Re: [Asterisk-Users] VoIP Provider problems

[Joseph Gutowski]

No, I'm not ignorant of how this works. You'll notice I put it
"appears" bad when I posted my results. Yes, it's not a perfect way to
show problems -- but taken with a grain of salt it's not half bad.
Especially when sampled over a longer period of time, and if the
original poster can correlate the PingPlotter results to the quality
of his calls.

Now if he shows 30% loss during good and bad calls, that's another story.

I posted my results to help the original poster. If he's trying to
troubleshoot an apparent bad connection with Sprint, he needs all the
help he can get. If they can proove the connection works even the
littlest bit, they'll say it's fine and blame Broadvoice.

If everyone gets similar levels of loss at those points, one could
conclude its a side effect of the routers having better things to do.
But if he's the only one showing them, then it would be a starting
point to conclude something is wrong with his connection or something
along Sprint's backbone.
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Re: [Asterisk-Users] Asterisk Discussion Form

[Francesco Peeters]

On Sun, April 3, 2005 5:42, [EMAIL PROTECTED] said:
> List:
>
> With recent discussions in regards to a forum, I have set-up a
> multi-faceted Asterisk and Open Source Discussion Board. The link is
> www.voipnewbie.com/forum It is open and ready for use.
>
> Enjoy!
>
> VoIPNewbie
>

Thank you for the effort... I will be sure to take a look there!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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Re: [Asterisk-Users] How does asterisk know the did called on?

[Damian Funnell]

Hi Courtney,

This really depends on the type of channel you are using.  We use CAPI
with one of our customers and they have ten DID's (DDI's in NZ) and use
four BRI lines.  Here is a snippet from their extensions.conf that
shows how it dials certain extensions based on the number dialled. The
three digit number in the dialling rules are presented by the ISDN
channel/CAPI and will probably be slightly different based on your
telephony provider.  We had to dial one of the DDI's and see Asterisk
reported it as being on the console to figure out how long the DDI
number was.

(Note that you would be much better off using macros rather than the
numerous entries that we have in extensions.conf, but we have
structured it like this upon customer request).

;[incoming]
exten => s,1,Answer
exten => s,2,Dial(SIP/9295,20,t,)
; No answer or busy, so dial other extension(s) - can add others to
this list if required
exten => s,3,Dial(SIP/9295&SIP/9287,,t,)    
;Queue not set up, do not use the following unless configuring it first.
;exten => s,3,Queue(receptionqueue)
;DDI's follow
;Manager (290)
exten => 290,1,Answer
exten => 290,2,Dial(SIP/9290,,t,)
exten => 290,3,Voicemail(9290)
exten => 290,4,Congestion
;Supervisor 1 (291)
exten => 291,1,Answer
exten => 291,2,Dial(SIP/9291,,t,)
exten => 291,3,Voicemail(9291)
exten => 291,4,Congestion

Hope this helps.
Damian.


FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz


Courtney Couch wrote:
If I were
to buy 20 did's how do I know within asterisk which number was dialed?
(like say I want a few of the did's to ring specific extensions if they
are dialed and others to go through the menu)
  
  
Is there any ${var} that has the number dialed in on? (that would be
optimum).
  
  
-Courtney.
  
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Re: [Asterisk-Users] Re: Are there online forums instead of, this email

[Jean-Michel Hiver]

 1. Support for message threads - replies to messages are shown right 
below the original message.
Any decent mailer has that functionality.
2. Support for subject matter sub forums - different message 
categories can be established.
You just split the list into differents mailing lists and you're done.
3. Built in search engine - messages relevant to the problem you are 
working on can easily be located.
You can either use your mailer search functionality on your existing 
archives or google on the web archives.

4. Moderated forums - postings and discussions can easily be 
supervised. Trouble makers can be banned from posting.
I'm grown up enough to deal with trouble makers myself. I don't want 
some daddy to decide what's good for me and what's not. Thanks.

(besides, mailing lists can be moderated too so this point is moot. You 
*really* should learn more about mailman...)

I have been involved in supporting another Linux based product for 
more than a year now via a proper web based support BBS setup, and it 
is a far superior approach to the Asterisk mailing list dump that this 
memo will end up in.
If you use digests, then you will get all messages in a big lumpy file. 
Don't use them, that's all. Again, you clearly show that you don't know 
squat about mailing lists.

Take a good look at:
http://www.astaro.org
It's a typical table based, invalid HTML, unusable, inaccessible web 
forum. It's just so nineties... thanks for making my point. My mail user 
agent provides a better interface than this website. If you can't 
understand why, maybe you should learn more about email and mailing lists...

