Re: [asterisk-users] The S word: Asterisk security

[randulo]

Aside from the 5g phone that will come out as soon as you plunk down
$300 for the 3g ($800 if you calculate your 2 year contract
obligation), don't forget to join us today for  "The S Word: Security"

Most of you on this list will be more qualified than I am to discuss
or even list the issues involved, but I would start with these:

* What are the principal risks?
 DoS
 Fraudulent usage of your minutes
 Compromising your user accounts (example, getting all the emails, CID, etc)
 Making your life miserable in various ways through resource abuse

* What's wrong with running as root?

* How to lock down your server
 Denying access using standard *nix tools
 Authentication
 Checking against known attackers

Those are just a few ideas. Please join us for the call this Friday 4th of July.

See http://VoipUsersConference.org

IRC.Freenode.net #voip-users-conference

PSTN;: Call (724) 444-7444 and enter 22622# 1#

Dial(SIP/[EMAIL PROTECTED],60,D(22622#${YOUR_PIN}#)) ; by default
your PIN is 1#

ts.x2z.eu resolves to the above IP

http://food4wine.ning.com has news, forums, blogs, etc

http://voipuserstv.com has videos of Asterisk Tag and other asterisk
and voip stuff

RSS http://feeds.feedburner.com/AstUser

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Re: [asterisk-users] DIDs required of Paris and Gottenburg Sweden

[randulo]

On Fri, Jul 4, 2008 at 7:25 AM, Kashif Naeem <[EMAIL PROTECTED]> wrote:
> We need the DIDs of Paris and Gottenburg, Sweden. Can anyone provide ?

IdeaSIP.com can provide this. I don't know their rates, see the site
for that. Their call quality is excellent.

/r

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Re: [asterisk-users] DIDs required of Paris and Gottenburg Sweden

[Moe Navid]

Hi Kashif,

I use didx.net you can get did numbers in many countries

On Jul 3, 2008, at 10:25 PM, Kashif Naeem wrote:


Hello All,

We need the DIDs of Paris and Gottenburg, Sweden. Can anyone  
provide ? Please reply with rates.


Regards,


--
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com

Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766

Email: [EMAIL PROTECTED]
MSN: [EMAIL PROTECTED]
Gmail: [EMAIL PROTECTED]
Skype: kashif.naeem

302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.  
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[asterisk-users] DIDs required of Paris and Gottenburg Sweden

[Kashif Naeem]

Hello All,

We need the DIDs of Paris and Gottenburg, Sweden. Can anyone provide ?
Please reply with rates.

Regards,


-- 
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com

Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766

Email: [EMAIL PROTECTED]
MSN: [EMAIL PROTECTED]
Gmail: [EMAIL PROTECTED]
Skype: kashif.naeem

302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.
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Re: [asterisk-users] How to get a clean, basic configuration?

[Octavio Ruiz]

On Fri, Feb 22, 2008 at 10:15 AM, Tilghman Lesher
<[EMAIL PROTECTED]> wrote:
> On Friday 22 February 2008 04:55:13 Vincent wrote:
>> On Thu, 21 Feb 2008 22:04:41 +0200, Tzafrir Cohen wrote:
>> >For the brave: use modules.conf without 'autoload = yes'. This promises
>> >you many hours of interesting dialplan debugging. Enjoy.
>>
>> Yup, that's what I anticipated, which is why I was asking which
>> modules I can _safely_ remove without breaking things :-)
> Generally, the rule is that you can't remove any of the res_*

Based on experience you almost always need res_features.so, otherwise
you will experience crashes.
app_dial and many chan_* depends on it.


-- 
Octavio H. Ruiz Cervera
Tel.: (+52 55) 8590-9000 Ext. 7016
Mobile: (+52 1 55) 14-087790
Mobile: (+52 1 55) 41-351242

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Re: [asterisk-users] Spoofing CID

[Alex Balashov]

Alexander Lopez wrote:

> I may create an IVR Hell for them, so that I can transfer the calls to, 
> Hey, Its their dime.

If you do end up going that route, please share the details, and 
possibly the code.  We could all benefit from a good IVR from hell.  I 
certainly could use one.

No, I'm not being sarcastic.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Spoofing CID

[Sherwood McGowan]

Robert Goodyear wrote:
> So who out there is aware of the FCC or FTC laws concerning spoofing 
> caller ID for deceptive purposes? There's a collection agency out 
> there who has my wife's name crossed with someone else's, and they are 
> picking numbers from our area code to present themselves as when 
> calling us (over and over and over.) I of course would like to turn 
> this around on them as they refuse to believe who we say we are.
> 
>
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Quick note, Congress is attempting to make this practice illegal:
http://www.engadget.com/2007/06/29/congress-looking-to-make-caller-id-spoofing-illegal/
http://www.govtrack.us/congress/bill.xpd?tab=summary&bill=s110-704



-- 
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]


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Re: [asterisk-users] Spoofing CID

[Alexander Lopez]

Snip

> On Thursday 03 July 2008 15:54:36 Alexander Lopez wrote:
Snip

> > There is one Alex Lopez (NOT ME) here in Miami that owes a lot
of
> > people a lot of money. I get calls at all times of the day and
night,
> > they forge the number, and so what do they care about following the
FTC
> > rules
> >
> > I feel your pain. I may create an IVR Hell for them, so that I can
> > transfer the calls to, Hey, Its their dime.
> 
> A registered letter, with return receipt is the best way to deal with
> these
> clowns.  If that doesn't work, your only alternative is to file a
lawsuit.
> There are some good online resources for this.  Filing a lawsuit is
> actually
> fairly inexpensive, but when you compare that to the expense they
incur
> simply to have a lawyer show up in court, unless this guy owes
thousands
> of dollars (enough to make it worth their while in court), they will
do
> everything in their power to avoid getting into that scenario.
> 
> Collection efforts are all about making you feel badly enough that you
> cave
> in to their demands.  Making them spend money to collect money is
exactly
> the opposite of what they want to do, and thus it's the most effective
way
> to
> get them to back down.
> 
> --
> Tilghman
> 
Snip

I stay quit most of the time and let them provide the sensitive data. I
never give them my Social but I will confirm that they have the wrong
one.

Problem with a certified letter is that they hang up before you can get
that info from them.

Another problem with this is that some calls may actually be phishing.
Fishing is big in Florida, but it seams that phishing is big all over
the world!!!


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Re: [asterisk-users] Spoofing CID

[Tilghman Lesher]

On Thursday 03 July 2008 15:54:36 Alexander Lopez wrote:
> Neither DHS nor FTC has any legislation on this. Florida house had a
> bill. Unfortunately, Collection agencies are deceptive by nature as most
> other options have been exhausted before an account goes to collections.
> I get the same thing here; they once even called me from a number that
> had my same last name!!
>
> There is one Alex Lopez (NOT ME) here in Miami that owes a lot of
> people a lot of money. I get calls at all times of the day and night,
> they forge the number, and so what do they care about following the FTC
> rules
>
> I feel your pain. I may create an IVR Hell for them, so that I can
> transfer the calls to, Hey, Its their dime.

A registered letter, with return receipt is the best way to deal with these
clowns.  If that doesn't work, your only alternative is to file a lawsuit.
There are some good online resources for this.  Filing a lawsuit is actually
fairly inexpensive, but when you compare that to the expense they incur
simply to have a lawyer show up in court, unless this guy owes thousands
of dollars (enough to make it worth their while in court), they will do
everything in their power to avoid getting into that scenario.

Collection efforts are all about making you feel badly enough that you cave
in to their demands.  Making them spend money to collect money is exactly
the opposite of what they want to do, and thus it's the most effective way to
get them to back down.

-- 
Tilghman

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Re: [asterisk-users] (no subject)

[Brian Capouch]

Alex Balashov wrote:

 ) How about rejecting emails that don't have a subject?

That is an excellent idea.

If a person doesn't have enough clue to use a subject, then we're really 
just feeding the beast when we indulge the question with an answer.

And the archived version of that question/answer are pretty useless, too.

Thx.

b.

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Re: [asterisk-users] Asterisk VXML... Help.

[Douglas Garstang]

Not for file:// access, No...


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

Sent: Thursday, July 3, 2008 2:21:42 PM
Subject: Re: [asterisk-users] Asterisk VXML... Help.

 
Does vxml let you use absolute paths?
 
Wouldn’t it have the equivalent of a
DocRoot???
 
Alex
 
 


 
From:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Thursday, July 03, 2008 5:03
PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk
VXML... Help.
 
So, I'm trying to get the
Asterisk vxml (from i6net) working.
Having no luck with it.

My dial plan has:

exten => _X.,1,Answer()
exten => _X.,n,Wait(1)
exten => _X.,n,Vxml(file:///tmp/menu.vxml)

The /tmp/menu.vxml file has:


 
  
   
   Hello world!
 


