[asterisk-users] End of call

[Michael Strelnikov]

Is there any way to know that call is finished? I know there are special tones sent by phone companies but how can I detect them and then configure Asterisk to use it?Thanks,Michael
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Re: [asterisk-users] Asterisk 1.2.11 and # key

[Michael Strelnikov]

But the behaviour should like: if pressed once the # should be transmitted. If pressed ## (fast) the it should be blind transfer. Isn't it?On 9/5/06, David Gagnon
 <[EMAIL PROTECTED]> wrote:














That why, when you dial
one # then Asterisk wait to see if you dial two of them. You should consider
changing the blindxfer function or play with the timer in the features.conf. In
think its look like featuresdigittimeut.

 

For the moment, if you
dial #( wait 1 sec) then press it again, the second will work.

 

David









De :

[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] De la part de
 Michael Strelnikov
Envoyé : 4 septembre 2006
22:47
À : Asterisk
 Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users]
Asterisk 1.2.11 and # key



 

I have "blindxfer
=> ##" line in my features.conf



On 9/5/06, David Gagnon
<[EMAIL PROTECTED]> wrote: 







Are you sure this is not because of the dynamic
features in features.conf ?

By default, # is defined for the transfer
feature.

 

David

 









De :
 [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
De la part de Michael Strelnikov
Envoyé : 4 septembre 2006
09:53
À : asterisk-users@lists.digium.com
Objet : [asterisk-users]
Asterisk 1.2.11 and # key







 

Hello,

   Does anybody have problems with recognition of the hash (#) key
with * 1.2.11? It seams that after pressing # the call is in a progress but no
data is sent.

Thanks in advance,
Michael










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Re: [asterisk-users] includes in realtime ??

[RR]

Ben,

The family name is not sipuser, its sipusers. So try this command

"realtime load sipusers name " and see if you get nothing. What about?

realtime load sipusers username  ?

To answer your question, any change in the tables holding this sip
users information comes into affect immediately. That's the whole
point of realtime :)

Cheers,
\R
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[asterisk-users] Re: [asterisk-biz] Re: G729 Replacement Codec - FREE or may ne cheaper than existing one.

[Kannaiyan Natesan]

Digium really did make an effort in this case and it's not worth

   I appreciate Digium and Mark. They have released the
G729 and G731 source code and you need to pay the license directly to
voiceage, not directly to digium. Also with the code of asterisk,
there are lot of authors put in their code into asterisk and endedup
in nothing. (See Asterisk Disclaimer Policy)


in another area instead. History suggests this will come back to you in some

History suggests, the authors of the source need to
be given with proper rights but not renamed with Mark Spencer.

G729 is developed by Voiceage or related group. The fees can directly
go to them not swallowed by middle commission agents. There is no harm
in paying to voiceage and everyone must do the same in protecting
their intelligence in that.

The step which Digum recently took is really admiring in releasing the
source code of G729 and pay the royalty fees to voiceage.

Don't waste anyone's time in discussing further on this issue, since
Digum is moving to a different strategy with regards to the G729
licenses.

Regards,
Kannaiyan


On 9/5/06, Justin Newman <[EMAIL PROTECTED]> wrote:

Illegal or not, the license charge is so small it isn't worth the risk.
Digium really did make an effort in this case and it's not worth
reproducing. If you can, buy their licenses and spend your time contributing
in another area instead. History suggests this will come back to you in some
way.

Justin

- Original Message -
From: "Andrew Joakimsen" <[EMAIL PROTECTED]>
To: "Justin Newman" <[EMAIL PROTECTED]>; "Commercial and
Business-Oriented Asterisk Discussion" 
Sent: Monday, September 04, 2006 5:37 PM
Subject: Re: [asterisk-biz] Re: G729 Replacement Codec - FREE or may ne
cheaper than existing one.


> So even if we license from Intel the code, it is illegal to use it with
> Asterisk because Asterisk is GPL? I still don't get that part
>
> On 9/4/06, Justin Newman <[EMAIL PROTECTED]> wrote:
>>
>> Kannaiyan,
>>
>> It may be helpful to read...
>>
>>
>> http://lists.digium.com/pipermail/asterisk-users/2004-September/057110.html
>>
>> http://www.voip-info.org/wiki-Asterisk+G.729+Licensing
>>
>> Justin
>>
>> --
>> Date: Sat, 2 Sep 2006 16:21:40 +0800
>> From: "Kannaiyan Natesan" <[EMAIL PROTECTED]>
>>
>> Hi,
>> I heard of a news, that there is a replacement codec available for
>> g729 and accept the g729 codec data for decoding. Anyone familier with
>> this? Also the good news is that it is noted that it works fine with
>> asterisk and the g729 encoded data.
>> Anyone has the link for the free asterisk distribution which can
>> take unilimited channels of g729 codec, data. If there is any royalty
>> need to pay, is that cheaper than the existing g729 cost?.
>> I read somewhere in the net and forgot to save the url and google
>> did not brought me that again.
>>
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>>
>

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[asterisk-users] Reading the raw E1 channels ?

[Azher Amin]

Hi there,

With the help of digium E1 card, is there any possibility / solution to 
tap into an E1 circuit (while sitting in a telco house) ??


Plz provide some guidance / external links.

Regards
Azher

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RE: [asterisk-users] Blind transfer 3/4 digits

[Koopmann, Jan-Peter]

On Tuesday, September 05, 2006 2:06 AM Ronald Wiplinger wrote:

> In my opinion Asterisk remembers all numbers and therefore it does
> not wait for the 4, since it found a match. This is in VoIP (in my

If both phones enter the dialplan the same way and one phone does work then it 
should not be a problem with the dialplan or with the way Asterisk is doing the 
match. You pointed that out yourself. AFAIK there is no overlapping in the 
dialplan. Either the phone (when dialing, doing a SIP transfer etc.) or 
Asterisk (when doing an attended/unattended transfer) is waiting the specified 
time for more digits. If no other number is received it then feeds the received 
number in the dialplan. 

So either your phone is just transmitting 601, Asterisk only understands 601 or 
you do have a problem with your dialplan. The only other option would be a 
general dialplan bug which is not too likely since most of us would have run 
into the exact same problem.

What do debugs on the Asterisk show you? Do a SIP debug etc. Are you using 
inband or outband DTMF? 

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RE: [asterisk-users] Blind transfer 3/4 digits

[Koopmann, Jan-Peter]

On Monday, September 04, 2006 3:22 PM Ronald Wiplinger wrote:

> What's happen to you guys? 

Nothing. Why?

> I am not yelling, just asking.

Maybe in a bit stressed out kind of way.

> It is sure not a dialplan question! 

Without having all necessary information that is hard to say. Maybe one phone 
comes in a different context than the other etc. Lot's of things that could go 
wrong in the dialplan.

> If it would be a dialplan
> question, than it would be for each dialing, but it isn't. 

If we are talking about the same context and same way of dialing: True. 

> You mentioned SIP message and that makes me wonder! Are we not using
> here dtmf ?? 

I somehow had the impression that you are using the transfer button on the SNOM 
which would tell the SNOM to transfer the call. You are obviously talking about 
attended/unattended transfer via Asterisk only, correct? Then ignore my 
suggestion.

> If it is a sequence of "tones", 

Well... If you are using inband DTMF: correct. Otherwise DTMF may correspond to 
SIP messages as well but let's not get into that. I suppose you are using 
inband DTMF and G.711?

> than why is it different if it is in
> a string (like snom) or another phone, with single tones? 

If the dialplan is not responsible obviously the phones are behaving 
differently. Maybe the DTMF sequence is not transmitted correctly but on the 
other hand I am using SNOMs with inband DTMF without any problems. Maybe the 
phone (as others suggested) is doing some number/pattern matching magic which 
you have to fiddle with.

> If we understand this part, than is the question, where can I turn on
> the system to take a longer break between "tones" still as a string? 

The default setup should not be a problem with SNOMs (at least I never read 
anything about it) but have a look at the features.conf options.

> That should proof my thoughts (and that
> without yelling, ... hehehehe)

But a lot of exclamationsmarks. :-) Just kidding.

As others pointed out: We (at least I) would need the entire picture (the 
relevant parts of your dialplan etc.) to really help you here.

Regards,
  JP
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[asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware

[Tomislav Parčina]

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> > Where did you find 8.0.3 SIP image?
> 
> Cisco website...

I didn't noticed 8.0.3 SIP firmware there... 

> Just tried now with the 8.0.2.SR1 image...
> 
> Keeps on saying "registering"

Have you tried the one on the end - "Another SEP.xml.cnf example"


--
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] Asterisk 1.2.11 and # key

[David Gagnon]

That why, when you dial
one # then Asterisk wait to see if you dial two of them. You should consider
changing the blindxfer function or play with the timer in the features.conf. In
think its look like featuresdigittimeut.

 

For the moment, if you
dial #( wait 1 sec) then press it again, the second will work.

 

David









De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Michael Strelnikov
Envoyé : 4 septembre 2006
22:47
À : Asterisk
 Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users]
Asterisk 1.2.11 and # key



 

I have "blindxfer
=> ##" line in my features.conf



On 9/5/06, David Gagnon
<[EMAIL PROTECTED]> wrote: 







Are you sure this is not because of the dynamic
features in features.conf ?

By default, # is defined for the transfer
feature.

 

David

 









De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
De la part de Michael Strelnikov
Envoyé : 4 septembre 2006
09:53
À : asterisk-users@lists.digium.com
Objet : [asterisk-users]
Asterisk 1.2.11 and # key







 

Hello,

   Does anybody have problems with recognition of the hash (#) key
with * 1.2.11? It seams that after pressing # the call is in a progress but no
data is sent.

Thanks in advance,
Michael










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[asterisk-users] HITBSecConf2006 Final Call !

[Praburaajan]

Hello everybody HITBSecConf2006 - Malaysia is only 13 days away and we
will be having loads of speakers down to give talks and presentations on
highly interesting topics, so why don't you register now @
http://conference.hitb.org/hitbsecconf2006kl. Come and experience Asia's
 Largest Security Conference !.

Date : 18th - 21st September 2006
Venue : The Westin, Kuala Lumpur

Keynote Speakers :

Bruce Schneir
CTO, Counterpane Internet Security

Presentation Title : Schneir On Security

Always interesting and entertaining, Bruce Schneier will talk about
current topics in security, economics, and society.

About Bruce Schneier:

Internationally-renowned security technologist and author Bruce Schneier
is both a Founder and the Chief Technical Officer of Counterpane
Internet Security, Inc. the world’s leading protector of networked
information - the inventor of outsourced security monitoring and the
foremost authority on effective mitigation of emerging IT threats.

--

Mark Curphey
VP, Foundstone Professional Services - A Division of McAfee Inc.

Presentation Title : What application security tools vendors don’t want
you to know and holes they will never find!

About Mark Curphey:

Mark founded OWASP, the Open Web Application Security Project that has
become a well thought of reference site for developers and system
architects and recommended reading by the US Federal Trade Committee. He
has a Masters Degree in Information Security from the renowned Royal
Holloway, University of London where he specialized in advanced
cryptography. Mark is a Microsoft MVP for developer security.

--

John Viega
Chief Security Architect, McAfee Inc.

Presentation : hat application security tools vendors don’t want you to
know and holes they will never find! *With Mark Curphey*

About John Viega

John is the co-author of three books on application security, Building
Secure Software (Addison Wesley, 2001), Network Security with OpenSSL
(O’Reilly, 2002) and the Secure Programming Cookbook (O’Reilly, 2003).
He also built the CLASP application security process, which is available
on-line.

--

The Other Speakers we have in store for you are :

1.) Anthony Zboralski
2.) Arnaud Ebalard
3.) Carlos Sarraute
4.) Ching Tim Meng
5.) Dave Tamasi
6.) Douglas MacIver
7.) Fabio Ghioni
8.) Fabrice Marie
9.) Fyodor Yarochkin
10.) Javier Burroni
11.) Jim Geovedi
12.) Joanna Rutkowska
13.) Jonathan Limbo
14.) Lisa Thalheim
15.) Marc Schonefeld
16.) Meder Kydyraliev
17.) Michael Davis
18.) Nguyen Anh Quynh
19.) Nish Bhalla
20.) Paul Boehm
21.) Philippe Biondi
22.) Raditya Iryandi
23.) Raoul Chiesa
24.) Roberto Preatoni
25.) Rohyt Belani
26.) Saumil Shah
27.) Shreeraj Shah
28.) Dr. Stefania Ducci
29.) Thorsten Holz
30.) The Grugq
31.) Van Hauser
32.) Wes Brown
33.) Yen Ming Chen

--

Security Trainings

TECH TRAINING 1 - Advanced Web Application & Services Hacking
TECH TRAINING 2 - Attacking & Defending Networks (Advanced Linux Edition)
TECH TRAINING 3 - The Exploit Laboratory
TECH TRAINING 4 - Tactical VoIP : Applied VoIPhreaking
TECH TRAINING 5 - War Driving .Gov
TECH TRAINING 6 - Structured Network Threat Analysis and Forensics


Register Now @ http://conference.hitb.org/hitbsecconf2006kl/register.php

*Walk In Registrations are accepted at the venue*

We hope to see you in 2 weeks @ HITBSecConf2006 !










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RE: [asterisk-users] Blind transfer 3/4 digits

[David Gagnon]

Ronald,

If I understand well, the second phones have a building digit map.
It has nothing to do with Asterisk! Asterisk only executes what it receives
from the phone.

## In your scenario is the Asterisk built-in transfer function? The
ways your phone sends the DTMF depend on what you have configured in you
SIP.conf.

http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode

 David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 4 septembre 2006 20:06
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits

David Gagnon wrote:
> Ronald,
>
>   Like someone already told you, you should explain more clearly the
> way you try to transfer, we need more details on the procedure, using
which
> button on which phone. We need every detail to help you. This as nothing
to
> do with the way the dial plan is loaded, this is totally false.
>
>   I'm sure most of the people here don't understand how you try to
> transfer.
>
>   David
>
>   

David,

I am not sure how the explanation how to punch the keys changes 
something,  ;-.)

Ok, here we go:

Snom:
pick up the phone and hit ##6014 followed by [ok]

Noname:
pick up the phone and hit ##6014 must be pushed very fast!!! No end 
# needed, since the phone 601 starts to ring as soon I reach 1.

In my opinion Asterisk remembers all numbers and therefore it does not 
wait for the 4, since it found a match. This is in VoIP (in my opinion) 
wrong, since overlapping numbers are allowed.

