all of the 
digits.

I saw a few suggestions in the WIKI and mailing list, but none worked as is.

The issue that complicated the exaples the most was the fact that sometimes 
I would recieve 1 digit and sometimes 4 or 6 etc.
If dialed fast enough, I would get the whole number in the fist pass.

[panasonic-catchall] is included last because it is the catchall for all non 
found numbers.
I am using this T1 for both 4 digit extension and as a trunk in the 
panasonic, so I do not have my 9 to route with.

exten => _X, is catching if only 1 digit is passed.
exten => _X., is catching if it is more than one.
exten => _X,5,GotoIf($["${Predigits1}" = ""]?s-gathermoredigits,1) ; this 
was the trick to make sure I didn't loop from the WaitExten() .

Here is the solution that I found that works 100% for me:


---
[panasonic]

include => ext-local
include => outbound-allroutes
; include => outrt-005-tollfree
; include => outrt-004-dial911
; include => outrt-003-dial9
; include => outrt-002-fwd
include => panasonic-catchall

[panasonic-catchall]

exten => _1X.,2,Dial(Zap/g0/${EXTEN},,r)
exten => _1X.,3,Congestion

exten => _X,1,NoOp( only got a few digit. It was ${EXTEN})
exten => _X,2,SetVar(Predigits1=${Predigits2})
exten => _X,3,SetVar(Predigits2=${EXTEN})
exten => _X,4,GotoIf($["${Predigits1}" = ""]?s-gathermoredigits,1)
exten => _X,5,NoOp(${TIMESTAMP} ok, now we're going to dial 
${Predigits1}${Predigits2}${EXTEN})
exten => _X,6,Dial(Zap/g0/${Predigits1}${Predigits2}${EXTEN},,r)
exten => _X,7,Congestion

exten => _X.,1,NoOp( only got a few digit. It was ${EXTEN})
exten => _X.,2,SetVar(Predigits1=${Predigits2})
exten => _X.,3,SetVar(Predigits2=${EXTEN})
exten => _X.,4,GotoIf($["${Predigits1}" = ""]?s-gathermoredigits,1)
exten => _X.,5,NoOp(${TIMESTAMP} ok, now we're going to dial 
${Predigits1}${Predigits2}${EXTEN})
exten => _X.,6,Dial(Zap/g0/${Predigits1}${Predigits2}${EXTEN},,r)
exten => _X.,7,Congestion

exten => t,1,NoOp( timed out dialing ${Predigits1}${Predigits2})
exten => t,2,Dial(Zap/g0/${Predigits1}${Predigits2},,r)
exten => t,3,Congestion

exten => s-gathermoredigits,1,NoOp( users have slow fingers - lets increase 
the DigitTimeout and try again)
exten => s-gathermoredigits,2,DigitTimeout,5; Increase the 'finished 
dialing' timeout to 5 seconds
exten => s-gathermoredigits,3,WaitExten(4)  ; and give the caller 8 
seconds overall to do their thing



-- 
-- 
Steven

May you have the peace and freedom that come from abandoning all hope of 
having a better past.
----  ---  - - -   -- -   -   --  - - - --- - --   - 
 - --- - - -- -  -- --   -   -- 



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RE: [Asterisk-Users] Asterisk and Mitel S X 200 Slip and Frame Err ors causing Major Ala rms

[Dennis Walker]

I'm not sure any more but; I think like 2400 per day. You can tell by just 
putting a number in it won't accept a value over it's limit. 

Unfortunately it isn't real high.

--
From:   Geoff Manning[SMTP:[EMAIL PROTECTED]
Reply To:   Asterisk Users Mailing List - Non-Commercial Discussion
Sent:   Tuesday, October 11, 2005 11:31 AM
To: '[EMAIL PROTECTED]'; 'Asterisk Users Mailing List - Non-Commercial 
Discussion'
Subject:RE: [Asterisk-Users] Asterisk and Mitel SX 200 Slip and Frame 
Err   ors causing Major Ala rms

Dennis Walker wrote:
> 
> But I did find that down in the t1 parameter settings you can set the
> limits higher.  I maxed them out and the problem went away, it would
> reset the count in a rolling 24 hours luckily the slip count just
> stayed below the limit.
> 

Do you by chance now what the max values are? The onsite tech set it to 1000
but we aren't sure that was an arbitrarily "high" number or the max value.