The tmp directory also has the tt-monkeys.gsm file:

[EMAIL PROTECTED] tmp]# ls -l tt-monkeys.gsm
-rw-r--r--  1 root root 26697 Jul  3 20:57 tt-monkeys.gsm

The openvxi daemon is running:

[EMAIL PROTECTED] tmp]# ps -ef | grep openvxi
root  2076 1  0 18:33
?00:00:00 /bin/sh
/usr/sbin/safe_openvxi
root  2114  2076  0 18:33
?00:00:00 openvxi -channels 100
-config /etc/openvxi/client.cfg
root  2606  2409  0 21:00
pts/200:00:00 grep openvxi
[EMAIL PROTECTED] tmp]# 

The /etc/asterisk/vxml.conf file contains:

; VoiceXML Configuration
;
[general]
wav_codec=gsm
videosilence=
audiosilence=

[license]
max=1
video=no
key=

And, finally here's my console output:

-- Executing Vxml("SIP/xxx.201.84.142-b7600c30",
"file:///tmp/menu.vxml") in new stack
VoiceBrowser interface file:///tmp/menu.vxml
 Initialiting
  == VXML_URL=(null)
  == VXML_ID=(null)
  == VXML_PARAM=(null)
  == url=file:///tmp/menu.vxml
  == session=1
  == id=0
  == param=0
  == Opening (url=file:///tmp/menu.vxml, id=(null), param=(null))
  == (dnid=1yyy3160157)
  == (name=1xxx8635808)
  == (num=1xxx8635808)
  == remote=1xxx8635808
  == local=1yyy3160157
-- >
open|session=1|module=2|url=file:///tmp/menu.vxml|remote=1xxx8635808|local=1yyy3160157
-- < open|session=1|result=ok
 Waiting
-- < close|session=1
 Exiting
  == VXML_RESULT=


I hear NOTHING. Asterisk drops though to the next command in the dial plan.
Shouldn't I hear the tt-monkeys.gsm sound file being played? I tried to keep
this as simple as I could. I thought it was interesting too that when I tried
this with a web server instead of a local file, if the URL was wrong, the
VXML() app still said it connected and got the data ok.


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Re: [asterisk-users] Asterisk VXML... Help.

[Alexander Lopez]

Does vxml let you use absolute paths?

 

Wouldn't it have the equivalent of a DocRoot???

 

Alex

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, July 03, 2008 5:03 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk VXML... Help.

 

So, I'm trying to get the Asterisk vxml (from i6net) working.
Having no luck with it.

My dial plan has:

exten => _X.,1,Answer()
exten => _X.,n,Wait(1)
exten => _X.,n,Vxml(file:///tmp/menu.vxml)

The /tmp/menu.vxml file has:


 
  
   
   Hello world!
 


The tmp directory also has the tt-monkeys.gsm file:

[EMAIL PROTECTED] tmp]# ls -l tt-monkeys.gsm
-rw-r--r--  1 root root 26697 Jul  3 20:57 tt-monkeys.gsm

The openvxi daemon is running:

[EMAIL PROTECTED] tmp]# ps -ef | grep openvxi
root  2076 1  0 18:33 ?00:00:00 /bin/sh
/usr/sbin/safe_openvxi
root  2114  2076  0 18:33 ?00:00:00 openvxi -channels 100
-config /etc/openvxi/client.cfg
root  2606  2409  0 21:00 pts/200:00:00 grep openvxi
[EMAIL PROTECTED] tmp]# 

The /etc/asterisk/vxml.conf file contains:

; VoiceXML Configuration
;
[general]
wav_codec=gsm
videosilence=
audiosilence=

[license]
max=1
video=no
key=

And, finally here's my console output:

-- Executing Vxml("SIP/xxx.201.84.142-b7600c30",
"file:///tmp/menu.vxml") in new stack
VoiceBrowser interface file:///tmp/menu.vxml
 Initialiting
  == VXML_URL=(null)
  == VXML_ID=(null)
  == VXML_PARAM=(null)
  == url=file:///tmp/menu.vxml
  == session=1
  == id=0
  == param=0
  == Opening (url=file:///tmp/menu.vxml, id=(null), param=(null))
  == (dnid=1yyy3160157)
  == (name=1xxx8635808)
  == (num=1xxx8635808)
  == remote=1xxx8635808
  == local=1yyy3160157
-- >
open|session=1|module=2|url=file:///tmp/menu.vxml|remote=1xxx8635808|loc
al=1yyy3160157
-- < open|session=1|result=ok
 Waiting
-- < close|session=1
 Exiting
  == VXML_RESULT=


I hear NOTHING. Asterisk drops though to the next command in the dial
plan. Shouldn't I hear the tt-monkeys.gsm sound file being played? I
tried to keep this as simple as I could. I thought it was interesting
too that when I tried this with a web server instead of a local file, if
the URL was wrong, the VXML() app still said it connected and got the
data ok.





 

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Re: [asterisk-users] Choppy audio

[bkruse]

Eric "ManxPower" Wieling wrote:
> Make the card stop sharing it's IRQ with your IDE controller.  Try 
> moving the card to another slot.
>
> Asterisk has to send an audio packet every 20ms for VoIP calls.  I 
> believe Zaptel expects no more than a few ms of latency.  If something 
> is causing a delay, like the IDE controller locking interrupts and doing 
> disk activity then you're not going to get your interrupts serviced fast 
> enough and you will have audio issues.
>
>   

Good suggestion. Can I also recommend to use the same types
of soundfiles as your current playing channel? (eg ulaw sounds
for phones using the ulaw codec)

-bk
> Doug Crompton wrote:
>   
>> I am not sure who all see's this list but I do have a few questions that
>> probably only the developers or somone really in the know of Asterisk
>> could answer.
>>
>> - What is the requirement for timing vs. audio playback in Asterisk.
>> Specifically voicemail and IVR's (Not meetme)
>>
>> - Has this requirment changed in newer versions?
>>
>> This obviously is when using Asterisk with no internal cards. I used
>> Asterisk for several years with a P3 Linux system, NO timing, and it
>> worked flawlessly. Now with this new Pentium Dual core system I do not
>> have the perfect audio I experienced with the less powerful system.
>>
>> I fully know there are MANY variable here. It could be a combination of
>> many things, including the OS (Linux Kernel) etc. BUT I offer this input,
>> Music on Hold works fine. This uses mpg123. So why can this palyback fine
>> and the other wav/gsm audio be choppy?
>>
>> I would gladly switch to a newer Asterisk (using 1.2.29) if someone said
>> this was solved in that version.
>>
>> My system can obviously play (mpg123 - background) audio fine. Why then
>> does Asterisk internal audio not also play well?
>>
>> Doug
>>
>> 
>> *  Doug Crompton*
>> *  Richboro, PA 18954   *
>> *  215-431-6307 *
>> *   *
>> * [EMAIL PROTECTED]*
>> * http://www.crompton.com  *
>> 
>>
>>
>>
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>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> 
>
>   


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Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?

[Matt Gibson]

Hi Roland, 

 

No problem, glad this works for you. We don't find it too bad. 

 

Hm, I'm not sure why you're having difficulty with the editing tool, you can
check on xda-developers.org forum for more information on the editing tool,
there may be a newer version. If you need help, feel free to email me at
[EMAIL PROTECTED] 

 

Thanks,

Matt G

 

: http://www.voipphreak.ca

: http://www.ratemydialplan.com

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd
Sent: Thursday, July 03, 2008 2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?

 

Hey Matt!!

thanks for the advice! 

appreciate it.. just installed it and everything worked fine ( i got
internet calling in my menu) though i cant seem to access the editing tool..

keeps on giving me some error even after soft reseting..
any idea?!





  _  

From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Thu, 3 Jul 2008 13:11:40 -0400
Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?

Hi Roland, 

 

Did you try:

 

http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/

 

We have this successfully working on a Touch (ELF), and a HTC Tilt (Tytn II)

 

Thanks,

Matt G

 

: http://www.voipphreak.ca

: http://www.ratemydialplan.com

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd
Sent: Thursday, July 03, 2008 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?

 

Hey! 
i'm facing the same prob.. 
i bought an HTC vox (s710) 2 weeks ago, and im still looking for a sip
client..! 
so far i found these 3:

AGEphone mobile: http://www.ageet.com/

SJphone: http://www.sjlabs.com/sjp.html

Bria Mobile: http://www.counterpath.com/enterprise-mobility-gateway.html


so far i just tried AgePhone (trial mode) sound is great though im facing 2
problems with it:
1: u can only use handsfree option, tht mean its a privacy killer.
2: you cant get a dial tone on it, tht means if you got 2 stage dialing on
your asterisk (if u call an extension, and wait to hear a dialtone) it wont
work.


as for the other 2 i didnt try them yet...

ps: if you found out anything else bout this matter id appreciate if you
could let me know :)

 

  _  

> Date: Mon, 30 Jun 2008 21:51:57 +0200
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Windows Mobile 6 IAX/SIP client?
> 
> I just bought a HTC TyTn II phone, but unfortunately it doesn't even have 
> a SIP client in it.
> 
> I tried the wiki searching for a SIP or IAX client but only found some 
> PocketPC stuff (Windows Mobile 2003).
> 
> Does anyone know of a good quality SIP or IAX softphone that will run on 
> Windows Mobile 6?
> 
> I only have a data subscription, no voice so the quality should be 
> sufficient to be used constantly.
> 
> Thanks!!
> 
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  _  

Discover the new Windows Vista Learn more!
 

 

  _  

Explore the seven wonders of the world Learn more!
 

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[asterisk-users] Asterisk VXML... Help.

[Douglas Garstang]

So, I'm trying to get the Asterisk vxml (from i6net) working.
Having no luck with it.

My dial plan has:

exten => _X.,1,Answer()
exten => _X.,n,Wait(1)
exten => _X.,n,Vxml(file:///tmp/menu.vxml)

The /tmp/menu.vxml file has:


 
  
   
   Hello world!
 