Sip message / dtmf, this is something different! How is the transfer 
made? Maybe snom does send a sip message, while the noname only send 
dtmf tones.

bye

Ronald

> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de Ronald
> Wiplinger
> Envoyé : 4 septembre 2006 09:22
> À : Asterisk Users Mailing List - Non-Commercial Discussion
> Objet : Re: [asterisk-users] Blind transfer 3/4 digits
>
> Koopmann, Jan-Peter wrote:
>   
>> On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:
>>
>>   
>> 
>>> try that way. However, I have doubts as well. If you are right, than
>>> why snom phone does not have this problem? Would not here also the
>>> first match count?   
>>> 
>>>   
>> Because the transfer button on the SNOM is using a totally different
>> 
> mechanism than sending # to Asterisk. On your snom configuration (like
ours)
> the phone does not start to create/send a SIP message until you hit "OK".
At
> that time the entire number is there and a complete SIP transfer is
created.
> Cool down a bit. The problem you are having is most probably just a
dialplan
> problem. It takes some time and experience to get those things right. No
> need to yell here...
>   
>>   
>> 
> What's happen to you guys? I am not yelling, just asking.
> It is sure not a dialplan question! If it would be a dialplan question, 
> than it would be for each dialing, but it isn't.
>
> You mentioned SIP message and that makes me wonder! Are we not using 
> here dtmf ?? that is in my opinion not a sip message, isn't it?
> If it is a sequence of "tones", than why is it different if it is in a 
> string (like snom) or another phone, with single tones?
> If we understand this part, than is the question, where can I turn on 
> the system to take a longer break between "tones" still as a string?
>
> Back to the dialplan:
> A Voip number can have different length of digits. Each number is seen 
> as a complete "picture", and so a three digit and a four digit number is 
> something different. While in the legacy telephony the digits are worked 
> down one by one and if there is no more use of the digits, they are just 
> garbage and will be not used. Unlike in VoIP, where you can have a three 
> digit number and if you dial four digit, than it is a WRONG number  
> I just verified that: I dialed from 601 to  61522, however, 61522 does 
> not exist, but 615 exists. Guess what? I get a busy tone! That should 
> proof my thoughts (and that without yelling, ... hehehehe)
>
> bye
>
> Ronald
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>
>
> ---
> avast! Antivirus: Inbound message clean.
> Virus Database (VPS): 0635-4, 2006/09/01
> Tested on: 2006/9/4 ¤U¤È 11:40:24
> avast! - copyright (c) 1988-2006 ALWIL Software.
> http://www.avast.com
>
>
>
>
>   


-- 
Ronald Wiplinger  (CEO of ELMIT)
http://www.elmit.com  http://voip.elmit.com  http://e-paper.elmi

Re: [asterisk-users] includes in realtime ??

[Benjamin Jacob]

Rushowr wrote:


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Benjamin Jacob

Sent: Monday, September 04, 2006 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] includes in realtime ??

Hello ppl,
Is it possible to include contexts in the RealTime scenario??
If not, wots the work around??

Thanks in advance.
Ben.
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Amazing how the wiki has this vast amount of AT LEAST info to start your
research on
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions


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Sorry mate.
Just slipped the eye.

Now to another question, which I tried about.
With the Realtime arch, can we change parameters of certain users, say 
sipusers, at runtime, for e.g. the codec and the change being reflected 
back immediately?


The two SIP users I had, had allow set to "gsm;g729;ulaw;alaw", and the 
two Xlite phones have gsm,ulaw and alaw configured.Calls work fine .


I changed the codec(set allow to have only g729).  But still the calls 
go thru.


I tried realtime load sipuser name , to no effect. (anyway, 
realtime load is only for reading values, if i am not wrong).


So is it possible to change user parameters at realtime?
or am I missing something again?

Thanks again.
Ben.
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[asterisk-users] Warning about using PAP2-NA ATA recent firmware 3.1.12 LS

[joy panlilio]

Fellow List,  Hi! Good day, I have cross a configuration nightmare for a couple of days of finding what really broke my setup, i have given a task on replacing our legacy pbx with asterisk.The problem i have encounter during the transition from legacy to IP based PBX is DTMF detection not working realiably on the following scenarioFailed ScenarioPAP2-NA - SIP -> TDM400 -> PSTNSuccessfull ScenarioPAP2-NA -> SIP -> FWD -> -PSTNPAP2-NA -> SIP -> SIP Carrier -> PSTNMy PAP2-NA Firmware is 3.1.12 LS   When i test this with siemens optipoint 410 it was clear that DTMF issues might be on PAP2 Since it really wrecks on the other endSolution:  After googling for a day i found no real answer for this problem when i talk to linksys tech support when i say i'm using my Asterisk as my SIP GW, they say that PAP2 wasn't meant to be compatible to asterisk or might
 be an Asterisk bugs. What i did is banging the phone and back to my web browser and continue my journey. this time i finally decided to play around with PAP2-NA firmware  I search on google if somebody has an OLD firmware hoping this will be fix.I found one 3.1.3 LS firmware which this is the default firmware upon buying this stuff.To my surprise it works like a charm after successfully downgrading the firmware from 3.1.12 LS to 3.1.3 LS  To all of you guys out there PLEASE DON'T UPGRADE YOUR FIRMWARE Linksys  might intentionally do this on purpose ^__^ or this might be *BUG* i doubt it.Best Regards,Joy 
	
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[asterisk-users] Re: FAX handling

[Justin Newman]

Let me know if you guys need help with this...

Justin

--

Message: 15
Date: Mon, 4 Sep 2006 17:16:00 -0400
From: "Technical Support" <[EMAIL PROTECTED]>
Subject: RE: [asterisk-users] FAX handling
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"

Look into NVDETECT, and fax2mail script on www.generationd.com

Fax detection is automatic

MD 
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[asterisk-users] Codec Thread

[Joe shmoe]

Well you can call me a newb all you want..  The
software was released to me by a birdie from digium. 
This is just the source code.  Nothing more.  You
still need the license for the g729 or g723 but this
code from digium will allow you to test.

You still need to purchase your license remember that.
 

/spy

PS.  Be kind.  Digg.com this
http://www.digg.com/software/g723_and_g729_codecs_source_for_asterisk_leaked


--- brandon kruz <[EMAIL PROTECTED]> wrote:

> haha
> im reporting this
> but i think your a nub
> and what happend was you just took the actual files
> so it wasnt hard, you just dont have a physical
> liscense
> there are things all over the web on how to enable
> g729
> its not hard
> its the liscensing that can get you sued for a very
> much amount of money...
> 
> 
> you can get any software you want and illegally
> install it
> using that in an enterprise envirnment, thats just
> stupid, and you will get 
> caught.
> 
> 
> of course if this is in fact legit, im very very
> sorry, im a nub
> but from the sound of it, it sounds illegal, you
> will get caught
> 
> fess up
> get a job
> a pay for business edition
> 
> my 2 cents
> `KruZ~
> 
> please someone let me know if this is in fact legit,
> else i am now 
> researching this, and it will be stopped
> 
> please dont do anything illegal and stupid
> it doesnt make you any more 1337 than you can find
> ways to
> enable these codecs without paying for them
> 
> its your chance to get sued.
> go for it.
> 
> 
> 
> >From: "Kannaiyan Natesan" <[EMAIL PROTECTED]>
> >Reply-To: Asterisk Users Mailing List -
> Non-Commercial 
> >Discussion
> >To: "Asterisk Users Mailing List - Non-Commercial 
> >Discussion",
> "Commercial and 
> >Business-Oriented Asterisk
> Discussion", 
> >"Asterisk Developers Mailing List"
> 
> >Subject: Re: [asterisk-users] Digum g729 and g723
> >Date: Tue, 5 Sep 2006 10:47:37 +0800
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> >
> >Hey,
> >
> >Is this code released by Digium?
> >Looks like directly from digium. Is it GPL with
> License and Royalty?
> >
> >Unlimited channels and no restriction
> !
> >
> >Author mentioned as Mark Spencer.
> >If we want to pay the license fees, should we have
> to Pay to VoiceAge 
> >directly?
> >
> >Hats off. Thanks once again to Mark.
> >
> >Kannaiyan
> >
> >On 9/5/06, Joe shmoe <[EMAIL PROTECTED]> wrote:
> >>Would you like to have the codecs written by Mark
> >>Spencer for Asterisk?  The same binary codecs
> >>available when you purchas

Re: [asterisk-users] File structure question

[Jay Moore]

Peter Bowyer wrote:

On 04/09/06, Jay Moore <[EMAIL PROTECTED]> wrote:

Marco: Ah I see.  There's a [general] context.  I'm pretty new to this
Asterisk stuff and I didn't realize there was a general context that you
could do things like global includes.  Thanks, I'll give it a shot when
I'm back in the office on Tuesday.

Peter:  No need to be an ass about it, pal.  Not all of us are as adept
at this as you are.


You've still not got it. #include is a general text include - can be
used anywhere. Well, perhaps it has to be at the start of a line.

Contexts, not even the [general] section which isn't actually a
context, has any relevance. It will insert the contents of the
included file as though it was in the main file, wherever you put it.

You could put the whole of the sip.conf file in an #include'd file.
The whole of one context. One and a half contexts. 2 lines out of the
[general] section. And so on.

All of which, to repeat, could be experienced with a small investment
of your time. It really does pay to experiment with the simple things,
you find your learning curve is so much flatter than if you ask
questions in a vacuum.

Peter




Perhaps if answering the simple things politely is too difficult for 
you, you'd be better off not answering at all.  Someday, I hope, you'll 
find that 'simple' is a relative term.


Jay
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Re: [asterisk-users] Asterisk calling through FWD?

[Nick Ellson]

-- Call accepted by 192.246.69.186 (format ulaw)
-- Format for call is ulaw
-- IAX2/192.246.69.186:4569-3 is busy
-- Hungup 'IAX2/192.246.69.186:4569-3'
  == Everyone is busy/congested at this time (1:1/0/0)
-- Executing Congestion("SIP/4003-5d5e", "") in new stack
  == Spawn extension (default, 393<6 digits>, 3) exited non-zero on
'SIP/4003-5d5e'



heh.. i got that all the time when i had the 8xx #'s routed thru FWD.. maybe 
50-60% of the
calls actually went thru

incoming did from ipkall using fwd seems to work ok most of the time


I am checking now to see if my Brother actually set up his voice mail, I 
wonder if that is the issue foe me tonight now that I have the dialer 
going out with no errors. Now the call to a 5 digit FWD number when first 
shot.. Ugh... Still fun though :)


Nick

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Re: [asterisk-users] Digum g729 and g723

[brandon kruz]

haha
im reporting this
but i think your a nub
and what happend was you just took the actual files
so it wasnt hard, you just dont have a physical liscense
there are things all over the web on how to enable g729
its not hard
its the liscensing that can get you sued for a very much amount of money...


you can get any software you want and illegally install it
using that in an enterprise envirnment, thats just stupid, and you will get 
caught.



of course if this is in fact legit, im very very sorry, im a nub
but from the sound of it, it sounds illegal, you will get caught

fess up
get a job
a pay for business edition

my 2 cents
`KruZ~

please someone let me know if this is in fact legit, else i am now 
researching this, and it will be stopped


please dont do anything illegal and stupid
it doesnt make you any more 1337 than you can find ways to
enable these codecs without paying for them

its your chance to get sued.
go for it.




From: "Kannaiyan Natesan" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussion
To: "Asterisk Users Mailing List - Non-Commercial 
Discussion", "Commercial and 
Business-Oriented Asterisk Discussion", 
"Asterisk Developers Mailing List" 

Subject: Re: [asterisk-users] Digum g729 and g723
Date: Tue, 5 Sep 2006 10:47:37 +0800
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Hey,

Is this code released by Digium?
Looks like directly from digium. Is it GPL with License and Royalty?

Unlimited channels and no restriction !

Author mentioned as Mark Spencer.
If we want to pay the license fees, should we have to Pay to VoiceAge 
directly?


Hats off. Thanks once again to Mark.

Kannaiyan

On 9/5/06, Joe shmoe <[EMAIL PROTECTED]> wrote:

Would you like to have the codecs written by Mark
Spencer for Asterisk?  The same binary codecs
available when you purchase a licence?  You're in
luck!  The following link will allow you to have
Digiums codecs.


http://www.savefile.com/files/20972

Come one come all!


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Re: [asterisk-users] Blind transfer 3/4 digits

[Ronald Wiplinger]

wendell hamilton wrote:

Please excuse the top-posting.

  

... so we are faster at the solution, ... ;-)

In features.conf, uncomment transferdigittimeout and adjust its timing as 
desired.  You may also want to uncomment and adjust featuredigittimeout to a 
higher value as well.

That was it!!! Now it works!!!


  Also, since the dialplan does first match, you can eliminate the problem by 
putting the 4 digit extensions before the 3 digit extensions in the dialplan.

See the "match as you go" section at
http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching

  


Thank you for the link, btw. your comment above does not "match" the 
link. Copy of the important part of your provided link:




  Example

FooBar Incorporated wants their incoming telephone calls to be 
answered with a voice message welcoming the caller and inviting them 
to choose which extension they want. FooBar has six telephone 
extensions. Their extension numbers are 1, 2, 21, 22, 31, 32. So this 
is the context created for incoming calls for FooBar Incorporated:


   [incoming]
   exten => s,1,Background(welcome-to-foobar-incorporated)
   exten => 1,1,Dial(Zap/1)
   exten => 2,1,Dial(Zap/2)
   exten => 21,1,Dial(Zap/3)
   exten => 22,1,Dial(Zap/4
   exten => 31,1,Dial(Zap/5)
   exten => 32,1,Dial(Zap/6)

When you call FooBar, Asterisk plays the 
"welcome-to-foobar-incorporated.gsm" sound file. After that, having 
run out of commands to execute, it waits for you to dial something. 
This is what Asterisk would do if you dialed various options:


   Number DialedAsterisk's Action
 1  Immediately performs Dial (Zap/1)
 2  Waits for timeout, then performs Dial(Zap/2)
21  Immediately performs Dial (Zap/3)
22  Immediately performs Dial (Zap/4)
 3  Waits for timeout, then hangs up.
31  Immediately performs Dial (Zap/5)
32  Immediately performs Dial (Zap/6)
 4  Immediately hangs up.

Note that when a caller tries to dial extension 2, they are not 
connected immediately. Asterisk waits to see if the caller dials more 
digits, to determine whether the caller wants extension 2 or 21 or 22. 
As callers would like to be connected immediately if possible, it 
would be more user-friendly to avoid using ambiguous extension numbers. 




Thanks for the solution, 

bye

Ronald


HTH

routerguy

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger
Sent: Monday, September 04, 2006 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind transfer 3/4 digits

David Gagnon wrote:
  

Ronald,

Like someone already told you, you should explain more clearly the
way you try to transfer, we need more details on the procedure, using which
button on which phone. We need every detail to help you. This as nothing to
do with the way the dial plan is loaded, this is totally false.

I'm sure most of the people here don't understand how you try to
transfer.

David

  



David,

I am not sure how the explanation how to punch the keys changes 
something,  ;-.)