* I have an asterisk subfolder which I call "useful" where I move 
messages which I find useful. Does your web thingy do that?

* I can go away for two weeks and see quickly messages which are read 
and messages which aren't. Does your web thingy do that?

* I can manage multiple mailing list through a single, simple, 
consistent interface - without having to remember different usernames / 
passwords. Does your web thingy do that?

This is the support BBS for Astaro Security Linux, a hardened Linux 
firewall distribution that delivers around 80% of the features of 
CheckPoint Software's Firewall-1, but at 20% of the cost. And they 
give away free licenses for non-commercial home use. Their support BBS 
currently has more than 9000 registered users.
I use firehol. I have 100% of the firewall functionality I've got for 
zero cost. And they have a mailing list too :)

It would do Digium well to establish a similar BBS, since it would 
dramatically ease the support issues for the membership. Running a web 
based BBS forum is not particularly load intensive, even if it ends up 
having many thousands of registered users.
Well maybe a web forum thingy would be good for people who buy support 
from digium. But even in the case of paid support, I think email 
combined with "RT" (request tracker) makes a superior product.

Regards,
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
---> http://ykoz.net/voip/max <---
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[Asterisk-Users] New to asterisk.

[Paul Lacatus]

Hi everybody,

I am very new in asterisk community. I just compiled & installed
asterisk on a fedora core 3 machine and I want for test purpose to do
a small PBX that use X-lite windows sip clients and no trunk for the
begining.

Where can I find a good how-to to do this job.  A small starting
how-to that let me understand the principles of setting a PBX with
asterisk. The handbook does not like starting guide.

PL
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[Asterisk-Users] How does asterisk know the did called on?

[Courtney Couch]

If I were to buy 20 did's how do I know within asterisk which number was 
dialed? (like say I want a few of the did's to ring specific extensions 
if they are dialed and others to go through the menu)

Is there any ${var} that has the number dialed in on? (that would be 
optimum).

-Courtney.
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[Asterisk-Users] sipcall.ch configuration problem

[gramels]

Hi,

i tried to connect my asterisk with sipcall.ch und run into the
following problem:


-- Executing Dial("SIP/2206-0df7", "SIP/[EMAIL PROTECTED]|240")
in new stack
-- Called [EMAIL PROTECTED]
Apr  2 11:40:41 NOTICE[32163]: chan_sip.c:6796 handle_response: Failed
to authenticate on INVITE to '"user"
;tag=as088a8ebb'


This is unclear to me, since the similar conf works with other
providers. Any ideas


sip.conf:
-
register => 41325111234:[EMAIL PROTECTED]/41325112342

[sipcall]
type=peer
secret=meinpasswort
username=41325112342
fromuser=41325112342
fromdomain=sip.backbone.ch
host=sip.backbone.ch
dtmfmode=rfc2833
context=sipcall-in
canreinvite=yes
qualify=1
nat=yes
insecure=very
-
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Re: [Asterisk-Users] Packetization

[Steven Critchfield]

On Sat, 2005-04-02 at 21:16 -0500, Matt wrote:
> I'm aware that asterisk only supports 20ms packetization rates.  Due
> to the fact that I will be using some voip devices on a wireless
> network which is highly sensative to framerate.. is there any way I
> can hard code the packetization rate at say 30 or 40ms and then
> compile astrisk?  If so, can anyone in the know tell me what variables
> I need to look at to change?

Are you sure your other devices support different packet sizes? Are you
sure the added delay in audio delivery can be handled decently and not
cause added echo? 

Have you considered what IAX trunking can do for you? It will reduce
frame rate as you add channels since each packet will then hold the
frames for each of the consecutive calls.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Fwd: NDN: Re: [Asterisk-Users] Delaying answer of incoming calls

[C F]

What can be done to this shmuck?
Everytime I post anything to the list I get one of these. I'm sure
I'll get one for posting this one as well.

-- Forwarded message --
From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
Date: Apr 3, 2005 12:24 AM
Subject: NDN: Re: [Asterisk-Users] Delaying answer of incoming calls
To: C F <[EMAIL PROTECTED]>


Sorry. Your message could not be delivered to:

Joshua Chessman (Mailbox or Conference is full.)
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Re: [Asterisk-Users] Delaying answer of incoming calls

[C F]

On Apr 2, 2005 5:42 AM, Justin Ridge <[EMAIL PROTECTED]> wrote:
> Hi all,
> 
> I'd like Asterisk to answer an incoming SIP call and bridge with a POTS
> line.  However, I do NOT want Asterisk to answer the SIP call unless the
> called POTS number answers (otherwise SIP caller is charged).
> 
> The wiki says this is the default behaviour, and so does this message
> http://lists.digium.com/pipermail/asterisk-users/2004-April/041881.html
> 
> But it's not working as expected.  The SIP call is answered immediately
> before the POTS line even starts ringing.
> 
> The dial plan is simple:
> [from-sip]
> exten => s,1,Dial(Zap/1/*829x,20)

The reason is because you are using Zap, a zap channel is answered as
soon (if you are using FXO moduels) as the line goes off hook. You can
use c in the dial string so that it is not considered answered until #
is pressed by the caller.
 