Thanks,
Geoff
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RE: [Asterisk-Users] Asterisk and Mitel S X 200 Slip and Frame Errors causing Major Ala rms

[Dennis Walker]

I use to do the same thing with an SX200 digital, and had the same problem.

I could not get the slips to go away no matter what I tried.

My sx200 would go offline once a day in the middle of the night luckily 
once the limit was exceded.

But I did find that down in the t1 parameter settings you can set the 
limits higher.  I maxed them out and the problem went away, it would reset 
the count in a rolling 24 hours luckily the slip count just stayed below 
the limit.

I think you get to them in the system config, ds0 settings.  The same place 
you set the t1 parameters.  I'm can't remember exactly the old SX200 has 
been go for 6 months now.

--
From:   Geoff Manning[SMTP:[EMAIL PROTECTED]
Reply To:   Asterisk Users Mailing List - Non-Commercial Discussion
Sent:   Monday, October 10, 2005 5:30 PM
To: Asterisk Users (E-mail)
Subject:[Asterisk-Users] Asterisk and Mitel SX 200 Slip and Frame 
Errors  causing Major Ala rms

We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get
over  500 frame errors and over a 500 slip errors per hour. When the errors
reach 1000 per hour the Mitel will take it's T1 card offline. At that point
no calls can be routed from the Asterisk server to the Mitel and the TE110P
reports a Yellow alarm.

What can be causing all these Frame and Slip errors? We have been working
with a Mitel tech to get all the configurations correct and we still 
haven't
been able to resolve the issue.

We are currently connecting via crossover so we'll try a straight through
just for kicks. We have a spare TE110P so we are going to try that. I just
don't know enough about these errors to know what to try next.  

Any other thoughts?



CONFIG INFO:

I am running "Asterisk CVS-v1-0-10/10/05-15:00:08".


# cat /etc/zaptel.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order

# Global data

loadzone= us
defaultzone = us

#span =, ,,,coding>
# timing values can be
# 0 - not used as timing source
# 1 - primary timing source
# 2 - Secondary timing source

span=1,1,0,d4,ami
e&m=1-24


===


# cat /etc/asterisk/zapata.conf
;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]
;
; Trunk groups are used for NFAS or GR-303 connections.
;
; Group: Defines a trunk group.
;group => ,[,...]
;
;trunkgroup  is the numerical trunk group to create
;dchannelis the zap channel which will have the
;d-channel for the trunk.
;backup1 is an optional list of backup d-channels.
;
;trunkgroup => 1,24,48
;
; Spanmap: Associates a span with a trunk group
;spanmap => ,[,]
;
;zapspan is the zap span number to associate
;trunkgroup  is the trunkgroup (specified above) for the mapping
;logicalspan is the logical span number within the trunk group to
use.
;if unspecified, no logical span number is used.
;
;spanmap => 1,1,1
;spanmap => 2,1,2
;spanmap => 3,1,3
;spanmap => 4,1,4

[channels]
;
; Default language
;
;language=en

musiconhold=default

busydetect=1
busycount=7

relaxdtmf=yes
callwaiting=yes
;
; Support Caller*ID on Call Waiting
;
callwaitingcallerid=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; Support flash-hook call transfer (requires three way calling)
;
transfer=yes
;
; Support call forward variable
;
cancallforward=yes
;
; DR: Commented out temporarily
;echocancel=yes
;
; Generally, it is not necessary (and in fact undesirable) to echo cancel
; when the circuit path is entirely TDM.  You may, however, reverse this
; behavior by enabling the echo cancel during pure TDM bridging below.
;
;echocancelwhenbridged=yes

; Logical groups can be assigned to allow outgoing rollover.  Groups
; range from 0 to 31, and multiple groups can be specified.
;
;group=1
; Ring groups (a.k.a. call groups) and pickup groups.  If a phone is 
ringing
; and it is a member of a group which is one of your pickup groups, then
; you can answer it by picking up and dialing *8#.  For simple offices, 
just
; make these both the same
;
callgroup=1
pickupgroup=1

;
; Specify whether the channel should be answered immediately or
; if the simple switch should provide dialtone, read digits, etc.
;
immediate=yes
; For fax detection, uncomment one of the following lines.  The default is
*OFF*
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no


rxgain=0
txgain=0

signalling=em
wink=200
debounce=100
flash=300

;context=zap-incoming
context=gv-incoming
group = 1
channel => 1-17


signalling=em_w
context=zap-incoming
group = 2
channel => 21-24
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RE: [Asterisk-Users] sip register incoming call contexts?