Ok, here we go:

Snom:
pick up the phone and hit ##6014 followed by [ok]

Noname:
pick up the phone and hit ##6014 must be pushed very fast!!! No end 
# needed, since the phone 601 starts to ring as soon I reach 1.


In my opinion Asterisk remembers all numbers and therefore it does not 
wait for the 4, since it found a match. This is in VoIP (in my opinion) 
wrong, since overlapping numbers are allowed.


Sip message / dtmf, this is something different! How is the transfer 
made? Maybe snom does send a sip message, while the noname only send 
dtmf tones.


bye

Ronald



  

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 4 septembre 2006 09:22
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits

Koopmann, Jan-Peter wrote:
  


On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:

  

  

try that way. However, I have doubts as well. If you are right, than
why snom phone does not have this problem? Would not here also the
first match count?   

  


Because the transfer button on the SNOM is using a totally different

  

mechanism than sending # to Asterisk. On your snom configuration (like ours)
the phone does not start to create/send a SIP message until you hit "OK". At
that time the entire number is there and a complete SIP transfer is created.
Cool down a bit. The problem you are having is most probably just a dialplan
problem. It takes some time and experience to get those things right. No
need to yell here...
  

  

  

What's happen to you guys? I am not yelling, just asking.
It is sure not a dialplan question! If it would be a dialp

Re: [asterisk-users] Asterisk calling through FWD?

[Derek Whitten]

Nick Ellson wrote:
> 
> Hi Michael,
> 
> I tried what you had said and then tried calling you, and it worked.
> Then I called my brother and while I did not get the error, I still got
> the "busy" message i was getting before I borked my config trying too
> many ideas ;)
> 
> So, any other 6 digit FWD users willing to take a call from me? Just so
> I can eliminate the call string?
> 
> My two back to back calls..
> 
> *CLI>
> -- Executing SetCallerID("SIP/4003-508e", "Nick Ellson") in new stack
> -- Executing Dial("SIP/4003-508e",
> "IAX2/776754:@iax2.fwdnet.net/<5 digits>|60|r") in new stack
> -- Called 776754:@iax2.fwdnet.net/<5 digits>
> -- Call accepted by 192.246.69.186 (format ulaw)
> -- Format for call is ulaw
> -- IAX2/192.246.69.186:4569-2 answered SIP/4003-508e
> -- Hungup 'IAX2/192.246.69.186:4569-2'
>   == Spawn extension (default, 393<5 digits>, 2) exited non-zero on
> 'SIP/4003-508e'
> -- Executing SetCallerID("SIP/4003-5d5e", "Nick Ellson") in new stack
> -- Executing Dial("SIP/4003-5d5e",
> "IAX2/776754:@iax2.fwdnet.net/<6 digits>|60|r") in new stack
> -- Called 776754:@iax2.fwdnet.net/<6 digits>
> -- Call accepted by 192.246.69.186 (format ulaw)
> -- Format for call is ulaw
> -- IAX2/192.246.69.186:4569-3 is busy
> -- Hungup 'IAX2/192.246.69.186:4569-3'
>   == Everyone is busy/congested at this time (1:1/0/0)
> -- Executing Congestion("SIP/4003-5d5e", "") in new stack
>   == Spawn extension (default, 393<6 digits>, 3) exited non-zero on
> 'SIP/4003-5d5e'
> 
> 
heh.. i got that all the time when i had the 8xx #'s routed thru FWD.. maybe 
50-60% of the
 calls actually went thru

incoming did from ipkall using fwd seems to work ok most of the time




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Re: [asterisk-users] Asterisk calling through FWD?

[Nick Ellson]

Hi Michael,

I tried what you had said and then tried calling you, and it worked. Then 
I called my brother and while I did not get the error, I still got the 
"busy" message i was getting before I borked my config trying too many 
ideas ;)


So, any other 6 digit FWD users willing to take a call from me? Just so I 
can eliminate the call string?


My two back to back calls..

*CLI>
-- Executing SetCallerID("SIP/4003-508e", "Nick Ellson") in new stack
-- Executing Dial("SIP/4003-508e", 
"IAX2/776754:@iax2.fwdnet.net/<5 digits>|60|r") in new stack

-- Called 776754:@iax2.fwdnet.net/<5 digits>
-- Call accepted by 192.246.69.186 (format ulaw)
-- Format for call is ulaw
-- IAX2/192.246.69.186:4569-2 answered SIP/4003-508e
-- Hungup 'IAX2/192.246.69.186:4569-2'
  == Spawn extension (default, 393<5 digits>, 2) exited non-zero on 
'SIP/4003-508e'

-- Executing SetCallerID("SIP/4003-5d5e", "Nick Ellson") in new stack
-- Executing Dial("SIP/4003-5d5e", 
"IAX2/776754:@iax2.fwdnet.net/<6 digits>|60|r") in new stack

-- Called 776754:@iax2.fwdnet.net/<6 digits>
-- Call accepted by 192.246.69.186 (format ulaw)
-- Format for call is ulaw
-- IAX2/192.246.69.186:4569-3 is busy
-- Hungup 'IAX2/192.246.69.186:4569-3'
  == Everyone is busy/congested at this time (1:1/0/0)
-- Executing Congestion("SIP/4003-5d5e", "") in new stack
  == Spawn extension (default, 393<6 digits>, 3) exited non-zero on 
'SIP/4003-5d5e'



--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.


On Mon, 4 Sep 2006, Michael Graves wrote:


Nick,

Now I remember why this doesn't work. It's the caller ID settings. The syntax 
you use is older and makes two separate calls


 exten => _393.,1,SetCIDNum(${FWDNUMBER})
 exten => _393.,2,SetCallerID,${FWDCIDNAME}


This won't work for some reason. As soon as I changed my settings to:

exten => _393.,1,SetCallerID,${FWDCIDNAME}
exten => _393.,2,Dial(IAX2/54245:[EMAIL PROTECTED]/${EXTEN:3},60)
exten => _393.,3,Congestion

Then it worked. The SetCIDNum function broke it. I can't say why, only that I 
inquired with folk at FWD who told me that it was most definitely at my end.

Feel free to call my fwd number. It rings at my desk. If I'm there I answer but 
you may just get VM.

Michael

On Mon, 4 Sep 2006 16:06:42 -0700 (PDT), Nick Ellson wrote:



I thought maybe my configs would have been a good idea to post:



iax.conf:



[general]
bindport=4569
bindaddr=10.0.0.20
bandwidth=medium
disallow=lpc10
allow=gsm
jitterbuffer=no
forcejitterbuffer=no



register => 776754:@iax2.fwdnet.net
allow=ulaw
tos=lowdelay
autokill=yes



[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup



Extensions.conf



 [globals]



 FWDNUMBER=776754
 FWDCIDNAME=Nick Ellson
 FWDPASSWORD=
 FWDRINGS=SIP/4003
 FWDVMBOX=4003




 [default]
 include => mainmenu
 exten => _393.,1,SetCIDNum(${FWDNUMBER})
 exten => _393.,2,SetCallerID,${FWDCIDNAME}
 exten => _393.,3,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:3},60,r)
 exten => _393.,4,Congestion



[fromiaxfwd]
exten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
exten => ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
exten => ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}



-



And I am not sure if something changed but now I get:



  -- Executing SetCIDNum("SIP/4003-4dcc", "776754") in new stack
-- Executing SetCallerID("SIP/4003-4dcc", "Nick Ellson") in new stack
-- Executing Dial("SIP/4003-4dcc", "IAX2/776754:[EMAIL 
PROTECTED]/XX|60|r") in new stack
-- Called 776754:[EMAIL PROTECTED]/XX
-- IAX2/fwd-gw-5 is circuit-busy
Sep  4 15:47:54 WARNING[28513]: chan_iax2.c:7013 socket_read: Call rejected by 
192.246.69.186: No authority found
Sep  4 15:47:54 NOTICE[28513]: chan_iax2.c:1601 iax2_destroy: Avoiding IAX 
destroy deadlock
-- Hungup 'IAX2/fwd-gw-5'
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion("SIP/4003-4dcc", "") in new stack
  == Spawn extension (default, 393XX, 4) exited non-zero on
'SIP/4003-4dcc'



The "No Authority found" I think is new? I am going to figure out how to
increase the logging, but does anyone see an obviuos boo-boo?



Nick





--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.




On Mon, 4 Sep 2006, Michael Graves wrote:



I had similar troubleat first. Try not specifying CID. As I recall FWD is 
sensitive to this.

Michael


On Sun, 3 Sep 2006 22:26:35 -0700 (PDT), Nick Ellson wrote:



Hi all,



I have been researching a dialing problem I am having with FWD. I followed
their IAX2 config notes, and I can receive calls from my brother from FWD,
and all the echo tests, call me services work. But I cannot call him.



-- Executing SetCallerID("SIP/4003-9de6", ""Nick Ellson"") in new
stack
-- Executing Dial("SIP/4003-9de6",
"IAX2/776754:@iax2.fwdnet.net/|60|r") in new stack
-- 

Re: [asterisk-users] Digum g729 and g723

[Kannaiyan Natesan]

Hey,

Is this code released by Digium?
Looks like directly from digium. Is it GPL with License and Royalty?

Unlimited channels and no restriction !

Author mentioned as Mark Spencer.
If we want to pay the license fees, should we have to Pay to VoiceAge directly?

Hats off. Thanks once again to Mark.

Kannaiyan

On 9/5/06, Joe shmoe <[EMAIL PROTECTED]> wrote:

Would you like to have the codecs written by Mark
Spencer for Asterisk?  The same binary codecs
available when you purchase a licence?  You're in
luck!  The following link will allow you to have
Digiums codecs.


http://www.savefile.com/files/20972

Come one come all!


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Re: [asterisk-users] Asterisk 1.2.11 and # key

[Michael Strelnikov]

I have "blindxfer => ##" line in my features.confOn 9/5/06, David Gagnon <[EMAIL PROTECTED]> wrote:














Are you sure this is not
because of the dynamic features in features.conf ?

By default, # is defined
for the transfer feature.

 

David

 









De :

[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] De la part de
 Michael Strelnikov
Envoyé : 4 septembre 2006
09:53
À :
asterisk-users@lists.digium.com
Objet : [asterisk-users]
Asterisk 1.2.11 and # key



 

Hello,

   Does anybody have problems with recognition of the hash (#) key
with * 1.2.11? It seams that after pressing # the call is in a progress but no
data is sent.

Thanks in advance,
Michael







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[asterisk-users] Digum g729 and g723

[Joe shmoe]

Would you like to have the codecs written by Mark
Spencer for Asterisk?  The same binary codecs
available when you purchase a licence?  You're in
luck!  The following link will allow you to have
Digiums codecs.


http://www.savefile.com/files/20972

Come one come all!


__
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Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel

[RR]

Hi there,

sorry I wasn't sure exactly where to start so didn't know what info to
provide. Now that I know, here's the info

1) using a P4 w/HT
2) Using CentOS 4.3 with the 2.6.9-34.0.1-smp (Note, this was
installed through an rpm, but the (*) and zaptel code is being
compiled against the source of this)
3) I have tried it with and without ztdummy, and nothing changes.
Although voicemail should have nothing to do with ztdummy, am I
correct?
4) I have also tried with and without uncommenting the line for GSM
optimisation for MMX processors line in the Makefile
5) I've also tried rebooting the machine with the line "acpi=ht" at
the Kernel command line
6) Also tried strictly using one codec so as to avoid transcoding to
see if that was it
7) I've tried booting into an SMP kernel without building (*) and
zaptel for an smp kernel

None of the above has helped. If I don't boot into an SMP kernel at
all, it works fine.

Also, at every start of (*), the "show translation" command shows
different transcoding times without changing a single thing in the
system in the way of config etc. Why is that?

Oh also, note that this system is running inside of a Virtual Machine
with 768 RAM and a 3.4GHz CPU although NO other VM is active on this
VM server.

Any ideas?

Thx
\R
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H.264 basic backport (was Re: [asterisk-users] Grandstream and H.264 !)

[Nic Bellamy]

Carlos Chavez wrote:


On Mon, 2006-09-04 at 16:26 -0300, Sergio (Red) wrote:
 


hi,
I´ need some help to implement the Grandstream GXV-3000 in my * 
platform.  Someone know the state of H.264 Video Codec for Asterrisk??
   


This has been answered multiple times in the last month.  Search the
list before posting.  


You have to use the SVN version of Asterisk if you wish to use H264.
 

I've been sitting on this for a while, but just got around to adding it 
to asterisk-backports.org: basic H.264 passthrough support for 1.2.11.


It's at the "Works for me with a pair of GXV3000s" stage.

http://asterisk-backports.org/wiki/index.php/Passthrough-h264

Cheers,
   Nic.

--
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/

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[asterisk-users] SNMP with 1.2.11 stable

[Azfhasterisk]

Can anyone tell me if it is possible to get the asterisk SNMP
module working with ver 1.2.11 stable. Everything that I am coming across is
talking about using the trunk version. 

 

If it is how do I get it compiled with this version?

 

Thanks

 

Rick

 






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RE: [asterisk-users] Zaptel-1.2.8 compile problem

[Rushowr]

 




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vidura
Senadeera
Sent: Monday, September 04, 2006 5:15 AM
To: asterisk-users@lists.digium.com
Cc: asterisk-dev@lists.digium.com
Subject: [asterisk-users] Zaptel-1.2.8 compile problem





Hi, 
 
I have problem in compiling zaptel-1.2.8. My Linux version is 2.6.
asterisk version and libpri versions are
1.2.11 and 1.2.3. 
 
Please refer the attached txt files for Linux version information
and output of zaptel compile.
 
I will be highly appreciated that any one can help me on this
regard.

-- 
Thanks & Regards,
Vidura B. Senadeera. 


-- 
Thanks & Regards,
Vidura B. Senadeera.  
 


For the love of all things you hold holy, why is it that people cannot learn
to NOT CROSS POST!?! I, for one, don't appreciate getting 4 copies of the
above message.





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RE: [asterisk-users] Blind transfer 3/4 digits

[wendell hamilton]

Please excuse the top-posting.

In features.conf, uncomment transferdigittimeout and adjust its timing as 
desired.  You may also want to uncomment and adjust featuredigittimeout to a 
higher value as well.  Also, since the dialplan does first match, you can 
eliminate the problem by putting the 4 digit extensions before the 3 digit 
extensions in the dialplan.

See the "match as you go" section at
http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching

HTH

routerguy

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger
Sent: Monday, September 04, 2006 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind transfer 3/4 digits

David Gagnon wrote:
> Ronald,
>
>   Like someone already told you, you should explain more clearly the
> way you try to transfer, we need more details on the procedure, using which
> button on which phone. We need every detail to help you. This as nothing to
> do with the way the dial plan is loaded, this is totally false.
>
>   I'm sure most of the people here don't understand how you try to
> transfer.
>
>   David
>
>   

David,

I am not sure how the explanation how to punch the keys changes 
something,  ;-.)