The tmp directory also has the tt-monkeys.gsm file:

[EMAIL PROTECTED] tmp]# ls -l tt-monkeys.gsm
-rw-r--r--  1 root root 26697 Jul  3 20:57 tt-monkeys.gsm

The openvxi daemon is running:

[EMAIL PROTECTED] tmp]# ps -ef | grep openvxi
root  2076 1  0 18:33 ?00:00:00 /bin/sh /usr/sbin/safe_openvxi
root  2114  2076  0 18:33 ?00:00:00 openvxi -channels 100 -config 
/etc/openvxi/client.cfg
root  2606  2409  0 21:00 pts/200:00:00 grep openvxi
[EMAIL PROTECTED] tmp]# 

The /etc/asterisk/vxml.conf file contains:

; VoiceXML Configuration
;
[general]
wav_codec=gsm
videosilence=
audiosilence=

[license]
max=1
video=no
key=

And, finally here's my console output:

-- Executing Vxml("SIP/xxx.201.84.142-b7600c30", "file:///tmp/menu.vxml") 
in new stack
VoiceBrowser interface file:///tmp/menu.vxml
 Initialiting
  == VXML_URL=(null)
  == VXML_ID=(null)
  == VXML_PARAM=(null)
  == url=file:///tmp/menu.vxml
  == session=1
  == id=0
  == param=0
  == Opening (url=file:///tmp/menu.vxml, id=(null), param=(null))
  == (dnid=1yyy3160157)
  == (name=1xxx8635808)
  == (num=1xxx8635808)
  == remote=1xxx8635808
  == local=1yyy3160157
-- > 
open|session=1|module=2|url=file:///tmp/menu.vxml|remote=1xxx8635808|local=1yyy3160157
-- < open|session=1|result=ok
 Waiting
-- < close|session=1
 Exiting
  == VXML_RESULT=


I hear NOTHING. Asterisk drops though to the next command in the dial plan. 
Shouldn't I hear the tt-monkeys.gsm sound file being played? I tried to keep 
this as simple as I could. I thought it was interesting too that when I tried 
this with a web server instead of a local file, if the URL was wrong, the 
VXML() app still said it connected and got the data ok.


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Re: [asterisk-users] Spoofing CID

[Alexander Lopez]

Neither DHS nor FTC has any legislation on this. Florida house had a
bill. Unfortunately, Collection agencies are deceptive by nature as most
other options have been exhausted before an account goes to collections.
I get the same thing here; they once even called me from a number that
had my same last name!!

 

There is one Alex Lopez (NOT ME) here in Miami that owes a lot of
people a lot of money. I get calls at all times of the day and night,
they forge the number, and so what do they care about following the FTC
rules

 

I feel your pain. I may create an IVR Hell for them, so that I can
transfer the calls to, Hey, Its their dime.

 

Alex

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Thursday, July 03, 2008 4:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Spoofing CID

 

Yeah I'm thinking either homeland security or some other
identity-critical legislation might be on my side here.

On Thu, Jul 3, 2008 at 12:40 PM, randulo <[EMAIL PROTECTED]>
wrote:

On Thu, Jul 3, 2008 at 9:26 PM, Robert Goodyear <[EMAIL PROTECTED]>
wrote:
> So who out there is aware of the FCC or FTC laws concerning spoofing
caller
> ID for deceptive purposes? There's a collection agency out there who
has my
> wife's name crossed with someone else's, and they are picking numbers
from
> our area code to present themselves as when calling us (over and over
and
> over.) I of course would like to turn this around on them as they
refuse to
> believe who we say we are.

That sucks!

Here's an older article about this seemingly common practice:

http://www.securityfocus.com/news/9822

 

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Re: [asterisk-users] Spoofing CID

[Brent Davidson]

Robert Goodyear wrote:
Yeah I'm thinking either homeland security or some other 
identity-critical legislation might be on my side here.


On Thu, Jul 3, 2008 at 12:40 PM, randulo <[EMAIL PROTECTED] 
> wrote:


On Thu, Jul 3, 2008 at 9:26 PM, Robert Goodyear
<[EMAIL PROTECTED] > wrote:
> So who out there is aware of the FCC or FTC laws concerning
spoofing caller
> ID for deceptive purposes? There's a collection agency out there
who has my
> wife's name crossed with someone else's, and they are picking
numbers from
> our area code to present themselves as when calling us (over and
over and
> over.) I of course would like to turn this around on them as
they refuse to
> believe who we say we are.
That sucks!

Here's an older article about this seemingly common practice:

http://www.securityfocus.com/news/9822



If I were you I would go to the local Sheriff's department (not local 
police) and file a harassment complaint.
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Re: [asterisk-users] (no subject)

[Steve Edwards]

On Fri, 4 Jul 2008, Peter Lindquist wrote:

>> Steve Edwards wrote:

>>> But deciphering posts from our non-English-speaking members is half the 
>>> challenge/fun :)
>>> 
>>> Seriously though, good for them for trying. I wouldn't.
>>> 
>>> What are you if you speak 3 languages? Trilingual.
>>> 
>>> What are you if you speak 2 languages? Bilingual.
>>> 
>>> What are you if you only speak 1 language? American :)
>> 
> Bilingual, Trilingual, -lingual does not necessarily include English as 
> one of the languages. It is for some a great effort just trying to write in 
> English, never mind the effort of knowing colloquialism, etc.  So not being 
> fluent, not being able to be as coherent as a native English speaker would, 
> does not make me or someone else eligible for an answer. No wonder so many 
> think that monolingual people with English as their only language are 
> arrogant
>
> Yes, diatribes and flames are accepted

Boy, did you miss the mark. I am a monolingual American. I was giving 
non-English-speakers props for trying and poking fun at myself and my 
countrymen. Lighten up.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Spoofing CID

[Robert Goodyear]

Yeah I'm thinking either homeland security or some other identity-critical
legislation might be on my side here.

On Thu, Jul 3, 2008 at 12:40 PM, randulo <[EMAIL PROTECTED]> wrote:

> On Thu, Jul 3, 2008 at 9:26 PM, Robert Goodyear <[EMAIL PROTECTED]>
> wrote:
> > So who out there is aware of the FCC or FTC laws concerning spoofing
> caller
> > ID for deceptive purposes? There's a collection agency out there who has
> my
> > wife's name crossed with someone else's, and they are picking numbers
> from
> > our area code to present themselves as when calling us (over and over and
> > over.) I of course would like to turn this around on them as they refuse
> to
> > believe who we say we are.
> That sucks!
>
> Here's an older article about this seemingly common practice:
>
> http://www.securityfocus.com/news/9822
>
>
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Re: [asterisk-users] Spoofing CID

[randulo]

On Thu, Jul 3, 2008 at 9:26 PM, Robert Goodyear <[EMAIL PROTECTED]> wrote:
> So who out there is aware of the FCC or FTC laws concerning spoofing caller
> ID for deceptive purposes? There's a collection agency out there who has my
> wife's name crossed with someone else's, and they are picking numbers from
> our area code to present themselves as when calling us (over and over and
> over.) I of course would like to turn this around on them as they refuse to
> believe who we say we are.
That sucks!

Here's an older article about this seemingly common practice:

http://www.securityfocus.com/news/9822

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[asterisk-users] Palm OS IAX client?

[Ricardo Cuevas]

any body know about a iax softphone for palm os ?



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[asterisk-users] Spoofing CID

[Robert Goodyear]

So who out there is aware of the FCC or FTC laws concerning spoofing caller
ID for deceptive purposes? There's a collection agency out there who has my
wife's name crossed with someone else's, and they are picking numbers from
our area code to present themselves as when calling us (over and over and
over.) I of course would like to turn this around on them as they refuse to
believe who we say we are.
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Re: [asterisk-users] (no subject)

[Peter Lindquist]

Alex Balashov wrote:

Steve Edwards wrote:
  

On Thu, 3 Jul 2008, Alex Balashov wrote:



Steve Edwards wrote:
  

On Thu, 3 Jul 2008, Alex Balashov wrote:



C F wrote:

  

The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P


Oh, if only more newbie posters on this list would heed that advice.
  

) How about rejecting emails that don't have a subject?

) How about rejecting top posted replies?

) How about rejecting posts to -dev until the poster's account is more
than a couple of days old? Or until they've earned a couple of karma
points? Or a challenge/response confirming "this post is about changing
the C source code?"


I would say the main thing that is needed is a grammar and spelling
checker, followed by some degree of nominal assessment of conceptual
integrity and coherence.  The latter may be impossible to implement, but
the former would be beneficial.
  
But deciphering posts from our non-English-speaking members is half the 
challenge/fun :)


Seriously though, good for them for trying. I wouldn't.

What are you if you speak 3 languages? Trilingual.

What are you if you speak 2 languages? Bilingual.

What are you if you only speak 1 language? American :)



I'm trilingual, but English is by far my best language.  If I had to 
write a post on a technical mailing list in one of the other languages, 
I would certainly take the time to ensure that it sounds reasonably 
coherent.


I cannot fault people for poor/limited English.  But there is a 
difference between someone who tried and someone who didn't, and it is 
reflected in the overall level of culture that comes across in the 
substance of their post, the formulation of their thoughts, and so on.


Somebody that *both* speaks/writes English poorly -- *and* uses 
incomprehensible, Philistine gibberish (excuse me, AOLer short-hand) -- 
deserves what they earn.  There seems to be a remarkable coincidence of 
these two proclivities as often as not.