> I don't have "Answer" in the dial plan.  I don't think I changed any
> settings that would cause this non-default behaviour, but can you help me
> find why this is happening?
> 
> Thanks
> JR
> 
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Re: [Asterisk-Users] Integrating door intercom?

[C F]

On Apr 2, 2005 2:57 PM, Jeff Noxon <[EMAIL PROTECTED]> wrote:
> On Fri, Apr 01, 2005 at 04:07:30PM -0500, C F wrote:
> > Well, depends how you set it up. If you leave it as is, it will only
> > ring 3 times. You can't just call up the box (without the chip), b/c
> > it will just throw you to the other end of the doorbell fon (the co
> > port). So no this is one of the more cheaper one and I wouldn't
> > recommend it with Asterisk, try Vikingelectronics instead (the c2000
> > from them even support callerid). Or you can try Valcom.
> 
> In my configuration, the Doorbell Fon has a dedicated FXO port.  I would
> not recommend using it any other way.  Asterisk answers immediately
> when the user hits the button on the intercom and indicates ringing to
> the intercom.  Asterisk sets the Caller ID, and rings my house phones
> (and cell phone) with distinctive ring.
> 
> Personally I have no need or desire to make calls to the intercom box.
> Unless someone pushed the button, I'd be unaware anyone was there in
> the first place.
> 

Well, in most intercoms it's an option, to answer the intercom even
when nobody called it because you want to listen whats going on in
that room/door.

> > The distinctive ring doesn't really work with asterisk, since it is
> > never (well, almost never 1 out of 5 might repeat, but then again it
> > might switch the pattaren with the other box) exactly the same
> > pattaren. Caller ID just simply doesn't work with this box, it does'nt
> > send callerid, the only thing you acomplish by turning it off, is to
> > ring the phones imediatly.
> 
> There's no need for the distinctive ring to work with Asterisk unless you
> are trying to get by with one FXO port for both the doorbell and a POTS

Why not? it can support 2 doors, so you want to know which door is ringing.

> line.  I have not tested that configuration and would not recommend it.
> With a dedicated FXO interface for the doorbell, those issues go away.

They do not go away, look above, since it supports 2 doors you want to
be able to know which door rang.
 
C F
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Re: [Asterisk-Users] Macro Extension with Realtime and Mysql DB

[Matthew Boehm]

AFAIK, you would configure a macro extension in RealTime just like you
configure a regular extension/context in RealTime.

-Matthew


> From: kritikus Araklidas <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sun, 03 Apr 2005 04:11:38 +
> To: 
> Subject: [Asterisk-Users] Macro Extension with Realtime and Mysql DB
> 
> Hi Everyone:
> 
> I need to know if somebody know how to configure macro extension
> (extension.conf) in the database for Asterisk Realtime support if is
> suported.
> 
> Regards,
> 
> Kritikus
> 
> _
> Donât just search. Find. Check out the new MSN Search!
> http://search.msn.click-url.com/go/onm00200636ave/direct/01/
> 
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[Asterisk-Users] Replace Adtran 608 With Asterisk

[David]

Hello All, I'm new to VoIP. I have a friend that has an Adtran 608 with
6 lines over a T-1. He likes my Asterisk box. Could I replace the Adtran
608 with an Asterisk box??? Any ideas on an interface card??

Thanks, David

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RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W

[Paul Dracevich]

Hi ya I have also three of these phone, here is my entry in my sip.conf

[4701721]
type=friend
username=4701721
secret=password721
host=dynamic
canreinvite=no
context=internal
disallow=all
allow=g729
dtmfmode=rfc2833
qualify=4
permit=0.0.0.0/0.0.0.0
[EMAIL PROTECTED]



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ugur
GUNCER
Sent: Sunday, 3 April 2005 4:37 p.m.
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sip registration Problems With Zyxel P2000W

Hi all,

I bougth zyxel wifi phone but i  cant register 
when i want to register phone to asterisk i recieve 
These errors I spend 6 hours to fix regist problem but i cant find the
solution 