Ok, here we go:

Snom:
pick up the phone and hit ##6014 followed by [ok]

Noname:
pick up the phone and hit ##6014 must be pushed very fast!!! No end 
# needed, since the phone 601 starts to ring as soon I reach 1.

In my opinion Asterisk remembers all numbers and therefore it does not 
wait for the 4, since it found a match. This is in VoIP (in my opinion) 
wrong, since overlapping numbers are allowed.

Sip message / dtmf, this is something different! How is the transfer 
made? Maybe snom does send a sip message, while the noname only send 
dtmf tones.

bye

Ronald



> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de Ronald
> Wiplinger
> Envoyé : 4 septembre 2006 09:22
> À : Asterisk Users Mailing List - Non-Commercial Discussion
> Objet : Re: [asterisk-users] Blind transfer 3/4 digits
>
> Koopmann, Jan-Peter wrote:
>   
>> On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:
>>
>>   
>> 
>>> try that way. However, I have doubts as well. If you are right, than
>>> why snom phone does not have this problem? Would not here also the
>>> first match count?   
>>> 
>>>   
>> Because the transfer button on the SNOM is using a totally different
>> 
> mechanism than sending # to Asterisk. On your snom configuration (like ours)
> the phone does not start to create/send a SIP message until you hit "OK". At
> that time the entire number is there and a complete SIP transfer is created.
> Cool down a bit. The problem you are having is most probably just a dialplan
> problem. It takes some time and experience to get those things right. No
> need to yell here...
>   
>>   
>> 
> What's happen to you guys? I am not yelling, just asking.
> It is sure not a dialplan question! If it would be a dialplan question, 
> than it would be for each dialing, but it isn't.
>
> You mentioned SIP message and that makes me wonder! Are we not using 
> here dtmf ?? that is in my opinion not a sip message, isn't it?
> If it is a sequence of "tones", than why is it different if it is in a 
> string (like snom) or another phone, with single tones?
> If we understand this part, than is the question, where can I turn on 
> the system to take a longer break between "tones" still as a string?
>
> Back to the dialplan:
> A Voip number can have different length of digits. Each number is seen 
> as a complete "picture", and so a three digit and a four digit number is 
> something different. While in the legacy telephony the digits are worked 
> down one by one and if there is no more use of the digits, they are just 
> garbage and will be not used. Unlike in VoIP, where you can have a three 
> digit number and if you dial four digit, than it is a WRONG number  
> I just verified that: I dialed from 601 to  61522, however, 61522 does 
> not exist, but 615 exists. Guess what? I get a busy tone! That should 
> proof my thoughts (and that without yelling, ... hehehehe)
>
> bye
>
> Ronald
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>
>
> ---
> avast! Antivirus: Inbound message clean.
> Virus Database (VPS): 0635-4, 2006/09/01
> Tested on: 2006/9/4 ¤U¤È 11:40:24
> avast! - copyright (c) 1988-2006 ALWIL Software.
> http://www.avast.com
>
>
>
>
>   


-- 
Ronald Wipling

Re: [asterisk-users] Blind transfer 3/4 digits

[Ronald Wiplinger]

David Gagnon wrote:

Ronald,

Like someone already told you, you should explain more clearly the
way you try to transfer, we need more details on the procedure, using which
button on which phone. We need every detail to help you. This as nothing to
do with the way the dial plan is loaded, this is totally false.

I'm sure most of the people here don't understand how you try to
transfer.

David

  


David,

I am not sure how the explanation how to punch the keys changes 
something,  ;-.)


Ok, here we go:

Snom:
pick up the phone and hit ##6014 followed by [ok]

Noname:
pick up the phone and hit ##6014 must be pushed very fast!!! No end 
# needed, since the phone 601 starts to ring as soon I reach 1.


In my opinion Asterisk remembers all numbers and therefore it does not 
wait for the 4, since it found a match. This is in VoIP (in my opinion) 
wrong, since overlapping numbers are allowed.


Sip message / dtmf, this is something different! How is the transfer 
made? Maybe snom does send a sip message, while the noname only send 
dtmf tones.


bye

Ronald


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 4 septembre 2006 09:22
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits

Koopmann, Jan-Peter wrote:
  

On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:

  


try that way. However, I have doubts as well. If you are right, than
why snom phone does not have this problem? Would not here also the
first match count?   

  

Because the transfer button on the SNOM is using a totally different


mechanism than sending # to Asterisk. On your snom configuration (like ours)
the phone does not start to create/send a SIP message until you hit "OK". At
that time the entire number is there and a complete SIP transfer is created.
Cool down a bit. The problem you are having is most probably just a dialplan
problem. It takes some time and experience to get those things right. No
need to yell here...
  
  


What's happen to you guys? I am not yelling, just asking.
It is sure not a dialplan question! If it would be a dialplan question, 
than it would be for each dialing, but it isn't.


You mentioned SIP message and that makes me wonder! Are we not using 
here dtmf ?? that is in my opinion not a sip message, isn't it?
If it is a sequence of "tones", than why is it different if it is in a 
string (like snom) or another phone, with single tones?
If we understand this part, than is the question, where can I turn on 
the system to take a longer break between "tones" still as a string?


Back to the dialplan:
A Voip number can have different length of digits. Each number is seen 
as a complete "picture", and so a three digit and a four digit number is 
something different. While in the legacy telephony the digits are worked 
down one by one and if there is no more use of the digits, they are just 
garbage and will be not used. Unlike in VoIP, where you can have a three 
digit number and if you dial four digit, than it is a WRONG number  
I just verified that: I dialed from 601 to  61522, however, 61522 does 
not exist, but 615 exists. Guess what? I get a busy tone! That should 
proof my thoughts (and that without yelling, ... hehehehe)


bye

Ronald
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---
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Virus Database (VPS): 0635-4, 2006/09/01
Tested on: 2006/9/4 ¤U¤È 11:40:24
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http://www.avast.com




  



--
Ronald Wiplinger  (CEO of ELMIT)
http://www.elmit.com  http://voip.elmit.com  http://e-paper.elmit.com 
Tel. (M) +886.939.775.516  (O) +886.2.2835.7765 (ENUM)   or FWD 511208

- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org

PS: Spam prevention!
Our system is protected with a spam prevention program. 
If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. 
After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.


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Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel

[Leo Ann Boon]

RR wrote:

Hi Zoa,

thanks for responding. Ok, now where do I find this? I'm running
2.6.9-34.0.1 kernel. I tried doing a bit of search and it seems like
that the ability to change the frequency doesn't appear till 2.6.13.
Am I looking at the right thing? Any hints?

You need to provide more info:
a. Are you using 2 CPU/core or just Hyperthreading?
b. Which distribution are your using? From your kernel version it looks 
like RedHat Enterprise 4 or one of its derivatives (Centos, Whitebox, etc).

c. Did you load ztdummy?

Leo.

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Re: [asterisk-users] Asterisk calling through FWD?

[Nick Ellson]

I thought maybe my configs would have been a good idea to post:

iax.conf:

[general]
bindport=4569
bindaddr=10.0.0.20
bandwidth=medium
disallow=lpc10
allow=gsm
jitterbuffer=no
forcejitterbuffer=no

register => 776754:@iax2.fwdnet.net
allow=ulaw
tos=lowdelay
autokill=yes

[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup

Extensions.conf

 [globals]

  FWDNUMBER=776754
  FWDCIDNAME=Nick Ellson
  FWDPASSWORD=
  FWDRINGS=SIP/4003
  FWDVMBOX=4003


  [default]
  include => mainmenu
  exten => _393.,1,SetCIDNum(${FWDNUMBER})
  exten => _393.,2,SetCallerID,${FWDCIDNAME}
  exten =>
  _393.,3,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:3},60,r)
  exten => _393.,4,Congestion

[fromiaxfwd]
exten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
exten => ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
exten => ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}

-

And I am not sure if something changed but now I get:

   -- Executing SetCIDNum("SIP/4003-4dcc", "776754") in new stack
 -- Executing SetCallerID("SIP/4003-4dcc", "Nick Ellson") in new stack
 -- Executing Dial("SIP/4003-4dcc",
 "IAX2/776754:[EMAIL PROTECTED]/XX|60|r") in new stack
 -- Called 776754:[EMAIL PROTECTED]/XX
 -- IAX2/fwd-gw-5 is circuit-busy
Sep  4 15:47:54 WARNING[28513]: chan_iax2.c:7013 socket_read: Call rejected by 
192.246.69.186: No authority found
Sep  4 15:47:54 NOTICE[28513]: chan_iax2.c:1601 iax2_destroy: Avoiding IAX 
destroy deadlock

 -- Hungup 'IAX2/fwd-gw-5'
   == Everyone is busy/congested at this time (1:0/1/0)
 -- Executing Congestion("SIP/4003-4dcc", "") in new stack
  == Spawn extension (default, 393XX, 4) exited non-zero on 'SIP/4003-4dcc'

The "No Authority found" I think is new? I am going to figure out how to 
increase the logging, but does anyone see an obviuos boo-boo?


Nick



--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.


On Mon, 4 Sep 2006, Michael Graves wrote:


 I had similar troubleat first. Try not specifying CID. As I recall FWD is
 sensitive to this.

 Michael

 fwd: 54245

 On Sun, 3 Sep 2006 22:26:35 -0700 (PDT), Nick Ellson wrote:


>  Hi all,

>  I have been researching a dialing problem I am having with FWD. I followed
>  their IAX2 config notes, and I can receive calls from my brother from FWD,
>  and all the echo tests, call me services work. But I cannot call him.

>  -- Executing SetCallerID("SIP/4003-9de6", ""Nick Ellson"") in new
>  stack
>  -- Executing Dial("SIP/4003-9de6",
>  "IAX2/776754:@iax2.fwdnet.net/|60|r") in new stack
>  -- Called 776754:@iax2.fwdnet.net/
>  -- Call accepted by 192.246.69.186 (format ulaw)
>  -- Format for call is ulaw
>  -- IAX2/192.246.69.186:4569-2 is busy
>  -- Hungup 'IAX2/192.246.69.186:4569-2'
>== Everyone is busy/congested at this time (1:1/0/0)
>  -- Executing Congestion("SIP/4003-9de6", "") in new stack
>== Spawn extension (default, 393, 3) exited non-zero on
>  'SIP/4003-9de6'

>  This is pretty much just what a few others from the FWD forums have posted
>  with no real response.

>  Has any one of you also had this problem with FWD?

>  Nick



>  --
>  Nick Ellson
>  CCDA, CCNP, CCSP, CCAI,
>  MCSE 2000, Security+, Network+
>  Network Hobbyist, VFR Private Pilot.

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[asterisk-users] Re: Nufone making changes

[Justin Newman]

Justin Newman wrote:

Looks like Nufone is making some positive changes...


Thanks Justin, but Asterisk-users is for Asterisk discussion only.
Perhaps a more appropriate list would be asterisk-biz.

Jeremy McNamara


Noted. Nonetheless, looks like you guys are making progress. :p

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Re: [asterisk-users] app_conference not working for me

[Matt Riddell (IT)]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Steve Edwards wrote:
> I'm having trouble getting app_conference to work and I'm feeling
> pretty clueless right know.

Probably the iaxclient list would be the better forum to discuss this as
its not in the Asterisk codebase.

To sign up for the iaxclient mailing list go to:

https://lists.sourceforge.net/lists/listinfo/iaxclient-devel

- --
Cheers,

Matt Riddell
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Re: [asterisk-users] FAX handling

[Tzafrir Cohen]

On Mon, Sep 04, 2006 at 05:16:00PM -0400, Technical Support wrote:
> Look into NVDETECT, and fax2mail script on www.generationd.com
> 
> Fax detection is automatic

B ut not needed, as calls already come from chan_zap that has its own
fax detection.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
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RE: [asterisk-users] FAX handling

[Technical Support]

Look into NVDETECT, and fax2mail script on www.generationd.com

Fax detection is automatic

MD 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Monday, September 04, 2006 5:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] FAX handling

On Mon, Sep 04, 2006 at 04:35:43PM +0200, Jose Limeres wrote:
> Hi all,
> 
> I am using asterisk 1.2.10 BRI stuffed 0.3.0-PRE-1s with zaptel 1.2.8
> and we are trying to have FAX receiving working in
> one of the BRI lines.
> 
> No problem with FAX transmissions but we can not receive. I have
> configured in zapata.conf faxdetect=both (tx and rx).
> FAX machine is connected to one FXS port on a PAP2 with G711a and no
> echo cancelation configured. When the FAX arrives at the
> FAX machine, they start negotiating but then it stops as if the format
> is not recognized by the Fax machine as a valid fax.
> 
> Does anyone have a similar configuration working?

Hmmm

How can asterisk detect that the call is a fax? Is it by answering it?

If you get rid of the fax detection and send everything to the fax
extension, will faxes get through?

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] Call center reports

[Hermann Wecke]

Technical Support wrote:

Can someone point me to call center reports available from Asterisk?


http://queuemetrics.loway.it/
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Re: [asterisk-users] missing pri connect (wwomera to pri)

[Tzafrir Cohen]

On Mon, Sep 04, 2006 at 01:27:59PM -0400, Rosario Pingaro wrote:
> I am using my asterisk box with an application linked through the WOOMERA 
> channel. Asterisk bridge the woomera chennel to zap (sangoma aft104d) and 
> vicecersa.
> 
> The strange think is that after some hours of hevy load asterisk miss some 
> time to relay the connect message from woomera to pri.
> The debug from woomera and pri is pretty easy, just no connect passed to 
> pri..

So what does happen to such a bad call? Could you be more specific?
(trace, please)

Versions of asterisk, zaptel and woomera may also help a bit.

> 
> If I run asterisk without priority I get the problem early, using priority I 
> have some hours of regular work. This is very wired.
> 
> Is someone experiencing some issue? 
> Does someone that has woomera knowledg help me to fix the issue?

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] FAX handling

[Tzafrir Cohen]

On Mon, Sep 04, 2006 at 04:35:43PM +0200, Jose Limeres wrote:
> Hi all,
> 
> I am using asterisk 1.2.10 BRI stuffed 0.3.0-PRE-1s with zaptel 1.2.8
> and we are trying to have FAX receiving working in
> one of the BRI lines.
> 
> No problem with FAX transmissions but we can not receive. I have
> configured in zapata.conf faxdetect=both (tx and rx).
> FAX machine is connected to one FXS port on a PAP2 with G711a and no
> echo cancelation configured. When the FAX arrives at the
> FAX machine, they start negotiating but then it stops as if the format
> is not recognized by the Fax machine as a valid fax.
> 
> Does anyone have a similar configuration working?