-- Alex

  
Bilingual, Trilingual, -lingual does not necessarily include English 
as one of the languages. It is for some a great effort just trying to 
write in English, never mind the effort of knowing colloquialism, etc.  
So not being fluent, not being able to be as coherent as a native 
English speaker would, does not make me or someone else eligible for an 
answer. No wonder so many think that monolingual people with English as 
their only language are arrogant


Yes, diatribes and flames are accepted

//Peter
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Re: [asterisk-users] (no subject)

[Alex Balashov]

Steve Edwards wrote:
> On Thu, 3 Jul 2008, Alex Balashov wrote:
> 
>> Steve Edwards wrote:
>>> On Thu, 3 Jul 2008, Alex Balashov wrote:
>>>
 C F wrote:

> The number one skill for setting up asterisk is learn how to
> communicate since it's a communication application :P
 Oh, if only more newbie posters on this list would heed that advice.
>>> ) How about rejecting emails that don't have a subject?
>>>
>>> ) How about rejecting top posted replies?
>>>
>>> ) How about rejecting posts to -dev until the poster's account is more
>>> than a couple of days old? Or until they've earned a couple of karma
>>> points? Or a challenge/response confirming "this post is about changing
>>> the C source code?"
>> I would say the main thing that is needed is a grammar and spelling
>> checker, followed by some degree of nominal assessment of conceptual
>> integrity and coherence.  The latter may be impossible to implement, but
>> the former would be beneficial.
> 
> But deciphering posts from our non-English-speaking members is half the 
> challenge/fun :)
> 
> Seriously though, good for them for trying. I wouldn't.
> 
> What are you if you speak 3 languages? Trilingual.
> 
> What are you if you speak 2 languages? Bilingual.
> 
> What are you if you only speak 1 language? American :)

I'm trilingual, but English is by far my best language.  If I had to 
write a post on a technical mailing list in one of the other languages, 
I would certainly take the time to ensure that it sounds reasonably 
coherent.

I cannot fault people for poor/limited English.  But there is a 
difference between someone who tried and someone who didn't, and it is 
reflected in the overall level of culture that comes across in the 
substance of their post, the formulation of their thoughts, and so on.

Somebody that *both* speaks/writes English poorly -- *and* uses 
incomprehensible, Philistine gibberish (excuse me, AOLer short-hand) -- 
deserves what they earn.  There seems to be a remarkable coincidence of 
these two proclivities as often as not.

-- Alex

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] (no subject)

[Steve Edwards]

On Thu, 3 Jul 2008, Alex Balashov wrote:

> Steve Edwards wrote:
>> On Thu, 3 Jul 2008, Alex Balashov wrote:
>>
>>> C F wrote:
>>>
 The number one skill for setting up asterisk is learn how to
 communicate since it's a communication application :P
>>> Oh, if only more newbie posters on this list would heed that advice.
>>
>> ) How about rejecting emails that don't have a subject?
>>
>> ) How about rejecting top posted replies?
>>
>> ) How about rejecting posts to -dev until the poster's account is more
>> than a couple of days old? Or until they've earned a couple of karma
>> points? Or a challenge/response confirming "this post is about changing
>> the C source code?"
>
> I would say the main thing that is needed is a grammar and spelling
> checker, followed by some degree of nominal assessment of conceptual
> integrity and coherence.  The latter may be impossible to implement, but
> the former would be beneficial.

But deciphering posts from our non-English-speaking members is half the 
challenge/fun :)

Seriously though, good for them for trying. I wouldn't.

What are you if you speak 3 languages? Trilingual.

What are you if you speak 2 languages? Bilingual.

What are you if you only speak 1 language? American :)

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?

[RoLaNd RoLaNd]

Hey Matt!!

thanks for the advice! 

appreciate it.. just installed it and everything worked fine ( i got internet 
calling in my menu) though i cant seem to access the editing tool..

keeps on giving me some error even after soft reseting..
any idea?!



From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Thu, 3 Jul 2008 13:11:40 -0400
Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?



















Hi Roland, 

 

Did you try:

 

http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windows-mobile-6x-for-free-voip-calls-using-asterisk/

 

We have this successfully working on a Touch (ELF), and a
HTC Tilt (Tytn II)

 

Thanks,

Matt G



 

: http://www.voipphreak.ca

: http://www.ratemydialplan.com



 





From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd
RoLaNd

Sent: Thursday, July 03, 2008 12:23 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?





 

Hey!


i'm facing the same prob.. 

i bought an HTC vox (s710) 2 weeks ago, and im still looking for a sip
client..! 

so far i found these 3:



AGEphone mobile: http://www.ageet.com/



SJphone: http://www.sjlabs.com/sjp.html



Bria Mobile: http://www.counterpath.com/enterprise-mobility-gateway.html





so far i just tried AgePhone (trial mode) sound is great though im facing 2
problems with it:

1: u can only use handsfree option, tht mean its a privacy killer.

2: you cant get a dial tone on it, tht means if you got 2 stage dialing on your
asterisk (if u call an extension, and wait to hear a dialtone) it wont work.





as for the other 2 i didnt try them yet...



ps: if you found out anything else bout this matter id appreciate if you could
let me know :)













> Date: Mon, 30 Jun 2008 21:51:57 +0200

> From: [EMAIL PROTECTED]

> To: asterisk-users@lists.digium.com

> Subject: [asterisk-users] Windows Mobile 6 IAX/SIP client?

> 

> I just bought a HTC TyTn II phone, but unfortunately it doesn't even have 

> a SIP client in it.

> 

> I tried the wiki searching for a SIP or IAX client but only found some 

> PocketPC stuff (Windows Mobile 2003).

> 

> Does anyone know of a good quality SIP or IAX softphone that will run on 

> Windows Mobile 6?

> 

> I only have a data subscription, no voice so the quality should be 

> sufficient to be used constantly.

> 

> Thanks!!

> 

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[asterisk-users] tone differentiation

[Fidel Garcia]

 

I have two different scenario where I would like to apply different tones:

 

1.   Incoming calls should have a different ringing tone than transfer
calls

2.   While on a call, transfer calls should have a different beep sound
on the handset than incoming calls.

 

How/where can I accomplish this?

 

 

Fidel Garcia

System Engineer

 

sysTeam.

7205 NW 19th Street, Suite 302
Miami, Florida 33126

Email: [EMAIL PROTECTED] 

Tel: (305)-477-7303 Fax: (305)-477-0013 

http://www.systeamusa.com

 

 

Fidel Garcia

System Engineer

 

sysTeam.

7205 NW 19th Street, Suite 302
Miami, Florida 33126

Email: [EMAIL PROTECTED] 

Tel: (305)-477-7303 Fax: (305)-477-0013 

http://www.systeamusa.com

 

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Re: [asterisk-users] (no subject)

[Alex Balashov]

Steve Edwards wrote:
> On Thu, 3 Jul 2008, Alex Balashov wrote:
> 
>> C F wrote:
>>
>>> The number one skill for setting up asterisk is learn how to
>>> communicate since it's a communication application :P
>> Oh, if only more newbie posters on this list would heed that advice.
> 
> ) How about rejecting emails that don't have a subject?
> 
> ) How about rejecting top posted replies?
> 
> ) How about rejecting posts to -dev until the poster's account is more 
> than a couple of days old? Or until they've earned a couple of karma 
> points? Or a challenge/response confirming "this post is about changing 
> the C source code?"

I would say the main thing that is needed is a grammar and spelling 
checker, followed by some degree of nominal assessment of conceptual 
integrity and coherence.  The latter may be impossible to implement, but 
the former would be beneficial.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] (no subject)

[Steve Edwards]

On Thu, 3 Jul 2008, Alex Balashov wrote:

> C F wrote:
>
>> The number one skill for setting up asterisk is learn how to
>> communicate since it's a communication application :P
>
> Oh, if only more newbie posters on this list would heed that advice.

) How about rejecting emails that don't have a subject?

) How about rejecting top posted replies?

) How about rejecting posts to -dev until the poster's account is more 
than a couple of days old? Or until they've earned a couple of karma 
points? Or a challenge/response confirming "this post is about changing 
the C source code?"

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] (no subject)

[Alex Balashov]

C F wrote:

> The number one skill for setting up asterisk is learn how to
> communicate since it's a communication application :P

Oh, if only more newbie posters on this list would heed that advice.

do u rely think this iz an acceptbl manner o/discoorse?

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?

[Matt Gibson]

Hi Roland, 

 

Did you try:

 

http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/

 

We have this successfully working on a Touch (ELF), and a HTC Tilt (Tytn II)

 

Thanks,

Matt G

 

: http://www.voipphreak.ca

: http://www.ratemydialplan.com

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd
Sent: Thursday, July 03, 2008 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?

 

Hey! 
i'm facing the same prob.. 
i bought an HTC vox (s710) 2 weeks ago, and im still looking for a sip
client..! 
so far i found these 3:

AGEphone mobile: http://www.ageet.com/

SJphone: http://www.sjlabs.com/sjp.html

Bria Mobile: http://www.counterpath.com/enterprise-mobility-gateway.html


so far i just tried AgePhone (trial mode) sound is great though im facing 2
problems with it:
1: u can only use handsfree option, tht mean its a privacy killer.
2: you cant get a dial tone on it, tht means if you got 2 stage dialing on
your asterisk (if u call an extension, and wait to hear a dialtone) it wont
work.


as for the other 2 i didnt try them yet...

ps: if you found out anything else bout this matter id appreciate if you
could let me know :)





  _  

> Date: Mon, 30 Jun 2008 21:51:57 +0200
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Windows Mobile 6 IAX/SIP client?
> 
> I just bought a HTC TyTn II phone, but unfortunately it doesn't even have 
> a SIP client in it.
> 
> I tried the wiki searching for a SIP or IAX client but only found some 
> PocketPC stuff (Windows Mobile 2003).
> 
> Does anyone know of a good quality SIP or IAX softphone that will run on 
> Windows Mobile 6?
> 
> I only have a data subscription, no voice so the quality should be 
> sufficient to be used constantly.
> 
> Thanks!!
> 
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Re: [asterisk-users] (no subject)

[C F]

The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P

As for your problem looks like you are trying to use the wrong span
for dial out.