[9875]
type=friend
username=9875
secret=5789
host=dynamic
context=default
callerid="Ugur Guncer" <9875>
canreinvite=no
dtmfmode=rfc2833
nat=no






Sip read: 
REGISTER sip:213.139.225.82:5060 SIP/2.0
Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0
From: ;tag=5175B05114E474A31693
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 12 REGISTER
User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone
Contact: 
Expires: 300
Content-Length: 0


10 headers, 0 lines
Using latest request as basis request
Sending to 85.99.110.143 : 43956 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0
From: ;tag=5175B05114E474A31693
To: ;tag=as369f8960
Call-ID: [EMAIL PROTECTED]
CSeq: 12 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 85.99.110.143:43956
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0
From: ;tag=5175B05114E474A31693
To: ;tag=as369f8960
Call-ID: [EMAIL PROTECTED]
CSeq: 12 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
WWW-Authenticate: Digest realm="asterisk", nonce="0f3403ce"
Content-Length: 


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[Asterisk-Users] Sip registration Problems With Zyxel P2000W

[Ugur GUNCER]

Hi all,

I bougth zyxel wifi phone but i  cant register 
when i want to register phone to asterisk i recieve 
These errors I spend 6 hours to fix regist problem but i cant find the
solution 

[9875]
type=friend
username=9875
secret=5789
host=dynamic
context=default
callerid="Ugur Guncer" <9875>
canreinvite=no
dtmfmode=rfc2833
nat=no






Sip read: 
REGISTER sip:213.139.225.82:5060 SIP/2.0
Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0
From: ;tag=5175B05114E474A31693
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 12 REGISTER
User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone
Contact: 
Expires: 300
Content-Length: 0


10 headers, 0 lines
Using latest request as basis request
Sending to 85.99.110.143 : 43956 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0
From: ;tag=5175B05114E474A31693
To: ;tag=as369f8960
Call-ID: [EMAIL PROTECTED]
CSeq: 12 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 85.99.110.143:43956
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0
From: ;tag=5175B05114E474A31693
To: ;tag=as369f8960
Call-ID: [EMAIL PROTECTED]
CSeq: 12 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
WWW-Authenticate: Digest realm="asterisk", nonce="0f3403ce"
Content-Length: 


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Re: [Asterisk-Users] Asterisk Discussion Form

[gsmith]

Ty:

The whole purpose of the site (and the other sites that will be up
shortly) is to help newer users of Asterisk. The site will also have
sections devoted to those with more advanced levels of understanding.

Enjoy!

VoIPNewbie

> Thank you for your contribution Now maybe this is a good place where
> people can ask a question without getting slammed because they don't
> understand their own ignorance. :-)
>
>
>
> [EMAIL PROTECTED] wrote:
>
>>List:
>>
>>With recent discussions in regards to a forum, I have set-up a
>>multi-faceted Asterisk and Open Source Discussion Board. The link is
>>www.voipnewbie.com/forum It is open and ready for use.
>>
>>Enjoy!
>>
>>VoIPNewbie
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>>
>
>
>
>
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[Asterisk-Users] Macro Extension with Realtime and Mysql DB

[kritikus Araklidas]

Hi Everyone:
I need to know if somebody know how to configure macro extension 
(extension.conf) in the database for Asterisk Realtime support if is 
suported.

Regards,
Kritikus
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[Asterisk-Users] How to reset IAXy?

[Lam H. Nguyen]

Can anyone tell me how to reset the IAXy? I used I put
it the wrong ip config in the IAXy and it conflicts
with my network whenever I plug it in. Currently the
DHCP is disable. I need to re-enable it to change the
settings.

The hard reset button on the IAXy doesn't seem to work 



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Re: [Asterisk-Users] Asterisk Discussion Form

[Ty Carter]

Thank you for your contribution Now maybe this is a good place where 
people can ask a question without getting slammed because they don't 
understand their own ignorance. :-)


[EMAIL PROTECTED] wrote:
List:
With recent discussions in regards to a forum, I have set-up a
multi-faceted Asterisk and Open Source Discussion Board. The link is
www.voipnewbie.com/forum It is open and ready for use.
Enjoy!
VoIPNewbie
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[Asterisk-Users] Problem with asterisk -> ohphone

[cmisip]

I seem to be having problems calling ohphone from asterisk.  The audio
cuts out after a few seconds with the error from ohphone:

Incoming call from "root [192.168.0.1]" at Sat, 02 Apr 2005 22:50:55
-0500, answer call (Y/n)? Offhook - answering call
Started logical channel: sending G.711-uLaw-64k{hw} <1>
Call with "root [192.168.0.1]" established.
Accepting call.
Started logical channel: receiving G.711-uLaw-64k{hw} <6>
Onhook - ending call.
  2:17.142 H323 Cleaner   assert.cxx(105)   PWLib  
Assertion fail: Transmit media thread did not terminate, file
channels.cxx, line 680, Error=107
 
bort, ore dump, gnore?