Hmmm

How can asterisk detect that the call is a fax? Is it by answering it?

If you get rid of the fax detection and send everything to the fax
extension, will faxes get through?

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] Call center reports

[Technical Support]

Can someone point me 
to call center reports available from Asterisk?  We setup a small call 
center with agents, and will now be looking at reports.
 
Ideas?
 
Thanks,
MD
 
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RE: [asterisk-users] includes in realtime ??

[Rushowr]

>-Original Message-
>From: [EMAIL PROTECTED] 
>[mailto:[EMAIL PROTECTED] On Behalf Of 
>Benjamin Jacob
>Sent: Monday, September 04, 2006 8:37 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [asterisk-users] includes in realtime ??
>
>Hello ppl,
>Is it possible to include contexts in the RealTime scenario??
>If not, wots the work around??
>
>Thanks in advance.
>Ben.
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Amazing how the wiki has this vast amount of AT LEAST info to start your
research on
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions


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Re: [asterisk-users] Looks like Nufone is changing around...

[Jeremy McNamara]

Justin Newman wrote:

Looks like Nufone is making some positive changes...



Thanks Justin, but Asterisk-users is for Asterisk discussion only.
Perhaps a more appropriate list would be asterisk-biz.



Jeremy McNamara
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[asterisk-users] app_conference not working for me

[Steve Edwards]

I'm having trouble getting app_conference to work and I'm feeling
pretty clueless right know.

With no flags, it doesn't exit when I press '#.'

With flags passed as "d," it just ignores '#.'

With flags passed as "MTV," it crashes Asterisk when I press '#.'

Any clues would be appreciated :)

Here's how I'm invoking conference:

exten = *,n,conference(test)

or

exten = *,n,conference(test|d)

or

exten = *,n,conference(test|MTV)

Here's the console output with no flags:

-- Accepting AUTHENTICATED call from a.b.c.d:
   > requested format = ulaw,
   > requested prefs = (),
   > actual format = ulaw,
   > host prefs = (ulaw),
   > priority = mine
-- Executing Conference("IAX2/a.b.c.d:1030-4", "test") in new stack
Sep  4 12:53:41 ERROR[10454]: frame.c:386 convert_frame: unable to translate 
frame

Here's what gets syslogged:

Sep  4 12:53:35 dt-ext asterisk[10222]: VERBOSE[10227]: -- Accepting AUTHENTICATED call from a.b.c.d:> requested format = ulaw,> requested prefs = (),> actual format = ulaw,> host prefs = (ulaw),> priority = mine 
Sep  4 12:53:35 dt-ext asterisk[10222]: DEBUG[10225]: chan_iax2.c:9434 in iax2_devicestate: Checking device state for device a.b.c.d 
Sep  4 12:53:35 dt-ext asterisk[10222]: DEBUG[10225]: devicestate.c:187 in do_state_change: Changing state for IAX2/a.b.c.d:1030 - state 4 (Invalid) 
Sep  4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: pbx.c:1677 in pbx_extension_helper: Launching 'Conference' 
Sep  4 12:53:35 dt-ext asterisk[10222]: VERBOSE[10453]: -- Executing Conference("IAX2/a.b.c.d:1030-4", "test") in new stack 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:415 in member_exec: [ $Revision: 1.9 $ ] begin processing member thread, channel => IAX2/a.b.c.d:1030-4 
Sep  4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: chan_iax2.c:3370 in iax2_answer: Answering IAX2 call 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:742 in create_member: attempting to parse passed params, stringp => test 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:793 in create_member: parsed data params, id => test, flags => , priority => 0, vad_prob_start => 0.05, vad_prob_continue => 0.02 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:1077 in create_member: created member, type => S, priority => 0, readformat => 4 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:451 in member_exec: CHANNEL INFO, CHANNEL => IAX2/a.b.c.d:1030-4, DNID => *, CALLER_ID => 21012006, ANI => 21012006 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:454 in member_exec: CHANNEL CODECS, CHANNEL => IAX2/a.b.c.d:1030-4, NATIVE => 4, READ => 4, WRITE => 4 
Sep  4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: channel.c:2376 in set_format: Set channel IAX2/a.b.c.d:1030-4 to read format ulaw 
Sep  4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: channel.c:2376 in set_format: Set channel IAX2/a.b.c.d:1030-4 to write format ulaw 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:504 in start_conference: attempting to find requested conference 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:548 in find_conf: conflist has not yet been initialized, name => test 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:511 in start_conference: attempting to create requested conference 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:583 in create_conf: entered create_conf, name => test 
Sep  4 12:53:35 dt-ext asterisk[10222]: WARNING[10453]: translate.c:116 in ast_translator_build_path: No translator path from unknown to unknown 
Sep  4 12:53:35 dt-ext asterisk[10222]: WARNING[10453]: translate.c:116 in ast_translator_build_path: No translator path from unknown to alaw 
Sep  4 12:53:35 dt-ext asterisk[10222]: WARNING[10453]: translate.c:116 in ast_translator_build_path: No translator path from unknown to unknown 
Sep  4 12:53:35 dt-ext last message repeated 5 times
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:796 in add_member: member added to conference, name => test 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:646 in create_conf: added new conference to conflist, name => test 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:663 in create_conf: started conference thread for conference, name => test 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:514 in member_exec: begin member event loop, channel => IAX2/a.b.c.d:1030-4 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:532 in member_exec: Conference Members: 1 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:538 in member_exec: Quiet debug 0 - 0 
Sep  4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:546 in member_exec: skipping ent

Re: [asterisk-users] File structure question

[Peter Bowyer]

On 04/09/06, Jay Moore <[EMAIL PROTECTED]> wrote:

Marco: Ah I see.  There's a [general] context.  I'm pretty new to this
Asterisk stuff and I didn't realize there was a general context that you
could do things like global includes.  Thanks, I'll give it a shot when
I'm back in the office on Tuesday.

Peter:  No need to be an ass about it, pal.  Not all of us are as adept
at this as you are.


You've still not got it. #include is a general text include - can be
used anywhere. Well, perhaps it has to be at the start of a line.

Contexts, not even the [general] section which isn't actually a
context, has any relevance. It will insert the contents of the
included file as though it was in the main file, wherever you put it.

You could put the whole of the sip.conf file in an #include'd file.
The whole of one context. One and a half contexts. 2 lines out of the
[general] section. And so on.

All of which, to repeat, could be experienced with a small investment
of your time. It really does pay to experiment with the simple things,
you find your learning curve is so much flatter than if you ask
questions in a vacuum.

Peter


--
Peter Bowyer
Email: [EMAIL PROTECTED]
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Re: [asterisk-users] Any Hardphone with VPNClient embedded?

[Francesco Peeters (Asterisk)]

On Mon, September 4, 2006 16:55, Cory Andrews said:
> Please be aware that from a future support standpoint, you may be a bit
> limited with Zultys.  Their future seems very uncertain they have recently
> just about ceased operations and let the majority of their employees go.
>
> Cory J Andrews
> 
> voice - 800.398.VoIP X3402
> email - [EMAIL PROTECTED]
> AIM - B2CORY
> - Original Message -
> From: "Leo Ann Boon" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Monday, September 04, 2006 10:35 AM
> Subject: Re: [asterisk-users] Any Hardphone with VPNClient embedded?
>
>
>> Marco Mouta wrote:
>>> Hi all,
>>>
>>> Does any of you knows an Hardphone with VPN client embedded?
>> Take a look at Zultys SIP phones. VPN enabled.
>>
>> www.zultys.com
>

As I too am interested in IPsec capable hardphones (or ATA's), do you have
a suggestion what to look at instead?

I mean: It's nice to say the company may not be around for long, but if
there's no alternative, what choice does one have?

TIA!

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0
  AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN
  2 Sweex HFC-PCI cards
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[asterisk-users] Looks like Nufone is changing around...

[Justin Newman]

Looks like Nufone is making some positive changes...

---

NuFone Announces the Creation of a New Executive Management, Support Team

NuFone Inc., a premium-service provider specializing in hosted SIP and IAX 
VoIP solutions, is proud to announce the creation of a new executive 
management and support team.


Eugene, OR (PRWEB) September 2, 2006 -- NuFone, the world's first commercial 
provider of IAX-based VoIP services, announced today the creation of a new 
management and support team to further solidify its dedication to providing 
reliable VoIP solutions to carrier, enterprise and residential environments.


Composed of 5 executive team members, who are highly experienced in the 
areas of business, sales and support, will provide the necessary leadership 
to properly manage NuFone.


"In the past, NuFone always had trouble properly managing and supporting our 
customer's needs. It has always been my goal to form a proper team to 
deliver the support our customers demand," Jeremy McNamara, founder and CTO 
of NuFone, said. "By listening to our customers, we were able to determine 
our weaknesses and have formed a proper team to bring NuFone to the next 
level."


The following are the executive members of the NuFone management and support 
team:


Allan Noorda, President and CEO. Noorda has been engulfed in the advancement 
of the Telecommunications Industry for over the past 10 years. Noorda 
recently comes from Newman Telecom, where he was the VP of Sales and 
Marketing. He brings energy and an understanding of customer needs as well 
as technology to direct the daily operations of NuFone.


Jeremy McNamara, Founder and CTO. Over the past 10 years, McNamara has 
assisted in the development and deployment of several ISPs, ITSPs and 
Application Service Providers around the United States. McNamara also has 
extensive development, testing and deployment expertise with Asterisk 
PBX-based solutions.


Greg Merriweather, Support Specialist. Merriweather has been providing 
operating system, hardware and application support for the past 10 years 
including working as a support engineer for Ford and Global-Crossing before 
assisting in the operation of NuFone beginning in early 2003.


Leon Salisbury, Senior Engineer. Salisbury has over 20 years of programming 
and engineering experience with a wide assortment of programming languages 
including Assembler, Perl, HTML, C and hardware including PICs, 68xx Series, 
various DSP and embedded x86 platforms.


Krystina Patterson, Customer Relations. Patterson has been working with the 
public for the past 4 years in marketing and customer relations. Patterson 
is well versed in problem solving and determining customer needs.


About NuFone

NuFone was originally deployed by Jeremy McNamara in January 2002, as an 
IAX-based solution for Asterisk PBX based users. NuFone has since grown into 
a leading provider of SIP and IAX based VoIP solutions for thousands of 
customers in the United states and more than 60 countries world-wide. NuFone 
is a privately held corporation based in Eugene, OR.


###

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Re: [asterisk-users] File structure question

[Jay Moore]

Marco: Ah I see.  There's a [general] context.  I'm pretty new to this 
Asterisk stuff and I didn't realize there was a general context that you 
could do things like global includes.  Thanks, I'll give it a shot when 
I'm back in the office on Tuesday.


Peter:  No need to be an ass about it, pal.  Not all of us are as adept 
at this as you are.


Jay

Marco Mouta wrote:

So the #include could be made just after the [general] section o
extensions.conf? outside of any specific context, i think this was the
question.



On 9/4/06, Peter Bowyer <[EMAIL PROTECTED]> wrote:


On 04/09/06, Jay Moore <[EMAIL PROTECTED]> wrote:

> Right, I guess I was wondering if it's possible to include a file
> without it being in a context.  The goal I wanted to achieve was to 
have

> as few contexts in the main extensions.conf file as possible.

Did you try it? It would take... perhaps 30 seconds? A minute if
you're a slow typist...

Yes, you can do this. #include is a literal text include, as the last
poster said.


--
Peter Bowyer
Email: [EMAIL PROTECTED]
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Re: [asterisk-users] Grandstream and H.264 !

[Carlos Chavez]

On Mon, 2006-09-04 at 16:26 -0300, Sergio (Red) wrote:
> hi,
> I´ need some help to implement the Grandstream GXV-3000 in my * 
> platform.  Someone know the state of H.264 Video Codec for Asterrisk??
> 
> Thanks!!!
> 
This has been answered multiple times in the last month.  Search the
list before posting.  

You have to use the SVN version of Asterisk if you wish to use H264.

-- 
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


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[asterisk-users] Grandstream and H.264 !

[Sergio (Red)]

hi,
I´ need some help to implement the Grandstream GXV-3000 in my * 
platform.  Someone know the state of H.264 Video Codec for Asterrisk??


Thanks!!!

p.D.: appreciate any help
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[asterisk-users] playback some digits to the caller from the callee (involves DTMF) prob

[umar tarar]

how come can i get or read DTMF digits from a callee(agent) & then playback them on the caller's channel

i am aware of the read() application, but the problem is that where shud i place it or how shuld i use it, because if i place it below the Dial()
 application then the moment Dial app terminates, it detroys the channels(i.e the call gets hungup) & leaving no callee channel for read()
 to read DTMF



then i tried another way using features.conf i.e. defining a feature to
read DTMF into a variable & then playing it back on either both of
the channels or anyone of them, herez my features.conf portion +
dialplan



-features.conf--

...

readFeature => #6,peer/callee,Read,var1

playbackFeature => #9,peer/callee,SayDigits,var1

---



extensions.conf

...

exten => 500 , 1 , Answer()

exten => 500, 2 , SetVar(DYNAMIC_FEATURES=readFeature#playbackFeature)

exten => 500 , 3 , Playback(connecting)

exten => 500 , 4 , Dial(SIP/100)

---



but whenever someone dials 500 & gets connected with the agent(100) & the agent(callee) presses #6 or 
#9, simply nothing happens.

and on CLI the command show features results this:



Builtin Feature..Default Current

===...=...=

Pickup...*8.*8

Blind Transfer#..#1

Attended Transfer.*2

One Touch Monitor

Disconnect Call..*..*0



Dynamic Feature...Default Current

===...=

(none)



Call parking



Parking extension   :   700

Parking context :   parkedcalls

Parked call extensions: 701-750



it seems that the features that i defined are not even getting registered


somebody plz suggest me if i am wrong somewhere doing that, or plz
try giving me any newer solution to get DTMF digits durring a call or a
way to acheive the overall objective i.e. playback some digits to the
caller from the callee
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[asterisk-users] Astbill DIALSTRING doesn't work

[Sebastian Milioto]

Hi all,

Im trying to setup Astbill in my Asterisk box, but I'm having some problems.
First I obtain the following when I want to make a call:

   -- Executing AGI("SIP/71423-081b9010",
"agiastar.agi|called_number") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/agiastar.agi
   -- SIP Seeding peer from astdb: '71423' at
[EMAIL PROTECTED]:5060 for 3600
   -- AGI Script agiastar.agi completed, returning 0
   -- Executing Dial("SIP/71423-081b9010", "") in new stack
Sep  4 15:23:25 WARNING[4224]: app_dial.c:781 dial_exec_full: Dial
requires an argument (technology/number)
 == Spawn extension (mycontext, called_number, 3) exited non-zero on
'SIP/71423-081b9010'
   -- Executing DeadAGI("SIP/71423-081b9010", "agistardead.agi") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/agistardead.agi
   -- AGI Script agistardead.agi completed, returning 0
   -- SIP Seeding peer from astdb: '71423' at
[EMAIL PROTECTED]:5060 for 3600

In extensions.conf  I have this:


exten => _XX,1,AGI(agiastar.agi,${EXTEN})
exten => _XX,2,Dial(${DIALSTRING})

I think I'm not getting nothing from DIALSTRING. How can I check that,
how can I resolve it?.
I tryed changing the second line bye for this:

exten => _XX,3,Dial(SIP/[EMAIL PROTECTED],90,Ttr)

and it starting work. At this point, I can make calls, but the billing
seems doesn't work. So, I think it is because I supressed the
DIALSTRING line.
Can anybody helpme with that? I cant find a manual or technical
information about astbill, except a few lines y Wiki. Is there
something like that?