On Thu, Jul 3, 2008 at 8:50 AM, Bikrish Amatya <[EMAIL PROTECTED]> wrote:
>
>
> Hello everybody
>
>
> I have configures asterisk server
> and i
> am using TE220P digium card.  Here is the content of
> the
> /etc/zaptel.conf file
> ###
> span=1,1,0,ccs,hdb3
> bchan=1-15,17-31
> dchan=16
>
> span=2,2,0,ccs,hdb3
> bchan=32-46,48-62
> dchan=47
>
>
> loadzone= in
> defaultzone = in
>
> 
>
> the content of
> /etc/asterisk/zapata.conf is as follow
>
> 
> [channels]
> context=incoming
> switchtype=national
> ;pridialplan=national
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> echocancel=yes
> rxgain=0.0
> txgain=0.0
> immediate=no
> callprogress=no
> callerid=asreceived
> group=1
> channel=>1-15,17-31
> #
>
> output of zttool is as follow
>
>
>
>
> │
> Alarms
> Span
> │
>
> │
> RED
> T2XXP (PCI) Card 0 Span
> 1
>
>
> │
> OK
> T2XXP (PCI) Card 0 Span
> 2
>
>
> │
>
>
>
> Output of  cat /prox/zaptel/1 is as follow
>
>
> Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span
> 1"
> HDB3/CCS RED
>
>1
> TE2/0/1/1
> Clear (In use) RED
>2
> TE2/0/1/2
> Clear (In use) RED
>3
> TE2/0/1/3
> Clear (In use) RED
>4
> TE2/0/1/4
> Clear (In use) RED
>5
> TE2/0/1/5
> Clear (In use) RED
>6
> TE2/0/1/6
> Clear (In use) RED
>7
> TE2/0/1/7
> Clear (In use) RED
>8
> TE2/0/1/8
> Clear (In use) RED
>9
> TE2/0/1/9
> Clear (In use) RED
>   10 TE2/0/1/10
> Clear (In use) RED
>   11 TE2/0/1/11
> Clear (In use) RED
>   12 TE2/0/1/12
> Clear (In use) RED
>   13 TE2/0/1/13
> Clear (In use) RED
>   14 TE2/0/1/14
> Clear (In use) RED
>   15 TE2/0/1/15
> Clear (In use) RED
>   16 TE2/0/1/16
> HDLCFCS (In use) RED
>   17 TE2/0/1/17
> Clear (In use) RED
>   18 TE2/0/1/18
> Clear (In use) RED
>   19 TE2/0/1/19
> Clear (In use) RED
>   20 TE2/0/1/20
> Clear (In use) RED
>   21 TE2/0/1/21
> Clear (In use) RED
>   22 TE2/0/1/22
> Clear (In use) RED
>   23 TE2/0/1/23
> Clear (In use) RED
>   24 TE2/0/1/24
> Clear (In use) RED
>   25 TE2/0/1/25
> Clear (In use) RED
>   26 TE2/0/1/26
> Clear (In use) RED
>   27 TE2/0/1/27
> Clear (In use) RED
>   28 TE2/0/1/28
> Clear (In use) RED
>   29 TE2/0/1/29
> Clear (In use) RED
>   30 TE2/0/1/30
> Clear (In use) RED
>   31 TE2/0/1/31
> Clear (In use) RED
>
> I
> am
> new to asterisk and googled around , configured the asterisk
> server. Now
> when i make a call from outside , it give me busy
> tone..  and when i
> call from softphone .. it shows me as show
> below
>
>
>-- Executing
> [EMAIL PROTECTED]:1]
> Dial("SIP/bikrish-09b21980",
> "Zap/g1/600833") in
> new stack
> [Jul  3
> 19:14:34] WARNING[6018]: app_dial.c:1183
> dial_exec_full: Unable to
> create channel of type 'Zap' (cause 34 -
> Circuit/channel
> congestion)
>   == Everyone is busy/congested at
> this time
> (1:0/1/0)
>   == Auto fallthrough, channel
> 'SIP/bikrish-09b21980' status is 'CONGESTION'
>
> I am not able
> to
> figure out the problem. Any kind of help would be appericiated.
>
> Thanking you
>
> bikrish
>
>
>
>
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Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?

[RoLaNd RoLaNd]

Hey! 
i'm facing the same prob.. 
i bought an HTC vox (s710) 2 weeks ago, and im still looking for a sip 
client..! 
so far i found these 3:

AGEphone mobile: http://www.ageet.com/

SJphone: http://www.sjlabs.com/sjp.html

Bria Mobile: http://www.counterpath.com/enterprise-mobility-gateway.html


so far i just tried AgePhone (trial mode) sound is great though im facing 2 
problems with it:
1: u can only use handsfree option, tht mean its a privacy killer.
2: you cant get a dial tone on it, tht means if you got 2 stage dialing on your 
asterisk (if u call an extension, and wait to hear a dialtone) it wont work.


as for the other 2 i didnt try them yet...

ps: if you found out anything else bout this matter id appreciate if you could 
let me know :)



> Date: Mon, 30 Jun 2008 21:51:57 +0200
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Windows Mobile 6 IAX/SIP client?
> 
> I just bought a HTC TyTn II phone, but unfortunately it doesn't even have 
> a SIP client in it.
> 
> I tried the wiki searching for a SIP or IAX client but only found some 
> PocketPC stuff (Windows Mobile 2003).
> 
> Does anyone know of a good quality SIP or IAX softphone that will run on 
> Windows Mobile 6?
> 
> I only have a data subscription, no voice so the quality should be 
> sufficient to be used constantly.
> 
> Thanks!!
> 
> ___
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> 
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
> 
> asterisk-users mailing list
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>http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Can't call my Extensions HELP!

[Raúl Gómez C.]

Tariq,

I cannot see the "context=" line in your sip.conf setup. Do you have the
appropriate context defined in your sip.conf that match your users context
in extension.conf???

On Tue, Jul 1, 2008 at 7:09 PM, Tariq .. <[EMAIL PROTECTED]> wrote:

> Greetings..
> i have 20 extensions with two queues.. i have members in the queues as
> SIP/
> now recently i have noticed that users are unable to call each other.. this
> is causing me a headache..
> calls comming to the queues are forwarded smoothly to the users.. but they
> can't call eachother.. what is going on??
> i'm using Asterisk 1.4.19-1 with FreePBX 2.4.0.1
> my SIP.CONF settings are
> 
> [3000]
> type=friend
> secret=3000
> record_out=Adhoc
> record_in=Adhoc
> qualify=yes
> port=5060
> pickupgroup=
> nat=yes
> [EMAIL PROTECTED]
> host=dynamic
> dtmfmode=rfc2833
> dial=SIP/3000
> context=from-internal
> canreinvite=no
> callgroup=
> callerid=device <3000>
> accountcode=
> call-limit=1
> busy-limit=1
> 
>
>
> my Extensions.conf are like this
>
> -
>
> [ext-local]
> include => ext-local-custom
> exten => 3000,1,Macro(exten-vm,novm,3000)
> exten => 3000,n,Hangup
> exten => 3000,hint,SIP/3000
>
> [from-did-direct-ivr]
> include => from-did-direct-ivr-custom
> exten => 3000,1,ExecIf($["${BLKVM_OVERRIDE}" !=
> ""],dbDel,${BLKVM_OVERRIDE})
> exten => 3000,n,Set(__NODEST=)
> exten => 3000,n,Goto(from-did-direct,3000,1)
>
> -
>
> my queues.conf
> --
> [8000]
> announce-frequency=0
> announce-holdtime=no
> eventmemberstatus=no
> eventwhencalled=no
> joinempty=yes
> leavewhenempty=no
> maxlen=0
> periodic-announce-frequency=0
> queue-callswaiting=silence/1
> queue-thereare=silence/1
> queue-youarenext=silence/1
> retry=1
> strategy=random
> timeout=5
> wrapuptime=0
> member=SIP/3000,0
>
>
>
> please help! i know i was able to call from an SIP to another SIP .. now i
> can't!
>
> --
> The other season of giving begins 6/24/08. Check out the i'm Talkathon. Check
> it out! 
>
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-- 
Raul
Linux Counter #156439
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Re: [asterisk-users] wait & pickup

[Enrico Pasqualotto]

On Thu, 2008-07-03 at 09:31 -0500, Eric "ManxPower" Wieling wrote:
> chan_iax2 does not support pickup (callpickup=, pickupgroup= and *8).

Wow! It's a very nice problem
And for redirect a call in wait state to a sip phone? Without pickup ...
Channelredirect don't work with ringing channel for me.

Thanks Pasqu.




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[asterisk-users] how to setup one stage dialing plan, instead of two! help!!!

[RoLaNd RoLaNd]

Hello all,

i recently finished setting up my asterisk with sipura 3102 using PSTN.

this is my dial plan relevant to wht i want:

exten =>_01,1,Dial(SIP/$(EXTEN)@200)

right now as u see i made my dial plan on a 2 stage dialing mode.
tht means i dial 01, i get the pstn dial tone, and then i call whichever number 
i want through it.
i want to have the option for my call to directly go through pstn without 
having to wait for the pstn dial tone.
 any help would be appreciated.. :)

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[asterisk-users] D-Link DVG-3104MS

[Bill Michaelson]

This appears to be a SIP gateway to four FXO ports for ~$250. Has 
anybody used it with Asterisk? Comments?


http://www.ipphoneshack.com/products/D_Link_DVG_3104MS_VoiceCenter_4_Port_PSTN_Gateway-193-12.html

Any good reason to pay for a Mediatrix 1204 or some other box instead?



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Re: [asterisk-users] wait & pickup

[Eric "ManxPower" Wieling]

chan_iax2 does not support pickup (callpickup=, pickupgroup= and *8).