[Dennis Walker]

Ok :)

--
From:   Rich Adamson[SMTP:[EMAIL PROTECTED]
Reply To:   Asterisk Users Mailing List - Non-Commercial Discussion
Sent:   Monday, October 10, 2005 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:Re: [Asterisk-Users] sip register incoming call contexts?


> Sorry this is a bit of a newbie question, I've been at this for a few
> months and still have not quite figured this one out.
> 
> 
> I've been able to setup one itsp (incoming calls) (sip account) with a
> register line like this:
> 
> register => nnn:[EMAIL PROTECTED]
> 
> -or-
> 
> register => nnn:[EMAIL PROTECTED]/nnn
> to come directly into an extension in the dialplan
> 
> 
> It seems that this only works with the default context in the dialplan.
> 
> 
> I have another sip account from another provider that I would like
> all of it's incoming calls to come into the s, extension of
> a new context but I have been unable to figure out
> how to bring calls from a register line into an alternate context.
> 
> It seems that register lines are limited to only being used in the
> general section of sip.conf and you are limited to one context=
> statement there.
> 
> Is there a way to register a second account and have it's calls come into
> another context in the dialplan?
> 
> register lines only seem to work in [general] and it seems like you
> are limited to only one inbound context here.
> 
> I would like the two inbound call accounts to be 'isolated' from each other
> and not have to come in on the same incoming context in the dialplan.
> 
> I'd also like to be able to have them have their own contexts with thier
> own s, (start) extension available.

Try using something like:
 deny=0.0.0.0/0.0.0.0  
 permit=147.135.8.129/255.255.255.0 
 permit=147.135.0.129/255.255.255.0
 permit=147.135.4.128/255.255.255.0

in each sip.conf itsp definition to limit which contexts will match.
Obviously, replace the above permit's IP addresses with the correct
ones for your provider.


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Re: [Asterisk-Users] sip phones on x86_64

[Dennis Gilmore]

Once upon a time Monday 03 October 2005 3:11 pm, Wayne Gemmell wrote:
> Hi all
>
> Can anyone recommend a good soft phone that can compile on x86_64 (linux)
> platform?
kphone compiles and is available in Fedora extras  and im sure is available 
for other distros.  If you want to get adventurous you could try cvs 
gnomemeeting.  it also has sip support.

Dennsi


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Re: [Asterisk-Users] Need quote for Asterisk and billing remote install

[Dennis Gilmore]

Once upon a time Sunday 28 August 2005 11:13 pm, Matt Riddell wrote:
> Dennis Gilmore wrote:
> >>But then, you listed your minimum price as $1.  What do I get for this?
> >>
> >  :D A one minute phone conversation.  :D  maybe a bag of chips.
>
> Is delivery included?  If so I'll have 3.  Send them to Dunedin, New
> Zealand.
>
> hehe
No Delivery.  you need to go down to the conor store and get some.  Ill have 
some salt and vinegar while your there.  


-- 
Dennis Gilmore  RHCE  
 http://www.ausil.us


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Re: [Asterisk-Users] Need quote for Asterisk and billing remote install

[Dennis Gilmore]

Once upon a time Sunday 28 August 2005 10:33 pm, Matt Riddell wrote:
> Paul wrote:
> > Based on the fine detail you provided my estimate is somewhere between 1
> > and 10 thousand US dollars.
>
> A max of 10K for any job?  /me thinks you're selling yourself short.
>
> Just imagine if you got the job and it required 2000 hours.
>
> At $100 per hour that would be $200,000
>
> Even at $50 it would still be 10 times more than your max!
>
> :D
>
> But then, you listed your minimum price as $1.  What do I get for this?
>
> :D

 :D A one minute phone conversation.  :D  maybe a bag of chips.