Thanks very much in advance,

Sebastian
e-mail:[EMAIL PROTECTED]
msn:[EMAIL PROTECTED]
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Re: [asterisk-users] Roundrobin not working on PRI

[Andres]

Zeeshan Zakaria wrote:

Can somebody send me a sample from their extension.conf to do the 
above mentioned thing, i.e. handling DIDs on PRI. This is the first 
time I am dealing with PRI, previously I always used SIP DIDs and had 
no problem at all. 


There is nothing fancy or misterious about it.  If you are receiving the 
last 4 digits of the DID then this will dial a SIP Phone with the same 4 
digits.


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

or if you prefer an auto-attendant then:

exten => _,1,Background(welcome_message)
exten => _,2,Background(more messages...)

--
Andres


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[asterisk-users] blf aastra 9133i working but can't pickup calls

[Jean-Louis curty]

Hi,I'm trying to get the blf / pickup working properly on the aastra 9133i,I read the wiki voip-info.org for the setup,setup is working fine for the snom, it works also for the aastra ( the light is flashing when a call comes in on another phone ) but I can't pickup the call ... when I press the prog key corresponding the extension I want to pickup, it just dial the extensions like a new call instead of the picking up
any idea ?jean-louis
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Re: [asterisk-users] File structure question

[Marco Mouta]

So the #include could be made just after the [general] section o extensions.conf? outside of any specific context, i think this was the question.On 9/4/06, 
Peter Bowyer <[EMAIL PROTECTED]> wrote:
On 04/09/06, Jay Moore <[EMAIL PROTECTED]> wrote:> Right, I guess I was wondering if it's possible to include a file> without it being in a context.  The goal I wanted to achieve was to have
> as few contexts in the main extensions.conf file as possible.Did you try it? It would take... perhaps 30 seconds? A minute ifyou're a slow typist...Yes, you can do this. #include is a literal text include, as the last
poster said.--Peter BowyerEmail: [EMAIL PROTECTED]___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] File structure question

[Peter Bowyer]

On 04/09/06, Jay Moore <[EMAIL PROTECTED]> wrote:


Right, I guess I was wondering if it's possible to include a file
without it being in a context.  The goal I wanted to achieve was to have
as few contexts in the main extensions.conf file as possible.


Did you try it? It would take... perhaps 30 seconds? A minute if
you're a slow typist...

Yes, you can do this. #include is a literal text include, as the last
poster said.


--
Peter Bowyer
Email: [EMAIL PROTECTED]
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Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK

[Elpidio Ramos]

But the soft phones have dynamic ip addresses. I have read this is why we use host=dynamic and nat=yes.Carlos Chavez <[EMAIL PROTECTED]> wrote:  On Mon, 2006-09-04 at 09:49 -0700, Elpidio Ramos wrote:> This seems to be an easy-to-solve problem but it may be again my lask> of knowledge in linux:> > My linux fedora core 3 asterisk box has a public IP and a private IP> (two NIC)> > I got the ports open in fedora core 3 (5060 and 1 thru 3) for> both interfaces.> > I was able con connect my sip soft phone from a NAT connection inside> my network pointing to the public IP. > You do not have either the externip or externhost directives in yoursip.conf. If you are connecting from the outside you need to tellAsterisk the IP address or hostname of
 the outside connection.> -- Carlos Chavez PratsDirector de TecnologíaTelecomunicaciones Abiertas de México S.A. de C.V.Tel: +52-55-91169161 Ext 2001  Elpidio Ramos President RM International Services SA CV Web: http://www.ramosoft.com Mex:  +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax    +52 (55) 1755-6601 CellUSA: +1 (801) 494-1415 Office   +1 (240) 250-8264 Fax   +1 (801) 938-4740 Direct   ___
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Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK

[Elpidio Ramos]

Not sure if that is the case.  I don't get to hear callers from the outside but the can dial my extension within my lan.     Does it sound like a DTMF problem?     I would think they could not dial my extension if DTMF was involved.Justin Tunney <[EMAIL PROTECTED]> wrote:  On 9/4/06, Elpidio Ramos <[EMAIL PROTECTED]>wrote:> When attempting to do the same from outside my network (from my dsl> connection from home), I get to hear the asterisk auto attendant but not> able to send any sound from my laptop.ARE YOU SURE IT ISN'T A DTMFPROBLEM!!  Elpidio Ramos President RM International Services SA CV Web: http://www.ramosoft.com Mex:  +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax    +52 (55) 1755-6601 CellUSA: +1 (801) 494-1415 Office   +1 (240) 250-8264 Fax   +1 (801) 938-4740 Direct   ___
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[asterisk-users] missing pri connect (wwomera to pri)

[Rosario Pingaro]

I am using my asterisk box with an application 
linked through the WOOMERA channel. Asterisk bridge the woomera chennel to zap 
(sangoma aft104d) and vicecersa.
 
The strange think is that after some hours of hevy 
load asterisk miss some time to relay the connect message from woomera to 
pri.
The debug from woomera and pri is pretty easy, just 
no connect passed to pri..
 
If I run asterisk without priority I get the 
problem early, using priority I have some hours of regular work. This is very 
wired.
 
Is someone experiencing some issue? 
Does someone that has woomera knowledg help me to 
fix the issue?
 
Regards
 
Rosario 
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Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK

[Rich Adamson]

Elpidio Ramos wrote:
This seems to be an easy-to-solve problem but it may be again my lask of 
knowledge in linux:
 
My linux fedora core 3 asterisk box has a public IP and a private IP 
(two NIC)
 
I got the ports open in fedora core 3 (5060 and 1 thru 3) for 
both interfaces.
 
I was able con connect my sip soft phone from a NAT connection inside my 
network pointing to the public IP.
 
When attempting to do the same from outside my network (from my dsl 
connection from home), I get to hear the asterisk auto attendant but not 
able to send any sound from my laptop.
 
This is my sip.conf file:
 
[general]
context=ramosoft  
allowguest=no
realm=ramosoft.com 
bindaddr=0.0.0.0  
bindport=5060   
srvlookup=yes   
pedantic=yes   
tos=184
tos=lowdelay   
maxexpirey=3600   
defaultexpirey=120  
disallow=all   
allow=ulaw   
allow=ilbc   
allow=gsm  
musicclass=default  
language=es   
relaxdtmf=yes   
rtptimeout=60   
rtpholdtimeout=300  
useragent=RamoSoftPBX  
regcontext=ramosoft
localnet=10.10.10.0/255.255.255.0 
rtcachefriends=yes   
 
[authentication]
 
[311]

type=friend
regexten=311
username=311
secret=311
callerid="Elpidio Ramos" <311>
host=dynamic
nat=yes
canreinvite=no
Is there anything I am missing here to get two way voice?
 
Thank you  in advance all


If you have two working nic's, then when the soft phone is on the inside 
of the network, it should register with the IP address of the inside nic.


When the soft phone is on the outside (eg Internet), then it should be 
registering with the IP address of the outside nic.


Any other combination is going to give you problems and particularly if 
you are using a firewall. The problems will be associated with basic 
layer-3 stuff and nating.



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Re: [asterisk-users] File structure question

[Jay Moore]

Tzafrir Cohen wrote:

On Thu, Aug 31, 2006 at 03:52:00PM -0500, Jay Moore wrote:

I have a question on how I can better organize my .conf files.

I have 3 different groups of people who use my VoIP service. Let's call 
them 'Office', 'Factory' and 'Public'. In my Asterisk directory, I have 
created three folders: 'office', 'factory' and 'public', inside each of 
which has a sip.conf and an extensions.conf file with appropriate 
account and extension information.


Say, for example, I need to limit some users of the 'Public' group so 
they cannot make calls outside the building. Obviously I would create 
two separate contexts. One for users who can make calls outside the 
build, and one for users who cannot. I would then assign the appropriate 
context to each user.


Right now, I have each appropriate context defined in the main 
extensions.conf. What I'd like to do is reduce the clutter in 
extensions.conf and move each context into the extensions.conf in the 
appropriate subfolder. How do I tell the main extensions.conf file to 
include the other extensions.conf files without putting an #include 
 in a context of its own?


I hope what I've explained makes sense. If not, please ask questions and 
I'll try to answer.


#include is a verbatim text include. 


if extensions.conf has:


[main]
exten => aaa,1,Line1

#include "otherfile.conf"

exten => aaa,2,Line2

and othererfile.conf has:

exten => aaa,2,OtherLine1

[other]

exten => aaa,1,OtherLine2



You'll eventually get:



main: aaa: 
  1. Line1

  2. OtherLine1

other: aaa:
  1. OtherLine2
  2. Line2
 


Right, I guess I was wondering if it's possible to include a file 
without it being in a context.  The goal I wanted to achieve was to have 
as few contexts in the main extensions.conf file as possible.


Jay
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Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK

[Carlos Chavez]

On Mon, 2006-09-04 at 09:49 -0700, Elpidio Ramos wrote:
> This seems to be an easy-to-solve problem but it may be again my lask
> of knowledge in linux:
>  
> My linux fedora core 3 asterisk box has a public IP and a private IP
> (two NIC)
>  
> I got the ports open in fedora core 3 (5060 and 1 thru 3) for
> both interfaces.
>  
> I was able con connect my sip soft phone from a NAT connection inside
> my network pointing to the public IP. 
>  
You do not have either the externip or externhost directives in your
sip.conf.  If you are connecting from the outside you need to tell
Asterisk the IP address or hostname of the outside connection.

> 
-- 
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


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Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK

[Justin Tunney]

On 9/4/06, Elpidio Ramos <[EMAIL PROTECTED]> wrote:

When attempting to do the same from outside my network (from my dsl
connection from home), I get to hear the asterisk auto attendant but not
able to send any sound from my laptop.


ARE YOU SURE IT ISN'T A DTMF
PROBLEM!!
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[asterisk-users] ONE WAY VOICE ONLY IN ASTERISK

[Elpidio Ramos]

This seems to be an easy-to-solve problem but it may be again my lask of knowledge in linux:     My linux fedora core 3 asterisk box has a public IP and a private IP (two NIC)     I got the ports open in fedora core 3 (5060 and 1 thru 3) for both interfaces.     I was able con connect my sip soft phone from a NAT connection inside my network pointing to the public IP.      When attempting to do the same from outside my network (from my dsl connection from home), I get to hear the asterisk auto attendant but not able to send any sound from my laptop.     This is my sip.conf file:     [general]context=ramosoft    allowguest=no  realm=ramosoft.com  
 bindaddr=0.0.0.0  bindport=5060   srvlookup=yes   pedantic=yes   tos=184tos=lowdelay   maxexpirey=3600   defaultexpirey=120  disallow=all   allow=ulaw   allow=ilbc   allow=gsm  musicclass=default  language=es   relaxdtmf=yes   rtptimeout=60   rtpholdtimeout=300  useragent=RamoSoftPBX  regcontext=ramosoftlocalnet=10.10.10.0/255.255.255.0 rtcachefriends=yes        [authentication]     [311]type=friendregexten=311username=311secret=311callerid="Elpidio Ramos" <311>host=dynamicnat=yescanreinvite=no  Is there anything I am
 missing here to get two way voice?     Thank you  in advance all___
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Re: [asterisk-users] Dropping extra frame of G.729 ?

[Hermann Wecke]

Noc Phibee wrote:

anyone know where i can solve this problems ? :


1) By doing a quick google search;
2) By reading previous posts regarding the same issue;
3) By disabling VAD (Voice activity detection) in your device.
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Re: [asterisk-users] FAX handling

[Marco Mouta]

Pls Post your Asterisk CLI when Fax is incoming.On 9/4/06, [EMAIL PROTECTED] <
[EMAIL PROTECTED]> wrote:I read on this list not so long ago that you should only enable alaw.  I've
never tested this.Phil. "Jose Limeres" <[EMAIL PROTECTED] om>To
 Sent by:  "Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion" [EMAIL PROTECTED]
  m.com  cc   Subje
 ct
 04/09/2006 15:35  [asterisk-users] FAX handling Please respond to  Asterisk Users  Mailing List -  Non-CommercialDiscussion
 <[EMAIL PROTECTED] ists.digium.com>Hi all,I am using asterisk 1.2.10 BRI stuffed 0.3.0-PRE-1s with zaptel 1.2.8
 andwe are trying to have FAX receiving working inone of the BRI lines.No problem with FAX transmissions but we can not receive. I have configuredin zapata.conf faxdetect=both (tx and rx).FAX machine is connected to one FXS port on a PAP2 with G711a and no echo
cancelation configured. When the FAX arrives at theFAX machine, they start negotiating but then it stops as if the format isnot recognized by the Fax machine as a valid fax.Does anyone have a similar configuration working?
Bests,Jose Limeres___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] FAX handling

[phil . dawson]

I read on this list not so long ago that you should only enable alaw.  I've
never tested this.


Phil.




   
 "Jose Limeres"
 <[EMAIL PROTECTED] 
 om>To 
 Sent by:  "Asterisk Users Mailing List -  
 asterisk-users-bo Non-Commercial Discussion"  
 [EMAIL PROTECTED]
 m.com  cc 
   
   Subject 
 04/09/2006 15:35  [asterisk-users] FAX handling   
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 <[EMAIL PROTECTED] 
 ists.digium.com>  
   
   




Hi all,

I am using asterisk 1.2.10 BRI stuffed 0.3.0-PRE-1s with zaptel 1.2.8 and
we are trying to have FAX receiving working in
one of the BRI lines.

No problem with FAX transmissions but we can not receive. I have configured
in zapata.conf faxdetect=both (tx and rx).
FAX machine is connected to one FXS port on a PAP2 with G711a and no echo
cancelation configured. When the FAX arrives at the
FAX machine, they start negotiating but then it stops as if the format is
not recognized by the Fax machine as a valid fax.

Does anyone have a similar configuration working?

Bests,

Jose Limeres___
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[asterisk-users] Dropping extra frame of G.729 ?