Enrico Pasqualotto wrote:
> Hi all, One question 
> I have set in the extensions.conf of my asterisk that all incoming call
> go in the wait application because I need to not "connect" the caller
> but remain in the ringing state.
> After that the call is on the wait exten for a N second I need from
> other sip phone to pickup this call.
> There is a way to pickup a call arrived from IAX to an exten wait(999)?
> 
> I see that the problem is the channel state, my channel in wait is in
> "LINE IS RING" but the pickup appl search for channel in "REMOTE
> RINGING".

-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

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[asterisk-users] wait & pickup

[Enrico Pasqualotto]

Hi all, One question 
I have set in the extensions.conf of my asterisk that all incoming call
go in the wait application because I need to not "connect" the caller
but remain in the ringing state.
After that the call is on the wait exten for a N second I need from
other sip phone to pickup this call.
There is a way to pickup a call arrived from IAX to an exten wait(999)?

I see that the problem is the channel state, my channel in wait is in
"LINE IS RING" but the pickup appl search for channel in "REMOTE
RINGING".

Anyone have solution for this problem?

TIA Pasqu...


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Re: [asterisk-users] problem in making call pc to phone & vice versa

[Lyle Giese]

Your E1 links are down. (red alarm)  Your card does not like or see your
providers E1.

Lyle

Bikrish Amatya wrote:
> Hello everybody
>
>
> I have configures asterisk server
> and i
> am using TE220P digium card.  Here is the content of
> the
> /etc/zaptel.conf file 
> ###
> span=1,1,0,ccs,hdb3
> bchan=1-15,17-31
> dchan=16
>
> span=2,2,0,ccs,hdb3
> bchan=32-46,48-62
> dchan=47
>
>
> loadzone= in
> defaultzone = in
>
> 
>
> the content of
> /etc/asterisk/zapata.conf is as follow
>
> 
> [channels]
> context=incoming
> switchtype=national
> ;pridialplan=national
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> echocancel=yes
> rxgain=0.0
> txgain=0.0
> immediate=no
> callprogress=no
> callerid=asreceived
> group=1
> channel=>1-15,17-31
> #
>
> output of zttool is as follow
>
> 
>
>
> │
> Alarms 
> Span  
> │
>
> │
> RED
> T2XXP (PCI) Card 0 Span
> 1 
>
>
> │
> OK 
> T2XXP (PCI) Card 0 Span
> 2  
>
>
> │ 
>
>
>
> Output of  cat /prox/zaptel/1 is as follow
>
>
> Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span
> 1"
> HDB3/CCS RED
>
>1
> TE2/0/1/1
> Clear (In use) RED
>2
> TE2/0/1/2
> Clear (In use) RED
>3
> TE2/0/1/3
> Clear (In use) RED
>4
> TE2/0/1/4
> Clear (In use) RED
>5
> TE2/0/1/5
> Clear (In use) RED
>6
> TE2/0/1/6
> Clear (In use) RED
>7
> TE2/0/1/7
> Clear (In use) RED
>8
> TE2/0/1/8
> Clear (In use) RED
>9
> TE2/0/1/9
> Clear (In use) RED
>   10 TE2/0/1/10
> Clear (In use) RED
>   11 TE2/0/1/11
> Clear (In use) RED
>   12 TE2/0/1/12
> Clear (In use) RED
>   13 TE2/0/1/13
> Clear (In use) RED
>   14 TE2/0/1/14
> Clear (In use) RED
>   15 TE2/0/1/15
> Clear (In use) RED
>   16 TE2/0/1/16
> HDLCFCS (In use) RED
>   17 TE2/0/1/17
> Clear (In use) RED
>   18 TE2/0/1/18
> Clear (In use) RED
>   19 TE2/0/1/19
> Clear (In use) RED
>   20 TE2/0/1/20
> Clear (In use) RED
>   21 TE2/0/1/21
> Clear (In use) RED
>   22 TE2/0/1/22
> Clear (In use) RED
>   23 TE2/0/1/23
> Clear (In use) RED
>   24 TE2/0/1/24
> Clear (In use) RED
>   25 TE2/0/1/25
> Clear (In use) RED
>   26 TE2/0/1/26
> Clear (In use) RED
>   27 TE2/0/1/27
> Clear (In use) RED
>   28 TE2/0/1/28
> Clear (In use) RED
>   29 TE2/0/1/29
> Clear (In use) RED
>   30 TE2/0/1/30
> Clear (In use) RED
>   31 TE2/0/1/31
> Clear (In use) RED
>
> I
> am
> new to asterisk and googled around , configured the asterisk
> server. Now
> when i make a call from outside , it give me busy
> tone..  and when i
> call from softphone .. it shows me as show
> below
>
>
>-- Executing
> [EMAIL PROTECTED]:1]
> Dial("SIP/bikrish-09b21980",
> "Zap/g1/600833") in
> new stack
> [Jul  3
> 19:14:34] WARNING[6018]: app_dial.c:1183
> dial_exec_full: Unable to
> create channel of type 'Zap' (cause 34 -
> Circuit/channel
> congestion)
>   == Everyone is busy/congested at
> this time
> (1:0/1/0)
>   == Auto fallthrough, channel
> 'SIP/bikrish-09b21980' status is 'CONGESTION'
>
> I am not able
> to
> figure out the problem. Any kind of help would be appericiated.
>
> Thanking you
>
> bikrish
>
>
>
>
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Re: [asterisk-users] problem in making call pc to phone & vice versa

[Tzafrir Cohen]

Hi

On Thu, Jul 03, 2008 at 06:21:27PM +0530, Bikrish Amatya wrote:
> 
> 
> Hello everybody
> 
> 
> I have configures asterisk server
> and i
> am using TE220P digium card.  Here is the content of
> the
> /etc/zaptel.conf file 
> ###
> span=1,1,0,ccs,hdb3
> bchan=1-15,17-31
> dchan=16
> 
> span=2,2,0,ccs,hdb3
> bchan=32-46,48-62
> dchan=47

You have two ports. Which of those is connected?

> 
> 
> loadzone    = in
> defaultzone = in
> 
> 
> 
> the content of
> /etc/asterisk/zapata.conf is as follow
> 
> 
> [channels]
> context=incoming
> switchtype=national
> ;pridialplan=national
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> echocancel=yes
> rxgain=0.0
> txgain=0.0
> immediate=no
> callprogress=no
> callerid=asreceived
> group=1
> channel=>1-15,17-31

Only the channels of ports 1 are configured in Asterisk?

> Output of  cat /prox/zaptel/1 is as follow
> 
> 
>     Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" HDB3/CCS RED
> 
>    1 TE2/0/1/1  Clear (In use) RED
>    2 TE2/0/1/2  Clear (In use) RED

[snip]

It is actually in use by Asterisk. It is also in RED alarm. That is: no
layer 1 connection to the remote side. One possible reason for thaat is
that there's nothing connected to that port.

No point trying to call through this port.

What do you have in /proc/zaptel/2 ?

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] problem in making call pc to phone & vice versa

[Bikrish Amatya]

Hello everybody


I have configures asterisk server
and i
am using TE220P digium card.  Here is the content of
the
/etc/zaptel.conf file 
###
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16

span=2,2,0,ccs,hdb3
bchan=32-46,48-62
dchan=47


loadzone    = in
defaultzone = in



the content of
/etc/asterisk/zapata.conf is as follow


[channels]
context=incoming
switchtype=national
;pridialplan=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
rxgain=0.0
txgain=0.0
immediate=no
callprogress=no
callerid=asreceived
group=1
channel=>1-15,17-31
#

output of zttool is as follow



   
│
Alarms 
Span  
│
   
│
RED
T2XXP (PCI) Card 0 Span
1 

   
│
OK 
T2XXP (PCI) Card 0 Span
2  

   
│ 
   


Output of  cat /prox/zaptel/1 is as follow


    Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span
1"
HDB3/CCS RED

   1
TE2/0/1/1
Clear (In use) RED
   2
TE2/0/1/2
Clear (In use) RED
   3
TE2/0/1/3
Clear (In use) RED
   4
TE2/0/1/4
Clear (In use) RED
   5
TE2/0/1/5
Clear (In use) RED
   6
TE2/0/1/6
Clear (In use) RED
   7
TE2/0/1/7
Clear (In use) RED
   8
TE2/0/1/8
Clear (In use) RED
   9
TE2/0/1/9
Clear (In use) RED
  10 TE2/0/1/10
Clear (In use) RED
  11 TE2/0/1/11
Clear (In use) RED
  12 TE2/0/1/12
Clear (In use) RED
  13 TE2/0/1/13
Clear (In use) RED
  14 TE2/0/1/14
Clear (In use) RED
  15 TE2/0/1/15
Clear (In use) RED
  16 TE2/0/1/16
HDLCFCS (In use) RED
  17 TE2/0/1/17
Clear (In use) RED
  18 TE2/0/1/18
Clear (In use) RED
  19 TE2/0/1/19
Clear (In use) RED
  20 TE2/0/1/20
Clear (In use) RED
  21 TE2/0/1/21
Clear (In use) RED
  22 TE2/0/1/22
Clear (In use) RED
  23 TE2/0/1/23
Clear (In use) RED
  24 TE2/0/1/24
Clear (In use) RED
  25 TE2/0/1/25
Clear (In use) RED
  26 TE2/0/1/26
Clear (In use) RED
  27 TE2/0/1/27
Clear (In use) RED
  28 TE2/0/1/28
Clear (In use) RED
  29 TE2/0/1/29
Clear (In use) RED
  30 TE2/0/1/30
Clear (In use) RED
  31 TE2/0/1/31
Clear (In use) RED
   
I
am
new to asterisk and googled around , configured the asterisk
server. Now
when i make a call from outside , it give me busy
tone..  and when i
call from softphone .. it shows me as show
below


       -- Executing
[EMAIL PROTECTED]:1]
Dial("SIP/bikrish-09b21980",
"Zap/g1/600833") in
new stack
[Jul  3
19:14:34] WARNING[6018]: app_dial.c:1183
dial_exec_full: Unable to
create channel of type 'Zap' (cause 34 -
Circuit/channel
congestion)
  == Everyone is busy/congested at
this time
(1:0/1/0)
  == Auto fallthrough, channel
'SIP/bikrish-09b21980' status is 'CONGESTION'

I am not able
to
figure out the problem. Any kind of help would be appericiated.