Calls from ohphone to asterisk work well though.  Any ideas what is
going on?

Thanks.



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[Asterisk-Users] TE410P and Fax Server

[Ma Zhiyong]

Hi, All
   Can I use a TE410P card to make a FAX server? Did 
anybody have some experiences to construct PBX and Fax Server in one 
box?
   Thanks.
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Re: [Asterisk-Users] Asterisk Discussion Form * Should Read Asterisk Discussion Forum!*

[voipnewb]

> List:
>
> With recent discussions in regards to a forum, I have set-up a
> multi-faceted Asterisk and Open Source Discussion Board. The link is
> www.voipnewbie.com/forum It is open and ready for use.
>
> Enjoy!
>
> VoIPNewbie
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Pardon my mis-spelled title. It was supposed to read forum :)

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[Asterisk-Users] Asterisk Discussion Form

[voipnewb]

List:

With recent discussions in regards to a forum, I have set-up a
multi-faceted Asterisk and Open Source Discussion Board. The link is
www.voipnewbie.com/forum It is open and ready for use.

Enjoy!

VoIPNewbie
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[Asterisk-Users] Dialing w/analog phone via FXS port.

[Ken D'Ambrosio]

Argh.  I can't figure out what I'm doing wrong.  I can dial with my SIP
phones just fine, but I want to set up an analog phone plugged into my FXS
port... and, while it gets dialtone, no matter what digit I press, I get
stuff like:

VERBOSE[21963]: -- Starting simple switch on 'Zap/1-1'
DEBUG[21963]: DTMF digit: 9 on Zap/1-1
DEBUG[21963]: Hangup: channel: 1 index = 0, normal = 13, callwait = -1,
thirdcall = -1
DEBUG[21963]: Set option TDD MODE, value: OFF(0) on Zap/1-1
DEBUG[21963]: Updated conferencing on 1, with 0 conference usersApr  2
VERBOSE[21963]: -- Hungup 'Zap/1-1'

I've tried to make it as similar to the SIP stuff in zapata.conf as
possible.  Any suggestions on what to read to get this right?  I've RTFM'd
no small amount, but, obviously, not the *right* stuff.  I'll gladly send
my config files to anyone who wants 'em, or will gladly look at
functioning config files anyone wants to send my way.

Thanks!

-Ken
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[Asterisk-Users] D-Link router/Voip gateway locked to Lingo?

[Jason Brown]

Anyone have experience with these (DG-1402S) from D-Link?
Only ones I can find are locked to Lingo service. Are they unlockable or no? I
just want one to test out.






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RE: [Asterisk-Users] *** Asterisk 2.0 Stable release out now

[James Gardiner]

Yes, 1.2 is the closer reality.  I am keen to see 1.2 schedules for a
milestone release mainly because I feel, since V 1 was released, popularity
has sky rocketed, and as such  commercial implementations have also gone up.
With this, some fundamental needs have been looked into that Asterisk needs.
Nothing overt the top (Wait for a new version) but still important ones,
that I would like to see land into a Stable tree, bug fixes only.

So I sit and wait patiently..


James


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch
Sent: Sunday, 3 April 2005 5:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now

James Gardiner wrote:
> Asterisk 2.0 on Windows..  This is all very much a bit of a Joke, but one
> does beg to ask.
> 
> When will version 2.0 be released???
> 

2.0 is just now really being talked about in earnest.

I think a better question would be when 1.2 is going to be out.

That one has more narrow bounds.

B.
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[Asterisk-Users] Re: Buying some Polycom IP300s

[Noah Miller]

The page in the wiki used to say that the person would not recomed
Polycom phones to anyone.
I missed that part of the WIKI!  Maybe it's more recent.  I don't want 
to invalidate anybody's experience, but in the process of picking 
phones for my company, I evaluated a huge number of IP phones (and 
bought a lot of them).  The only ones I would consider for us to use in 
real production are the phones from Polycom, Cisco, Zultys, Snom, 
Avaya, and Pingtel.  Of those, the Polycom and Cisco units seemed to be 
the highest quality with the best feature set (I was looking for high 
quality speakerphone, multiple line appearances, centralized 
administration, intercom, lots of hard keys, and something that looked 
both sturdy and flashy).  I went with Polycom in the end because they 
are cheaper (actually, the cheapest in my list above).  They are NOT 
the easiest to configure, and they are NOT well supported, but they 
have certainly met my requirements and expectations., and once you have 
an FTP server set up with a good config, it is very fast and easy to 
set up new phones.