[Noc Phibee]

Hi

anyone know where i can solve this problems ? :

Sep  4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping 
extra frame of G.729 since we already have a VAD frame at the end
Sep  4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping 
extra frame of G.729 since we already have a VAD frame at the end
Sep  4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping 
extra frame of G.729 since we already have a VAD frame at the end
Sep  4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping 
extra frame of G.729 since we already have a VAD frame at the end
Sep  4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping 
extra frame of G.729 since we already have a VAD frame at the end
Sep  4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping 
extra frame of G.729 since we already have a VAD frame at the end
Sep  4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping 
extra frame of G.729 since we already have a VAD frame at the end



Thanks

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[asterisk-users] Submenus

[Mir]

Hello
I'm doing an IVR-service, where pilot can check metar (airport weather
information), they enter the 4 letter airport code on their phone, and
get the metar read back by text-to-speech.


[Metar]

exten => 1,1,answer

exten => 1,2,Background(Met_welcome)

exten => 1,3,set(airport="")2,Background(Met_welcome)

exten => 1,3,set(airport="")3,set(airport="")

exten => 1,n,Background(Met_Instructions)

(When they press an airportcode, I set a variable)

exten => 3575,1,set(airport=ekrk)

exten => 3575,n,goto(Metar,s,1)

exten => 3598,1,set(airport=ekyt)

and so on

exten => 3575,n,goto(Metar,s,1)

exten => 3598,1,set(airport=ekyt)

and so on

exten => 3598,1,set(airport=ekyt)

and so on
In the S extension, I do all of the processing .
My problem is that some of the airports has the same code, for
instance EKCH wich is entered by pressing 3524, but EKAH has the same
digits, so I need to make a sub-menu, where I can ask the caller to
press 1 for EKCH or 2 for EKAH.
Right now, I do like this:


exten => 3524,1,background(Met_ch_bi_ah)background(Met_ch_bi_ah)

exten => 4,1,set(airport=ekch)

exten => 4,n,goto(Metar,s,1)

exten => 6,1,set(airport=ekbi)

exten => 6,n,goto(Metar,s,1)1,set(airport=ekch)

exten => 4,n,goto(Metar,s,1)

exten => 6,1,set(airport=ekbi)

exten => 6,n,goto(Metar,s,1)6,1,set(airport=ekbi)

exten => 6,n,goto(Metar,s,1)6,n,goto(Metar,s,1)

This is not a very good solution, if a user by mistake press 4 in the
main loop, it goes right to EKCH metar, instead of informing the user
that it is an invalid code.

So what I need is a way to make a submenu, that is only "visible" when
needed, and can set the airport variable.

How do I do this?



Michael
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RE: [asterisk-users] Asterisk 1.2.11 and # key

[David Gagnon]

Are you sure this is not
because of the dynamic features in features.conf ?

By default, # is defined
for the transfer feature.

 

David

 









De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Michael Strelnikov
Envoyé : 4 septembre 2006
09:53
À :
asterisk-users@lists.digium.com
Objet : [asterisk-users]
Asterisk 1.2.11 and # key



 

Hello,

   Does anybody have problems with recognition of the hash (#) key
with * 1.2.11? It seams that after pressing # the call is in a progress but no
data is sent.

Thanks in advance,
Michael






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RE: [asterisk-users] Blind transfer 3/4 digits

[David Gagnon]

Ronald,

Like someone already told you, you should explain more clearly the
way you try to transfer, we need more details on the procedure, using which
button on which phone. We need every detail to help you. This as nothing to
do with the way the dial plan is loaded, this is totally false.

I'm sure most of the people here don't understand how you try to
transfer.

David

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 4 septembre 2006 09:22
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits

Koopmann, Jan-Peter wrote:
> On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:
>
>   
>> try that way. However, I have doubts as well. If you are right, than
>> why snom phone does not have this problem? Would not here also the
>> first match count?   
>> 
>
> Because the transfer button on the SNOM is using a totally different
mechanism than sending # to Asterisk. On your snom configuration (like ours)
the phone does not start to create/send a SIP message until you hit "OK". At
that time the entire number is there and a complete SIP transfer is created.
Cool down a bit. The problem you are having is most probably just a dialplan
problem. It takes some time and experience to get those things right. No
need to yell here...
>   
What's happen to you guys? I am not yelling, just asking.
It is sure not a dialplan question! If it would be a dialplan question, 
than it would be for each dialing, but it isn't.

You mentioned SIP message and that makes me wonder! Are we not using 
here dtmf ?? that is in my opinion not a sip message, isn't it?
If it is a sequence of "tones", than why is it different if it is in a 
string (like snom) or another phone, with single tones?
If we understand this part, than is the question, where can I turn on 
the system to take a longer break between "tones" still as a string?

Back to the dialplan:
A Voip number can have different length of digits. Each number is seen 
as a complete "picture", and so a three digit and a four digit number is 
something different. While in the legacy telephony the digits are worked 
down one by one and if there is no more use of the digits, they are just 
garbage and will be not used. Unlike in VoIP, where you can have a three 
digit number and if you dial four digit, than it is a WRONG number  
I just verified that: I dialed from 601 to  61522, however, 61522 does 
not exist, but 615 exists. Guess what? I get a busy tone! That should 
proof my thoughts (and that without yelling, ... hehehehe)

bye

Ronald
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[asterisk-users] Handling Disconnection Causes

[Rafael J. Risco G.V.]

hi
I am sending all my 'not available' prefixes from h323 gnugk to an
asterisk box listening h323 in port 1721 (using oh323 module) to
handle disconnection causes based in this document:

Example macro for handling hangupcause:
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause

so in my extension.conf i put this:
...
[voip-h323]

;; All calls to Cause Code 34!!
exten => _.,1,Macro(dial-result|34)

[macro-dial-result]
; Handles Disconnect Cause Codes (see link above for example)
...

It works but I dont know why i'am getting cause 42 instead of 34 (No
circuit available) in gnugk, I think my termination parter can
re-route with cause 42 because its almost the same (switch congested
or overloaded) but i
would like to understand this anyway and force to get 34 cause code.

thank you
rafael
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[asterisk-users] PAP2-NA + Asterisk

[joy panlilio]

Lists,        Hi! good day, i have been task to install/replace our legacy pbx with asterisk, most of the people here in our office was amaze on how asterisk really works, except for i'm having a problem with dtmf detection on PAP2 ATA converter     here is the call flow     ATA -> SIP -> TDM400 -> PSTN dtmf detection not working as expected, the other asterisk pbx on the other company decode it with.     whenever i press extension 103 the other asterisk server decode it with 110 not 103     it doubles the 1 then omit the 3 :(     sip.conf snippet     [100]     type=friend  host=dynamic  disallow=all  allow=ulaw  allow=gsm  dtmfmode=rfc2833  username=100 
 secret=100  mailbox=100  nat=yes  canreinvite=no     thanks in advance     Joy                         
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Re: [asterisk-users] Any Hardphone with VPNClient embedded?

[Cory Andrews]

Please be aware that from a future support standpoint, you may be a bit 
limited with Zultys.  Their future seems very uncertain they have recently 
just about ceased operations and let the majority of their employees go.


Cory J Andrews

voice - 800.398.VoIP X3402
email - [EMAIL PROTECTED]
AIM - B2CORY
- Original Message - 
From: "Leo Ann Boon" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Monday, September 04, 2006 10:35 AM
Subject: Re: [asterisk-users] Any Hardphone with VPNClient embedded?



Marco Mouta wrote:

Hi all,

Does any of you knows an Hardphone with VPN client embedded?

Take a look at Zultys SIP phones. VPN enabled.

www.zultys.com

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Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel

[RR]

Hi Zoa,

thanks for responding. Ok, now where do I find this? I'm running
2.6.9-34.0.1 kernel. I tried doing a bit of search and it seems like
that the ability to change the frequency doesn't appear till 2.6.13.
Am I looking at the right thing? Any hints?
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Re: [asterisk-users] Any Hardphone with VPNClient embedded?

[Leo Ann Boon]

Marco Mouta wrote:

Hi all,

Does any of you knows an Hardphone with VPN client embedded?

Take a look at Zultys SIP phones. VPN enabled.

www.zultys.com

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[asterisk-users] FAX handling

[Jose Limeres]

Hi all,I am using asterisk 1.2.10 BRI stuffed 0.3.0-PRE-1s with zaptel 1.2.8 and we are trying to have FAX receiving working in one of the BRI lines.

No problem with FAX transmissions but we can not receive. I have configured in zapata.conf faxdetect=both (tx and rx).
FAX machine is connected to one FXS port on a PAP2 with G711a and no echo cancelation configured. When the FAX arrives at the FAX machine, they start negotiating but then it stops as if the format is not recognized by the Fax machine as a valid fax.

Does anyone have a similar configuration working?Bests,Jose Limeres
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Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)

[Marco Mouta]

Done,I've created ~/.vimrc file and inside this file:syntax onthks once moreOn 9/4/06, Marco Mouta <
[EMAIL PROTECTED]> wrote:BTW Could you tell me how to i make it load this option by default everytime?
On 9/4/06, Marco Mouta <

[EMAIL PROTECTED]> wrote:Just Great!What was missing is

:syntax onNow perfect! Thks guys! In fact i couldn't find this basic step any where except here. Ok I'm a newbie, but it will help others if is written in the tutorials. I'll look for wiki to post this there.
On 9/4/06, Victor Toofic <

[EMAIL PROTECTED]> wrote:
On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote:>> I've made vim /etc/asterisk/extensions_custom.conf then :set> syntax=asterisk, and nothing happens. No errors no warnings and also no> highlight syntax...
>Hi!!I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and:set filetype=asterisk:syntax on (optionally)works fine for me.--Víctor Toofic___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta

-- Com os melhores cumprimentos,Marco Mouta

-- Com os melhores cumprimentos,Marco Mouta
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Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)

[Marco Mouta]

BTW Could you tell me how to i make it load this option by default everytime?On 9/4/06, Marco Mouta <
[EMAIL PROTECTED]> wrote:Just Great!What was missing is
:syntax onNow perfect! Thks guys! In fact i couldn't find this basic step any where except here. Ok I'm a newbie, but it will help others if is written in the tutorials. I'll look for wiki to post this there.
On 9/4/06, Victor Toofic <
[EMAIL PROTECTED]> wrote:
On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote:>> I've made vim /etc/asterisk/extensions_custom.conf then :set> syntax=asterisk, and nothing happens. No errors no warnings and also no> highlight syntax...
>Hi!!I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and:set filetype=asterisk:syntax on (optionally)works fine for me.--Víctor Toofic___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta

-- Com os melhores cumprimentos,Marco Mouta
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[asterisk-users] Asterisk 1.2.11 and # key

[Michael Strelnikov]

Hello,   Does anybody have problems with recognition of the hash (#) key with * 1.2.11? It seams that after pressing # the call is in a progress but no data is sent.Thanks in advance,Michael
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Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)

[Marco Mouta]

Just Great!What was missing is:syntax onNow perfect! Thks guys! In fact i couldn't find this basic step any where except here. Ok I'm a newbie, but it will help others if is written in the tutorials. I'll look for wiki to post this there.
On 9/4/06, Victor Toofic <[EMAIL PROTECTED]> wrote:
On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote:>> I've made vim /etc/asterisk/extensions_custom.conf then :set> syntax=asterisk, and nothing happens. No errors no warnings and also no> highlight syntax...
>Hi!!I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and:set filetype=asterisk:syntax on (optionally)works fine for me.--Víctor Toofic___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta
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Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)

[Victor Toofic]

On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote:
> 
> I've made vim /etc/asterisk/extensions_custom.conf then :set
> syntax=asterisk, and nothing happens. No errors no warnings and also no
> highlight syntax...
> 

Hi!!

I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and

:set filetype=asterisk
:syntax on (optionally)

works fine for me.

--
Víctor Toofic
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[asterisk-users] usereqphone=yes seems to don't work

[Alexandre VERNIOL]

Hi all,

I'm looking at a function to add "user=phone" into sip's trame. So I 
include usereqphone=yes into the [general] of my sip.conf. But it seems 
to don't work; so is there an other way to add this "user=phone" through 
* ?


Cheers,

--
Alexandre VERNIOL
Technicien VoIP Revendeur Directcentrex
Hotline : 0892 46 05 12
Ticket : http://ticket.directcentrex.com 
www.directcentrex.com

www.frontier.fr
www.directnom.com


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Re: [asterisk-users] Blind transfer 3/4 digits

[Ronald Wiplinger]

Koopmann, Jan-Peter wrote:

On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:

  

try that way. However, I have doubts as well. If you are right, than
why snom phone does not have this problem? Would not here also the
first match count?   



Because the transfer button on the SNOM is using a totally different mechanism than 
sending # to Asterisk. On your snom configuration (like ours) the phone does not start to 
create/send a SIP message until you hit "OK". At that time the entire number is 
there and a complete SIP transfer is created. Cool down a bit. The problem you are having 
is most probably just a dialplan problem. It takes some time and experience to get those 
things right. No need to yell here...
  

What's happen to you guys? I am not yelling, just asking.
It is sure not a dialplan question! If it would be a dialplan question, 
than it would be for each dialing, but it isn't.


You mentioned SIP message and that makes me wonder! Are we not using 
here dtmf ?? that is in my opinion not a sip message, isn't it?
If it is a sequence of "tones", than why is it different if it is in a 
string (like snom) or another phone, with single tones?
If we understand this part, than is the question, where can I turn on 
the system to take a longer break between "tones" still as a string?


Back to the dialplan:
A Voip number can have different length of digits. Each number is seen 
as a complete "picture", and so a three digit and a four digit number is 
something different. While in the legacy telephony the digits are worked 
down one by one and if there is no more use of the digits, they are just 
garbage and will be not used. Unlike in VoIP, where you can have a three 
digit number and if you dial four digit, than it is a WRONG number  
I just verified that: I dialed from 601 to  61522, however, 61522 does 
not exist, but 615 exists. Guess what? I get a busy tone! That should 
proof my thoughts (and that without yelling, ... hehehehe)


bye

Ronald
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[asterisk-users] includes in realtime ??

[Benjamin Jacob]

Hello ppl,
Is it possible to include contexts in the RealTime scenario??
If not, wots the work around??