-- 
Dennis Gilmore  RHCE  
 http://www.ausil.us


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Re: [Asterisk-Users] Fedora Core 4 x86_64

[Dennis Gilmore]

Once upon a time Friday 26 August 2005 12:30 pm, Brian C. Fertig wrote:
> Take it from someone who owns 25 of them.  Stay away from FC anything.
>
> Use CentOS 4 its better more stable and has true multi-treading as FC
> doesn't thread anything..
>
What do you mean by FC doesnt thread anything?i have threaded applications 
usint NPTL  with over 1000 threads.  I really dont see how you can say FC is 
not stable  i have servers running FC  that have been up over 150 days.  the 
main issue i have has is the out of memory killer   going a little wild.  but 
i have had the same issue on some of my centos boxes.

I think they are both mostly as stable as the other.  tough CentOS base of 
RHEL  gives you a promise that the OS will be supported for a longer period 
of time.  This should mean  that you can run a CentOS based server for at 
least 5  years  without upgrading the server.

-- 
Dennis Gilmore  RHCE  
 http://www.ausil.us


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Re: [Asterisk-Users] grandstream bt100 help

[Dennis Gilmore]

Once upon a time Monday 22 August 2005 4:10 pm, Bartosz Jozwiak wrote:
> Hi Guys,
>
> Sorry about writing to that list but could not find better place.
> I have Grandstream BT-100 phone, btw, was working great with Asterisk.
> I have upgraded the phone, and during upgrade something went wrong.
> Right now when I power the phone I can only see some garbage on the LCD
> display. does not react on any buttons, pings,.
> Maybe somebody has any idea if it is fixable or I can just say "goodbye" to
> my phone ?
>
> Thank you in advance,
> Bartosz
I had mine messed up once.  i have static ip config in it  i connected to my 
laptop via a crossover cable.  set ip to my router  ran ethereal to see what 
it was doing.  once i worked out it was trying to tftp  a new image  i 
created a sub interface  with the ip it wanted  setup ipforwarding  and tftp 
server with a new image.  i could then upload the image.  main problem in my 
case  was that iconnecthere  turned there tftp server off.

-- 
Dennis Gilmore  RHCE  
 http://www.ausil.us


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Re: [Asterisk-Users] SPA-2100 Analog Telephone Adapter

[Dennis Gilmore]

Sean Rima wrote:

Does anyone have any experience of these, I have been offered one and am
thinking of adding sticking it onto the back of my Asterisk box and just
ignore the WAN port if possible, It would be to stick my exisiting
phones onto the asterisk box

Sean


I just bought 12 of them to link 5 offices PBX systems together.  so far 
in my testing they work extremmly well with asterisk.  you will want to 
modify the dial plan on it  otherwise  you will get a delay when calling 
extentions.


Dennis
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Re: [Asterisk-Users] To anyone seeking 911 Service Providers "stay away!!!"

[Dennis Gilmore]

Once upon a time Tuesday 26 July 2005 8:00 am, Julio Arruda wrote:

> >>Implement 1 single hard wire for this service and cover your tush!
> >
> > This is what I tell my customers as well, not because I can't do it but
> > because they typically have the line there anyway for fax and/or security
> > which carries the DSL circuit for VOIP.  :-)
>
>  From what I understand, the big deal is with cable providers and naked
> DSL providers, where you don't really have a 'landline' tied to your DSL  ?
> Would be interesting to people involved in asterisk biz. to have a
> summary of countries and their regulations on these issues, anyone know
> if a summary like this is posted somewhere ?
Well i have cable and dont have a landline,  and refuse to get one at $45 a 
month  the cheapest option and $65 a month the most expensive.  i have 
Asterisk with 2 external DID one a US based one and one is an Australian one.   
Yes i know i dont have 911 service   thats what my cell phone is for.  the 
only time it would be an issue is with guests. 