Also, is the IP500 worth the extra money?  What can it do that
the IP300 can't.
- Full duplex speakerphone (IP300 is listen only)
- Real graphical display (IP300 is more like an old-style calculator 
display)
- 3 line appearances (IP300 has 2)

From a qualitative perspective, the IP500's just look more 
professional.  The plastic stand is more solid and durable on the IP500 
than on the IP300.  The display is much more readable on the IP500.  
All our employees use IP500's or IP600's.  We only use the IP300's for 
lobby phones and in back rooms.  Depending on the supplier, you can get 
the IP500's for not much more than than the IP300's.  I would highly 
recommend the IP500's over the IP300.

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[Asterisk-Users] Packetization

[Matt]

I'm aware that asterisk only supports 20ms packetization rates. 
Due to the fact that I will be using some voip devices on a wireless
network which is highly sensative to framerate.. is there any way I can
hard code the packetization rate at say 30 or 40ms and then compile
astrisk?  If so, can anyone in the know tell me what variables I
need to look at to change?
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RE: [Asterisk-Users] Buying some Polycom IP300s

[Jim Sturtevant]

I’ve been using the IP 500 and like
it a great deal.  Be aware that the IP300 does not have a speaker phone.

 








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Re: [Asterisk-Users] SIP register more then 1 with same username

[Matthew Boehm]

Asterisk does not support multiple SIP registrations with same username.

-Matthew

> From: "Bashir Ullah - www.Lamsre.Com" <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sat, 2 Apr 2005 04:04:16 -0800
> To: 
> Subject: [Asterisk-Users] SIP register more then 1 with same username
> 
> Hi all * user
> 
> I did connected with * from 2  sip-softphone and i registered with asterisk
> under same username and password and working both fine. but * shows only
> one.
> 
> is there any way to find them both by using any tips.
> 
> Bashir
> 
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Re: [Asterisk-Users] Realtime

[Matthew Boehm]

Have you tried RealTime cacheing?

-Matthew

> From: David Choo <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sat, 2 Apr 2005 12:46:06 +0800
> To: 
> Subject: [Asterisk-Users] Realtime
> 
> Dear All,
> 
> I'm currently running Asterisk Relatime with pgSQL and unixODBC. However,
> I'm facing a little problem here. Whenever I reloads asterisk, current SIP
> registration will be lost. Is there anyway that I can retain this SIP
> registration after reloads, or at least set a timeout for reloads?
> 
> Forwarded are an example of what I mean.
> 
> Would appreciate any kind advise.
> 
> 
> === Pre Reload ===
> 
> 
> asterisk*CLI> sip show peers
> Name/username  HostDyn Nat ACL Mask
> Port Status
> 501/501192.168.100.153  D   N  255.255.255.255
> 5060 OK (84 ms)
> 708/708192.168.100.212  D   N  255.255.255.255
> 5060 OK (83 ms)
> 502/502192.168.100.196  D   N  255.255.255.255
> 5060 OK (79 ms)
> 103/103192.168.100.192  D   N  255.255.255.255
> 5060 OK (81 ms)
> 604/604192.168.100.198  D   N  255.255.255.255
> 5060 OK (126 ms)
> 701/701(Unspecified)D   N  255.255.255.255  0
> UNKNOWN
> 603/603192.168.100.203  D   N  255.255.255.255
> 5060 OK (80 ms)
> 602/602192.168.100.166  D   N  255.255.255.255
> 5060 OK (79 ms)
> 
> === Pre Reload ===
> 
> === Post Reload ===
> 
> asterisk*CLI> sip show peers
> Name/username  HostDyn Nat ACL Mask
> Port Status
> 0 sip peers [0 online , 0 offline]
> asterisk*CLI>
> 
> === Post Reload ===
> 
> 
> 
> Best Regards,
> 
> ==
> David Choo
> Systems Engineer
> Business & Technology Division
> "Engineered for Changing Businesses"
> Espore Corp Pte Ltd
> 68 Kallang Pudding Rd
> #04-03 SYH Logistics Bldg
> Singapore 349327
> Tel: 65-68487806
> Fax : 65-6842 2724
> E-mail :[EMAIL PROTECTED]
> =
> 
> Privileged/Confidential information may be contained in this message. If
> you are not the intended recipient, you must not copy it or use it for any
> purpose, nor deliver this message to anyone. Instead, please delete this
> message and destroy any other record of it immediately and kindly notify
> the sender by return email. Thank you for your co-operation.
> 
> Internet communications cannot be guaranteed to be secure or error-free as
> information could be intercepted, corrupted, lost, arrive late, or contain
> viruses. The sender therefore does not accept liability for any errors or
> omissions in the context of this message nor can the sender guarantee that
> this message is virus free.
> 
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Re: [Asterisk-Users] sangoma S508/FT1 ISA

[Cory Andrews]

Neal - If you have some specific questions you want to outline on this 
and email to me, I can get you the answers.  I know the folks at Sangoma 
have been very busy, they are usually very responsive to email inquiries 
in my experience.