Thanks in advance.
Ben.
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[Asterisk-Users] External calls from Asteris over a legacy Siemens BusinessPhone 250 PBX

[Llorenç Suau]

 Hello,The PBX Siemens BusinessPhone 250 uses the prefix 0
to make an external call(nationals, internationals,...) and I need to
make external calls from asterisk to this PBX, for if the IP provider
falls.With the internal calls I don't have problems, the PBX
make all without problems, but when the call coming from Asterisk, has
to be external it doesn't call, although I indicate the prefix in the
call, for example "0971539230", and it doesn't call to the number.
An bit of my dialplan to make the call across the PBXexten => _09.,1,Dial(Zap/g1/${EXTEN},20,tTr)exten => _09.,2,CongestionAny suggestions, to how I can make that the PBX receives correctly the call, PREFIX+number, to make the external call.
Thanks in advance,RegardsLlorenç
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Re: [asterisk-users] What I always get asked in SME * deployments

[Colin MacMillan]

This can be done with the 'hint' priority in the dialplan with the right hardware.Check out this link for an example:http://www.voip-info.org/wiki-Asterisk+phone+snom
look at section- SNOM SUBSCRIBE/NOTIFY support for monitoring extension statesOn 9/3/06, Dovid Bender <
[EMAIL PROTECTED]> wrote:






Some phones have the BLF feature. You can see on 
the phone who is and who is not on the phone. With the polycom's you need to get 
a side car. With the snom's you can use the buttons on the phone 
itself. 

  When ever we do a roll out of Asterisk in a small business environment 
  replacing an old key system or legacy PBX the receptionist always asks us, 
  "How do I know if someone is on a call before transferring them?". My typical 
  answer is "why do you need to know, just do an attended transfer and if they 
  can take the call they will, if they can't just tell the caller the person is 
  busy". If the receptionist insists on "knowing" we give them FOP.Has 
  anyone out there devised a better way to let a receptionist "know if someone 
  is on a call"?

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[asterisk-users] External calls from Asterisk over a Siemens(legacy) RDSI PBX

[Llorenç Suau]

 Hello,The PBX Siemens BusinessPhone 250 uses the prefix 0 to make an external call(nationals, internationals,...) and I need to make external calls from asterisk to this PBX, for if the IP provider falls.With the internal calls I don't have problems, the PBX make all without problems, but when the call coming from Asterisk, has to be external it doesn't call, although I indicate the prefix in the call, for example "0971539230", and it doesn't call to the number.
An bit of my dialplan to make the call across the PBXexten => _09.,1,Dial(Zap/g1/${EXTEN},20,tTr)exten => _09.,2,CongestionAny suggestions, to how I can make that the PBX receives correctly the call, PREFIX+number, to make the external call.
Thanks in advance,RegardsLlorenç
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[asterisk-users] Any Hardphone with VPNClient embedded?

[Marco Mouta]

Hi all,Does any of you knows an Hardphone with VPN client embedded? -- Best regards,Marco Mouta
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Re: [asterisk-users] Zaptel-1.2.8 compile problem

[yusuf]

Vidura Senadeera wrote:



Hi,
 
I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. 
asterisk version and libpri versions are

1.2.11 and 1.2.3.
 
Please refer the attached txt files for Linux version information and 
output of zaptel compile.
 
I will be highly appreciated that any one can help me on this regard.


--
Thanks & Regards,
Vidura B. Senadeera.


--
Thanks & Regards,
Vidura B. Senadeera.
--
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cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o gendigits.o 
gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits > tones.h
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWAREmakefw.c   -o makefw
./makefw tormenta2.rbt tor2fw > tor2fw.h
./makefw pciradio.rbt radfw > radfw.h
ZAPTELVERSION="1.2.8" build_tools/make_version_h > version.h.tmp
if cmp -s version.h.tmp version.h ; then echo; else \
mv version.h.tmp version.h ; \
fi

rm -f version.h.tmp
cc fw2h.c   -o fw2h
./fw2h OCT6114-128D.ima vpm450m_fw.h
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE
-DBUILDING_TONEZONE -o zonedata.lo zonedata.c
cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE
-DBUILDING_TONEZONE -o tonezone.lo tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o libtonezone.a -lm
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o torisatool.o 
torisatool.c
cc -o torisatool torisatool.o
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o ztmonitor.o 
ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o zttool.o 
zttool.c
cc -o zttool zttool.o -lnewt
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWAREzttest.c   -o zttest
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o fxotune.o 
fxotune.c
cc -o fxotune fxotune.o -lm
/lib/modules/2.6.9-34.EL/build
make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/home/vidura/zaptel-1.2.8 modules
make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686'
  CC [M]  /home/vidura/zaptel-1.2.8/zaptel.o
make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-i686'




Linux version 2.6.9-34.EL ([EMAIL PROTECTED]) (gcc version 3.4.5 20051201 (Red 
Hat 3.4.5-2)) #1 Wed Mar 8 00:07:35 CST 2006




most probably:

http://bugs.digium.com/view.php?id=6425

--
thanks,
yusuf

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Re: [asterisk-users] Zaptel-1.2.8 compile problem

[Tzafrir Cohen]

On Mon, Sep 04, 2006 at 03:14:50PM +0600, Vidura Senadeera wrote:
> Hi,
> 
> I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk
> version and libpri versions are
> 1.2.11 and 1.2.3.
> 
> Please refer the attached txt files for Linux version information and output
> of zaptel compile.

Please incluude the errors as well:

make>log 2>&1

> 
> I will be highly appreciated that any one can help me on this regard.

So please don't cross-post.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] Architecture:MainOffice(AstServer)-SmallOffices(ATA.-LegacyPBX)

[Marco Mouta]

Hi,I'm planning a solution to establish a connection between Main office company, and let's say 15 small offices:Plan:1- Asterisk Server @ Main Office (connected to main office legacy pbx)2- Linksys SPA 3000 to install on every small office.
Idea behind SPA 3000, the main goal is to keep every user with their traditional phone and just connect SPA3000 to the legacy pbx of the small office and then route the calls from lecagy PBX to MainOffice Asterisk Server via voIP.
Then SPA3000 would be used as a low cost solution to  allow any user from the small office to call Main office Company. This way with only one or two ATA per small office i would be able to connected every one with main office with very lowcost price
I would like to hear from you any suggestions or ideas, is this acceptable for a productions system?-- Com os melhores cumprimentos,Marco Mouta
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Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)

[Marco Mouta]

Hi Tzafrir,I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax...Thks,
On 9/2/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
On Fri, Sep 01, 2006 at 03:03:39PM +0100, Marco Mouta wrote:> Hi all,>> I've just installed vim70, looking for vim syntax highlighting( for> Asterisk.conf files) ,> 
http://voip-info.org/tiki-index.php?page=vim+syntax+highlighting, and i> notice that both: asterisk.vim and filetype.vim  already refer asterisk> configurations.>> But unfortunately i couldn't get yet the highlight syntax working fine for
> my asterisk.conf files.>> Any one can help me?What happens if you run manually::set syntax=asterisk--Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]+972-50-7952406  jabber:[EMAIL PROTECTED]
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-- Com os melhores cumprimentos,Marco Mouta
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[asterisk-users] Zaptel-1.2.8 compile problem

[Vidura Senadeera]

Hi, 
 
I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk version and libpri versions are
1.2.11 and 1.2.3. 
 
Please refer the attached txt files for Linux version information and output of zaptel compile.
 
I will be highly appreciated that any one can help me on this regard.-- Thanks & Regards,Vidura B. Senadeera. -- Thanks & Regards,Vidura B. Senadeera. 
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o gendigits.o 
gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits > tones.h
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWAREmakefw.c   -o makefw
./makefw tormenta2.rbt tor2fw > tor2fw.h
./makefw pciradio.rbt radfw > radfw.h
ZAPTELVERSION="1.2.8" build_tools/make_version_h > version.h.tmp
if cmp -s version.h.tmp version.h ; then echo; else \
mv version.h.tmp version.h ; \
fi

rm -f version.h.tmp
cc fw2h.c   -o fw2h
./fw2h OCT6114-128D.ima vpm450m_fw.h
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE
-DBUILDING_TONEZONE -o zonedata.lo zonedata.c
cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE
-DBUILDING_TONEZONE -o tonezone.lo tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o libtonezone.a -lm
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o torisatool.o 
torisatool.c
cc -o torisatool torisatool.o
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o ztmonitor.o 
ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o zttool.o 
zttool.c
cc -o zttool zttool.o -lnewt
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWAREzttest.c   -o zttest
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE   -c -o fxotune.o 
fxotune.c
cc -o fxotune fxotune.o -lm
/lib/modules/2.6.9-34.EL/build
make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/home/vidura/zaptel-1.2.8 modules
make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686'
  CC [M]  /home/vidura/zaptel-1.2.8/zaptel.o
make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-i686'
Linux version 2.6.9-34.EL ([EMAIL PROTECTED]) (gcc version 3.4.5 20051201 (Red 
Hat 3.4.5-2)) #1 Wed Mar 8 00:07:35 CST 2006
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[asterisk-users] "Asterisk Developers Mailing List" ,

[Vidura Senadeera]

Hi, 
 
I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk version and libpri versions are
1.2.11 and 1.2.3. 
 
Please refer the attached txt files for Linux version information and output of zaptel compile.
 
I will be highly appreciated that any one can help me on this regard.-- Thanks & Regards,Vidura B. Senadeera. 
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Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel

[Zoa]

Check the timer frequency, it might have a different setting on the two 
kernels.


RR wrote:


Hi all, (2nd attempt)

this is probably a weird question and something I'm not doing right
but I got this bizarre thing going on here. When I boot the system
with the SMP kernel and compile (*) with the smp kernel source
(actually even if I don't compile, but as long as I boot into the SMP
kernel), I get this problem where calling into the system, say to
check my voicemail, the prompt playback continously changes tempo. The
prompts are played in slow-motion, and then it speeds up to its normal
speed, then goes back in slow-mo and so on. It happens (I think) at
constant periods. Only the tempo changes, not the pitch of the prompt.

Does anyone have any idea what could be happening? I have watched
"top"constantly but haven't noticed anything bizarre in terms of CPU
or Mem usage. This is on a 100mbps LAN with nothing much else on it.
And it only happens when it's booted into the smp kernel. So it's
something to do with smp, thread scheduling, or some buffer BUT I
don't know what exactly.

All you champs out there, esp. the asterisk-dev people, any light you
can shed on this?

Thanks much
\R
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[asterisk-users] Prompts playback changing tempo w/ SMP kernel

[RR]

Hi all, (2nd attempt)

this is probably a weird question and something I'm not doing right
but I got this bizarre thing going on here. When I boot the system
with the SMP kernel and compile (*) with the smp kernel source
(actually even if I don't compile, but as long as I boot into the SMP
kernel), I get this problem where calling into the system, say to
check my voicemail, the prompt playback continously changes tempo. The
prompts are played in slow-motion, and then it speeds up to its normal
speed, then goes back in slow-mo and so on. It happens (I think) at
constant periods. Only the tempo changes, not the pitch of the prompt.

Does anyone have any idea what could be happening? I have watched
"top"constantly but haven't noticed anything bizarre in terms of CPU
or Mem usage. This is on a 100mbps LAN with nothing much else on it.
And it only happens when it's booted into the smp kernel. So it's
something to do with smp, thread scheduling, or some buffer BUT I
don't know what exactly.

All you champs out there, esp. the asterisk-dev people, any light you
can shed on this?

Thanks much
\R
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[asterisk-users] No more linux/compiler.h in Fedora Core 6.

[William F. Acker WB2FLW +1-303-722-7209]

Hi all,

 When the glibc-kernheaders package disappeared from FC6 test2, builds 
of Asterisk failed with reference to a missing linux/compiler.h, included 
from channels/chan_phone.c.

I applied the following patch

diff -urN asterisk-1.2.11.orig/channels/chan_phone.c 
asterisk-1.2.11/channels/chan_phone.c
--- asterisk-1.2.11.orig/channels/chan_phone.c  2006-08-05 05:08:50.0 
+
+++ asterisk-1.2.11/channels/chan_phone.c   2006-09-04 07:23:40.0 
+
@@ -37,9 +37,6 @@
 #include 
 /* Still use some IXJ specific stuff */
 #include 
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(2,6,0)
-# include 
-#endif
 #include 

 #include "asterisk.h"

 Everything builds fine, but I haven't a clue how to test it.  I'm 
inclined to think that if it builds, it should be OK.  What y'all think?



  TIA.

--
Bill in Denver
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Re: [asterisk-users] Fax with asterisk?

[DRi]

[EMAIL PROTECTED] wrote on 31.08.2006
05:41:52 PM:

> Matthias Fechner wrote:
> 
> >Hello Roger,
> >
> >* Roger Schreiter <[EMAIL PROTECTED]> [31-08-06 14:19]:
> >  
> >
> >>did google for asterisk and fax show no results?
> >>    
> >>
> >
> >yes I found spandsp but it will do everything in software.
> >Is it not a good idea to use my modem for the fax stuff?
> >  
> >
> Why would it not be a good idea to do things in software?
> 
Hi all,

software-solution would be a good idea... eg. spandsp/iaxmodem & hylafax
but is where a app_rxfax/app_txfax planned/available
for spandsp-0.0.3 ?
this would myke things much easier as handling with
hylafax, at least to receive fax.
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[asterisk-users] Re:sip giving problems, please help.

[Ma Zhiyong]

Yes, I also get these problems occasionally

Sep  4 17:44:49 WARNING[1365]: channel.c:787 channel_find_locked: Avoided 
deadlock for '0x8224468', 10 retries!
Sep  4 17:44:49 WARNING[1364]: channel.c:787 channel_find_locked: Avoided 
deadlock for '0x8224468', 10 retries!

Sep  4 17:52:15 WARNING[1597]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): 
syntax error: syntax error, unexpected TOK_LT, expecting TOK_MINUS or TOK_COMPL 
or TOK_LP or TOKEN; Input:
 < 60
 ^
Sep  4 17:52:15 WARNING[1597]: ast_expr2.fl:187 ast_yyerror: If you have 
questions, please refer to doc/README.variables in the asterisk source.
Sep  4 17:52:15 WARNING[1597]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): 
syntax error: syntax error, unexpected TOK_LT, expecting TOK_MINUS or TOK_COMPL 
or TOK_LP or TOKEN; Input:
 < 120
 ^
Sep  4 17:52:15 WARNING[1597]: ast_expr2.fl:187 ast_yyerror: If you have 
questions, please refer to doc/README.variables in the asterisk source.


Sep  4 18:50:49 ERROR[1290]: chan_sip.c:11346 sipsock_read: We could NOT get 
the channel lock for SIP/gw-442744f0! 
Sep  4 18:50:49 ERROR[1290]: chan_sip.c:11347 sipsock_read: SIP MESSAGE JUST 
IGNORED: BYE 
Sep  4 18:50:49 ERROR[1290]: chan_sip.c:11348 sipsock_read: BAD! BAD! BAD!
Sep  4 18:50:51 ERROR[1290]: chan_sip.c:11346 sipsock_read: We could NOT get 
the channel lock for SIP/gw-442744f0! 
Sep  4 18:50:51 ERROR[1290]: chan_sip.c:11347 sipsock_read: SIP MESSAGE JUST 
IGNORED: BYE 
Sep  4 18:50:51 ERROR[1290]: chan_sip.c:11348 sipsock_read: BAD! BAD! BAD!___
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