Dennis
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Re: [Asterisk-Users] iconnecthere

[Dennis Gilmore]

Once upon a time Wednesday 20 July 2005 10:42 pm, Rich Adamson wrote:
>
> Kind of sounds like the real question is "who is suppose to provide
> ringback when you are using asterisk"?
>
> My guess is that you are. Try "show application dial" and look for
> the "r" parameter, etc.

I guess so.  Works  now thank you.  teach me to not research more first
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[Asterisk-Users] iconnecthere

[Dennis Gilmore]

Hi All,

I have service with iconnecthere  they use sip and allow generic devices.  
outbound and inbound calls work as expected.  i had a few hicups  but nothing 
i couldnt workout with sip debugging turned on.  

I have one issue and its a doozie.  I dont think its related to asterisk  
though they claim otherwise.  

When you call my number  once the number is dialled  you hear nothing on the 
phone  until either the call is answered  or it goes to voicemail.there 
is no indication that the call is working at all.  

im at a loss as to what it would be.  i have forwarded the rtp ports and sip 
port to my asterisk box.  i even tried setting the box in the DMZ.  I dont 
want to waste  much time on this  as there customer service sucks.  the 
product itself  has always been good.  

i cant think of any configs to post  that will help  as it really does seem ok 
and does work.  

Dennis
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RE: [Asterisk-Users] connecting Asterisk to NEC NEAX system

[Dennis Walker]

in zapata.conf

use emdigitwait=###   number of milliseconds to wait for digits to be output

I had similar problems with MITEL system and had

emdigitwait=500



--
From:   Edwin Lam[SMTP:[EMAIL PROTECTED]
Reply To:   Asterisk Users Mailing List - Non-Commercial Discussion
Sent:   Tuesday, June 07, 2005 6:40 PM
To: asterisk-users@lists.digium.com
Subject:[Asterisk-Users] connecting Asterisk to NEC NEAX system

hi. i connected Asterisk to an NEC NEAX system with a crossover T1 cable
and the Digium TE405P using E&M wink signaling. the connection's ok. however
when dialing from the NEC to the Asterisk. most of the time the Asterisk only
sees the first digit of the dialed number(which is 4 digits). some time if i
dialed the 4 digits very fast it might get through. seems like there's a timming
issue of the DTMF. what can i do to solve this? i looked through the docs
for zaptel.conf & zapata.conf and doesn't seems there's any parameters to
control the DTMF timing.

-- 
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
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RE: [Asterisk-Users] Features.conf - atxfer

[Dennis Walker]

On superviced you cancel on a zap channel you can cancel the transfer by
a hook flash this will send you back to the original caller.

On sip phones you hit the cancel button or if you have line buttons you just
pick the original callers line.

--
From:   Mike Holloway[SMTP:[EMAIL PROTECTED]
Reply To:   Asterisk Users Mailing List - Non-Commercial Discussion
Sent:   Tuesday, June 07, 2005 7:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:Re: [Asterisk-Users] Features.conf - atxfer


Reading through the code, I don't see a way of exiting the transfer and 
regaining the call with the customer, unless the third party hangs up or 
maybe doesn't answer and the dialplan doesn't do anything else with the 
call (send the call into voicemail).

I suggest you request this feature (http://bugs.digium.com), but as an 
interim solution you can create a dialplan for internal extensions that 
does not send the call to voicemail if unanswered, and only dials the 
third party for a limited amount of time (20 seconds?).

You could preface these special extensions with a sequence, such as 9, 
or 777 or whatever. Assuming your extensions are 1xx:

exten => _7771XX,1,Dial(SIP/${EXTEN:3},20)
exten => _7771XX,2,Hangup

-mike


Mark Johnson wrote:
> I am trying out the new atxfer feature from CVS-HEAD.  I set atxfer 
> equal to *7 and it seems to work OK.  I am having a problem getting it 
> to work the way a receptionist would want.  If an extension calls me, I 
> hit *7 and I hear the voice say "transfer".  I dial another extension.  
> If the newly dialed extension goes to voicemail, I can't figure out how 
> to get the original call back to tell them the person they are trying to 
> reach is unavailable.  Anything I try bridges the call and the caller go 
> into like the 2nd half of the voicemail greeting.  Is there some trick 
> to this?
> 
> Thanks!
> 
> Mark
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