Thanks
Cory Andrews
Senior Partner
+++
VOIPSupply.com
A Subsidiary of b2 Technologies
+++
voice - 800.398.VOIP X22
fax - 716.630.1548
email - [EMAIL PROTECTED]

Michael Bielicki wrote:
it wan't do channelised stuff so it want be of any use for voice
On Apr 1, 2005 2:46 AM, Neal Walton <[EMAIL PROTECTED]> wrote:
 

Hi,
Does anyone have any experience with the Sangoma S508/FT1?  I can't seem to
find very much information on it, and Sangoma has not responded to my
e-mail.  The Sangoma wanpipe driver doesn't seem to support TDM on this
card, but I feel certain that the card can handle it.  I hope someone knows
how to make this card work so I don't have to hack the firmware.
Regards,
Neal
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Re: [Asterisk-Users] Sangoma VS. Digium

[Cory Andrews]

I have been told the Digium echo cancellation product will be shipping 
soon, as well as a new version of the IAXy, which looks to have a new 
form factor, I'm guessing to help with heat dissipation on the units.

Cory Andrews
Senior Partner
+++
VOIPSupply.com
A Subsidiary of b2 Technologies
+++
voice - 800.398.VOIP X22
fax - 716.630.1548
email - [EMAIL PROTECTED]

Chris Modesitt wrote:
Eric Wrote:
 

Digium has a hardware echo can?
   

 

Not shipping, according to their online store.
   

Crap!, I spend all my time reading emails from this list, now I have to
check Digium's online store twice a day so I can get my hands on one of
those cards!!
Chris. 

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[Asterisk-Users] Long Distance Acces Code

[kritikus Araklidas]

Hi everyone:
I need some help for configuring access code for long distance call for each 
extension (Sip friend), i cannot find any clear information for asigned 
acces code for each user.

Any idea will be appreciated.
Kritikus.
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Re: [Asterisk-Users] Buying some Polycom IP300s

[Chris Scott]

Dan Morin wrote:
Sorry for the double post, I tried to paste and accidently sent the
email
 
I've been playing with Asterisk for a few weeks now, and I've gotten
everything to work well with softphones, so I'm ready to move on to
normal VoIP phones.  I've been looking around and reading comments that
people have had, and I was convinced that the Polycom IP300 was a great
phone for a good price.  But, then I ran into this page, which has been
update in the last few days:
 
http://www.voip-info.org/wiki-Polycom+SoundPoint+IP+500
 
The page in the wiki used to say that the person would not recomed
Polycom phones to anyone.  So anyway, I just want to make sure that the
IP300 is a good choice.  I don't want to get cheap phones that aren't
business quality, since I do play on using them for my business after
testing.  Also, is the IP500 worth the extra money?  What can it do that
the IP300 can't.  And finally, will the IP300 do ulaw encoding?
 
I saw that page after I got a IP 300 for testing last week.  So far, so 
good.  IIRC, it does ulaw and alaw--and others.  The sound quality is 
good, the display is good, and the volume gets plenty loud for me.  The 
only thing I don't like about it is the headset port is made for a 
"real" phone headset (RJ11 jack), not the kind you plug into a 
cell/cordless phone (2.5mm jack).

The only problem I had configuring it was MWI and the wiki page on MWI 
and Polycom helped but I didn't have the context set correctly in my 
sip.conf (I edited the wiki page to clarify what it needs to be).

To save you some time, I wouldn't suggest doing any config through the 
web interface.  Just set up the config files and make them avail. on FTP 
and then config. the phone to use them.  In the long run, you'll want to 
do this for multiple phones anyway and it gives you more control anyway.

--
Chris Scott
Adaptive Hosting Solutions, Inc.
http://www.adaptivehostingsolutions.com/
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Re: [Asterisk-Users] Re: Are there online forums instead of, this email

[Andrew Kohlsmith]

On April 2, 2005 07:30 pm, Tore Hansen wrote:
> Having read a number of mailing list memos on this subject, there is
> much to be said for having a proper support forum BBS, rather than
> getting an awkward long memo with a string of messages every 3 to 5 hours.