Thanking you

bikrish




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[asterisk-users] (no subject)

[Bikrish Amatya]

Hello everybody


I have configures asterisk server
and i
am using TE220P digium card.  Here is the content of
the
/etc/zaptel.conf file 
###
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16

span=2,2,0,ccs,hdb3
bchan=32-46,48-62
dchan=47


loadzone    = in
defaultzone = in



the content of
/etc/asterisk/zapata.conf is as follow


[channels]
context=incoming
switchtype=national
;pridialplan=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
rxgain=0.0
txgain=0.0
immediate=no
callprogress=no
callerid=asreceived
group=1
channel=>1-15,17-31
#

output of zttool is as follow



   
│
Alarms 
Span  
│
   
│
RED
T2XXP (PCI) Card 0 Span
1 

   
│
OK 
T2XXP (PCI) Card 0 Span
2  

   
│ 
   


Output of  cat /prox/zaptel/1 is as follow


    Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span
1"
HDB3/CCS RED

   1
TE2/0/1/1
Clear (In use) RED
   2
TE2/0/1/2
Clear (In use) RED
   3
TE2/0/1/3
Clear (In use) RED
   4
TE2/0/1/4
Clear (In use) RED
   5
TE2/0/1/5
Clear (In use) RED
   6
TE2/0/1/6
Clear (In use) RED
   7
TE2/0/1/7
Clear (In use) RED
   8
TE2/0/1/8
Clear (In use) RED
   9
TE2/0/1/9
Clear (In use) RED
  10 TE2/0/1/10
Clear (In use) RED
  11 TE2/0/1/11
Clear (In use) RED
  12 TE2/0/1/12
Clear (In use) RED
  13 TE2/0/1/13
Clear (In use) RED
  14 TE2/0/1/14
Clear (In use) RED
  15 TE2/0/1/15
Clear (In use) RED
  16 TE2/0/1/16
HDLCFCS (In use) RED
  17 TE2/0/1/17
Clear (In use) RED
  18 TE2/0/1/18
Clear (In use) RED
  19 TE2/0/1/19
Clear (In use) RED
  20 TE2/0/1/20
Clear (In use) RED
  21 TE2/0/1/21
Clear (In use) RED
  22 TE2/0/1/22
Clear (In use) RED
  23 TE2/0/1/23
Clear (In use) RED
  24 TE2/0/1/24
Clear (In use) RED
  25 TE2/0/1/25
Clear (In use) RED
  26 TE2/0/1/26
Clear (In use) RED
  27 TE2/0/1/27
Clear (In use) RED
  28 TE2/0/1/28
Clear (In use) RED
  29 TE2/0/1/29
Clear (In use) RED
  30 TE2/0/1/30
Clear (In use) RED
  31 TE2/0/1/31
Clear (In use) RED
   
I
am
new to asterisk and googled around , configured the asterisk
server. Now
when i make a call from outside , it give me busy
tone..  and when i
call from softphone .. it shows me as show
below


       -- Executing
[EMAIL PROTECTED]:1]
Dial("SIP/bikrish-09b21980",
"Zap/g1/600833") in
new stack
[Jul  3
19:14:34] WARNING[6018]: app_dial.c:1183
dial_exec_full: Unable to
create channel of type 'Zap' (cause 34 -
Circuit/channel
congestion)
  == Everyone is busy/congested at
this time
(1:0/1/0)
  == Auto fallthrough, channel
'SIP/bikrish-09b21980' status is 'CONGESTION'

I am not able
to
figure out the problem. Any kind of help would be appericiated.

Thanking you

bikrish




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Re: [asterisk-users] (no subject)

[Neha Punia]

But if I m using this SendDTMF it does not send anything





I m using it like this in extension.conf

exten => 205,1,Answer



exten => 205,n,Wait(20)



exten => 205,n,Playback(dtmf-1)



exten => 205,n,Wait(20)



exten => 205,n,SendDTMF(9)



exten => 205,n,Wait(5)



exten => 205,n,Read(digito)



exten => 205,n,SayDigits(${digito})



exten => 205,n,Hangup



on the console it only shows tht the call completed and no message about the 
DTMF and in the log files it shows like :



Jul  3 17:21:01 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL 
PROTECTED]' of Request 102: Match Found

Jul  3 17:21:27 DEBUG[896] chan_sip.c: Setting NAT on RTP to 0

Jul  3 17:21:27 DEBUG[896] chan_sip.c: Outgoing Call for 205

Jul  3 17:21:27 DEBUG[896] chan_sip.c: (Provisional) Stopping retransmission 
(but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found

Jul  3 17:21:27 DEBUG[896] chan_sip.c: Acked pending invite 102

Jul  3 17:21:27 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL 
PROTECTED]' of Request 102: Match Found

Jul  3 17:21:27 DEBUG[896] chan_sip.c: build_route: Contact hop: 

Jul  3 17:21:47 DEBUG[896] chan_sip.c: * Detected inband DTMF '1'

Jul  3 17:22:18 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL 
PROTECTED]' of Request 102: Match Found

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '205'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is 'default'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is 'SIP/3001-008d8ce0'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is 'Hangup'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:21:27'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:21:27'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:22:23'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '56'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '56'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is 'ANSWERED'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is 'DOCUMENTATION'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '1215085887.0'

Jul  3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul  3 17:22:23 DEBUG[896] chan_sip.c: update_call_counter(205) - decrement 
call limit counter

Jul  3 17:22:23 NOTICE[896] pbx_spool.c: Call completed to SIP/3001

Jul  3 17:22:23 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL 
PROTECTED]' of Request 103: Match Found

Jul  3 17:22:24 DEBUG[896] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]'

Jul  3 17:23:57 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL 
PROTECTED]' of Request 102: Match Found

Jul  3 17:24:09 DEBUG[896] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]'

Jul  3 17:25:47 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL 
PROTECTED]' of Request 102: Match Found

Jul  3 17:25:54 DEBUG[896] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]'



It says "detected inband dtmf 1 but says nothing about 9.

Am I doing anything wrong in the extension.conf.





-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob
Sent: Thursday, July 03, 2008 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)





Use SendDTMF.







--- On Thu, 7/3/08, Neha Punia <[EMAIL PROTECTED]> wrote:



> From: Neha Punia <[EMAIL PROTECTED]>

> Subject: [asterisk-users] (no subject)

> To: "asterisk-users@lists.digium.com" 

> Date: Thursday, July 3, 2008, 10:29 AM

> Hi

> I  m making a call from one asterisk server to an asterisk

> client

> The call gets completed but I want it to send dtmf signals

>

> The dialplan I have made for this is like

> exten => 205,1,Answer

> exten => 205,n,Wait(15)

> exten => 205,n,Playback(dtmf-1)

> exten => 205,n,Wait(20)

>

> but it does not send any dtmf signal

> where is the problem??

>

>  CAUTION - Disclaimer *

> This e-mail contains PRIVILEGED AND CONFIDENTIAL

> INFORMATION intended solely

> for the use of the addressee(s). If you are not the

> intended recipient, please

> notify the sender by e-mail and delete the original

> message. Further, you are not

> to copy, disclose, or distribute this e-mail or its

> contents to any other person and

> any such actions are unlawful. This e-mail may contain

> viruses. Infosys has taken

> every reasonable precaution to minimize this risk, but is

> not liable for any damage

> you may sustain as a result of any virus in this e-mail.

> You should carry out your

> own virus checks before opening the e-mail or attachment.

> Infosys reserves the

Re: [asterisk-users] new install of asterisk appliance.

[Dean Collins]

My suggestion is you implement sbs2003 the correct way with a 2 nic
solution.

Yes a 1 nic installation is possible but you lose all the benefits.

Tell whoever set it up to tear it down and implement it properly.


Cheers,

Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis
Sent: Thursday, 3 July 2008 2:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] new install of asterisk appliance.

The Asterisk appliance can currently only auto-provision Polycom phones,

so you're going to need to manually configure the Grandstream phones.

Sydney Web Hosting wrote:
>
> I have 1 nic card which is linked to the router.
>
> Then I use 1 port on the router which is linked to the asterisk
appliance.
>
> but as Rob hillis said.
>
> It will work via WAN which ive now got. SO I can access the asterisk 
> appliance via 192.168.1.15
>
> The problem is now...How do I connect the phone.
>
> Ive got the phone (Ethernet) connected from the LAN port on the phone 
> to a LAN port in the asterisk appliance.
>
> im using the grandstream 2020.
>
>  
>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Dean 
> Collins
> *Sent:* Thursday, 3 July 2008 12:03 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] new install of asterisk appliance.
>
>  
>
> Ok so as you describe it you are only using 1 nic card in your sbs2003

> server?
>
>  
>
> But you have two devices connected to it? Care to explain?
>
>  
>
>
> Cheers,
>
> Dean
>
>

>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Sydney

> Web Hosting
> *Sent:* Wednesday, 2 July 2008 9:14 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* Re: [asterisk-users] new install of asterisk appliance.
>
>  
>
> I agree. ;-)
>
> here is our setup.
>
>  
>
> Router
>
> |
>
> |
>
> SBS server 2003 Nic 1
>
> |
>
> 4port gb switch
>
> |   |  
>
> |   |
>
> Pc'sAsterisk server
>
>  
>
>  
>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Dean 
> Collins
> *Sent:* Thursday, 3 July 2008 10:22 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] new install of asterisk appliance.
>
>  
>
> Davey, not to second guess you but sounds like you don't have a clue.
>
>  
>
> This is the way I have my network setup.
>
>  
>
>  
>
>  
>
> Router
>
> |
>
> |
>
> SBS server 2003 Nic 1
>
> SBS server 2003 Nic 2
>
> |
>
> |
>
> 16port gb switch
>
> |   |  
>
> |   |
>
> Pc'sAsterisk server
>
>  
>
>  
>
>
> Cheers,
>
> Dean
>
>

>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Sydney

> Web Hosting
> *Sent:* Wednesday, 2 July 2008 8:13 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* Re: [asterisk-users] new install of asterisk appliance.
>
>  
>
> Netcomm is directly connected to the server via server network card.
>
> ive now got it up and running. I changed the WAN info in asterisk 
> appliance, not the LAN info.
>
> Now the phone wont get connected. Ive added an ip directly into phone 
> with correct settings.
>
> but when I access it via web browser it doesn't come up.
>
> Should I be able to access it.
>
>  
>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Dean 
> Collins
> *Sent:* Thursday, 3 July 2008 9:40 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] new install of asterisk appliance.
>
>  
>
> But where is the windows 2003 server?
>
>  
>
>
> Cheers,
>
> Dean
>
>

>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Sydney

> Web Hosting
> *Sent:* Wednesday, 2 July 2008 7:20 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* Re: [asterisk-users] new install of asterisk appliance.
>
>  
>
> No.
>
> ive got the netcomm router.
>
> of the netcomm router ive got 1 LAN port going into the WAN port of 
> the appliance.
>
> Then 1 LAn port from the appliance goes to my PC.
>
>  
>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Dean 
> Collins
> *Sent:* Thursday, 3 July 2008 8:22 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] new install of asterisk appliance.
>
>  
>
> Can you ping the ip address from another machine on your lan.
>
>  
>
>
> Cheers,
>
> Dean
>
>

>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Sydney

> Web Hosting
> 

Re: [asterisk-users] (no subject)

[Benjamin Jacob]

Use SendDTMF.



--- On Thu, 7/3/08, Neha Punia <[EMAIL PROTECTED]> wrote:

> From: Neha Punia <[EMAIL PROTECTED]>
> Subject: [asterisk-users] (no subject)
> To: "asterisk-users@lists.digium.com" 
> Date: Thursday, July 3, 2008, 10:29 AM
> Hi
> I  m making a call from one asterisk server to an asterisk
> client
> The call gets completed but I want it to send dtmf signals
> 
> The dialplan I have made for this is like
> exten => 205,1,Answer
> exten => 205,n,Wait(15)
> exten => 205,n,Playback(dtmf-1)
> exten => 205,n,Wait(20)
> 
> but it does not send any dtmf signal
> where is the problem??
> 
>  CAUTION - Disclaimer *
> This e-mail contains PRIVILEGED AND CONFIDENTIAL
> INFORMATION intended solely 
> for the use of the addressee(s). If you are not the
> intended recipient, please 
> notify the sender by e-mail and delete the original
> message. Further, you are not 
> to copy, disclose, or distribute this e-mail or its
> contents to any other person and 
> any such actions are unlawful. This e-mail may contain
> viruses. Infosys has taken 
> every reasonable precaution to minimize this risk, but is
> not liable for any damage 
> you may sustain as a result of any virus in this e-mail.
> You should carry out your 
> own virus checks before opening the e-mail or attachment.
> Infosys reserves the 
> right to monitor and review the content of all messages
> sent to or from this e-mail 
> address. Messages sent to or from this e-mail address may
> be stored on the 
> Infosys e-mail system.
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[asterisk-users] OLPC Sound Samples

[Tzafrir Cohen]

Hi

Slightly off-topic,

The OLPC (One Laptop Per Child, "100$ Laptop") project has announced a
collection of 10GB of sound samples:

  http://wiki.laptop.org/go/Sound_samples

License: CC-BY (explicitly allows public performance for commercial
purpose).

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] (no subject)

[Neha Punia]

Hi
I  m making a call from one asterisk server to an asterisk client
The call gets completed but I want it to send dtmf signals

The dialplan I have made for this is like
exten => 205,1,Answer
exten => 205,n,Wait(15)
exten => 205,n,Playback(dtmf-1)
exten => 205,n,Wait(20)

but it does not send any dtmf signal
where is the problem??

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[asterisk-users] asterisk queues and database backend (clustered realtime)

[Vieri]

If I define a queue like in:
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue

but instead I define queue_table as a MySQL ndbcluster table shared between two 
asterisk servers (both active and receiving calls), will the queue calls be 
handled coherently (or a pgcluster table if you prefer)?

Eg. will fields such as "queue_thereare" behave correctly if accessed 
simultaneously by two "clustered" Asterisk servers?

In other words, is anyone running successfully a setup such as:

PSTNPSTN
 |   |
Asterisk1   Asterisk2
 \   /
  Clustered Realtime
|  |
 SIP members  Queues

Is Asterisk (1.2 and 1.4) "ready" for "clustered realtime"?

Vieri



  

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Re: [asterisk-users] ooh323 doesn't know what to do when bridging calls?

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Kelvin Chan <[EMAIL PROTECTED]> wrote:
> Hi guys,
> 
> I'm trying out ooh323 and couldn't bridge ooh323 and sip/zap. 
> I'm using netmeeting and set gateway to my asterisk. 
> 
> Here's my CLI dump:
> 
>   == Spawn extension (h323, , 1) exited non-zero on
> 'OOH323/(null)-8c76'
> -- Executing [EMAIL PROTECTED]:1] Dial("OOH323/(null)-3074",
> "Zap/8/604xxx") in new stack
> -- Called 8/604xxx
> -- Zap/8-1 is ringing
> [2008-07-02 15:48:55] WARNING[21544]: channel.c:2390 ast_indicate_data:
> Unable to handle indication 3 for 'OOH323/(null)-3074'
> -- Zap/8-1 is ringing
> -- Zap/8-1 answered OOH323/(null)-3074
> [2008-07-02 15:49:08] WARNING[21544]: chan_ooh323.c:1053
> ooh323_indicate: Don't know how to indicate condition 20 on ooh323c_5
> 
> My ooh323.conf:
> 
> [general]
> bindaddr=192.168.1.9
> h323id=ObjSysAsterisk
> e164=100
> callerid=asterisk
> gatekeeper = DISABLE
> gateway = yes
> context = h323
> disallow = all
> allow = ulaw
> dtmfmode = rfc2833
> 
> 
> extensions.conf
> [h323]
> Exten => ,1,Dial(Zap/8/604xxx)
> Exten => ,n,Hangup
> 
> 604xxx goes to my cell. it rings fine but no audio. After I picked
> up from cell, netmeeting still shows "watiting for  to answer"
> message.
> 
> Any ideas?

I don't like the look of the (null) in the channel names.

If what you quoted was the whole of your ooh323.conf file, you don't have
any peer, user or friend sections. Try adding something like:

[h323gw]
type=friend
context=h323
ip=192.168.1.200  (or whatever the IP of your remote H323 endpoint is)
port=1720

If that still doesn't help, please mention what versions of asterisk and
asterisk-addons you are using.

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

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[asterisk-users] Dial function exit, go to line n+1

[Jerome Poggi]

Yesturday I found a bug in Asterisk, in particular in Dial application.
When the Dial function exit it want to branch to n+1, but if n+1 do not
exist, it exit from the context.

Example :

exten => s,5,ChanIsAvail(SIP/604,s)
exten => s,6,Dial(SIP/604,15,wotr)
exten => s,106,NoOp(Matthieu)
exten => s,n,ChanIsAvail(SIP/605,s)

Won't work because Dial exit to 7, and line 7 don't exist

but

exten => s,5,ChanIsAvail(SIP/604,s)
exten => s,6,Dial(SIP/604,15,wotr)
exten => s,7,NoOp(Nopnopnopnopnop)
exten => s,106,NoOp(Matthieu)
exten => s,n,ChanIsAvail(SIP/605,s)

Work, because line 7 exist

I use Asterisk 1.4.18

Jerome.

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Re: [asterisk-users] Call quality

[Loic Didelot]

Hello,
this is the case. Idle goes to 0% and IRQ goes to 100%.

I have a Junghanns ISDN card (bristuff) card. And I guess it is using
that Echo Canceler.

Best regards,
Loic Didelot.



On Thu, 2008-07-03 at 14:52 +1200, Matt Riddell wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Loic Didelot wrote:
> > Hi,
> > I am using g711a everywhere.
> > 
> > I checked on a completely idle system (no calls at all) and idle CPU is
> > dropping from 100% to 0% more than once per minute.
> 
> If you run top, and the idle goes to 0% is it the IRQ that is using the
> other 100%?
> 
> If so, what echo canceller are you using?
> 
> - --
> Kind Regards,
> 
> Matt Riddell
> Director
> ___
> 
> http://www.venturevoip.com (Great new VoIP end to end solution)
> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
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> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.7 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
> 
> iD8DBQFIbD7dDQNt8rg0Kp4RAql0AJ9hUDFqaNbliJTCLiKvR9BT+rbdNwCgqUjh
> tZxWREnPeYuO5h1PgrXxv30=
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> -END PGP SIGNATURE-
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-- 
Loïc DIDELOT
MIXvoip S.a.
[EMAIL PROTECTED]
http://www.mixvoip.com


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