"awkward long memo with a string of messages" ?? don't use digests.  They're 
evil.

>   1. Support for message threads - replies to messages are shown right
> below the original message.
>
> 2. Support for subject matter sub forums - different message categories
> can be established.
>
> 3. Built in search engine - messages relevant to the problem you are
> working on can easily be located.
>
> 4. Moderated forums - postings and discussions can easily be supervised.
> Trouble makers can be banned from posting.

With the exception to #3 there is nothing you've mentioned that a mailing list 
doesn't already do.  And #4 I don't see as an advantage at all, as has been 
discussed ad nauseam in this thread already.

> It would do Digium well to establish a similar BBS, since it would
> dramatically ease the support issues for the membership. Running a web
> based BBS forum is not particularly load intensive, even if it ends up
> having many thousands of registered users.

And it has all the problems we've already discussed on this list many times.  
You've not given any example of how to overcome any of them.

> Since Asterisk is here to stay, why not get serious about the support,
> and do it right?

As we've said many times already...  You go do it.  If it's truly that much 
better, the subscribers will flock to them.  There have been numerous 
attempts over the last few years but the list still persists.

I really, *really* wish that the forum people would see that.  The forums that 
have been attempted must be missing *something* for them to fail.  What is 
it?  I believe it's the sheer simplicity, clarity, offline capabilities and 
semi-decentralized nature that keeps it strong.  But hey, feel free to prove 
me wrong.  Competition's healthy.

-A.
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RE: [Asterisk-Users] problem detecting answer on pri card

[Richard]

> 
> Richard wrote:
> 
> >A debug on the pri shows,
> >Ext: 1  Progress Description: Call is not end-to-end ISDN; further call
> >progress information may be available inband. (1) ]
> >
> >So maybe the inband information is not detected by *?
> >
> >
> I can't help you debug, but I see this same progress message and can
> hear their system fine.  No more messages appear until I hangup and a
> DISCONNECT goes out.
> 

Thanks, I upgraded to a newer version and the problem is fixed.

Richard


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[Asterisk-Users] Re: Are there online forums instead of, this email

[Tore Hansen]

Having read a number of mailing list memos on this subject, there is 
much to be said for having a proper support forum BBS, rather than 
getting an awkward long memo with a string of messages every 3 to 5 hours.

Some of the advantages of a proper support forum BBS are:
 1. Support for message threads - replies to messages are shown right 
below the original message.

2. Support for subject matter sub forums - different message categories 
can be established.

3. Built in search engine - messages relevant to the problem you are 
working on can easily be located.

4. Moderated forums - postings and discussions can easily be supervised. 
Trouble makers can be banned from posting.

I have been involved in supporting another Linux based product for more 
than a year now via a proper web based support BBS setup, and it is a 
far superior approach to the Asterisk mailing list dump that this memo 
will end up in. Take a good look at:

http://www.astaro.org
This is the support BBS for Astaro Security Linux, a hardened Linux 
firewall distribution that delivers around 80% of the features of 
CheckPoint Software's Firewall-1, but at 20% of the cost. And they give 
away free licenses for non-commercial home use. Their support BBS 
currently has more than 9000 registered users.

It would do Digium well to establish a similar BBS, since it would 
dramatically ease the support issues for the membership. Running a web 
based BBS forum is not particularly load intensive, even if it ends up 
having many thousands of registered users.

Since Asterisk is here to stay, why not get serious about the support, 
and do it right?

Tore Hansen
Calgary, Alberta
Canada
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[Asterisk-Users] Buying some Polycom IP300s

[Dan Morin]

Sorry for the double 
post, I tried to paste and accidently sent the email
 
I've been playing with Asterisk for a few weeks now, 
and I've gotten everything to work well with softphones, so I'm ready to move on 
to normal VoIP phones.  I've been looking around and reading comments that 
people have had, and I was convinced that the Polycom IP300 was a great phone 
for a good price.  But, then I ran into this page, which has been update in 
the last few days:
 
http://www.voip-info.org/wiki-Polycom+SoundPoint+IP+500
 
The page in the wiki used to say that the person would 
not recomed Polycom phones to anyone.  So anyway, I just want to make sure 
that the IP300 is a good choice.  I don't want to get cheap phones that 
aren't business quality, since I do play on using them for my business after 
testing.  Also, is the IP500 worth the extra money?  What can it do 
that the IP300 can't.  And finally, will the IP300 do ulaw 
encoding?
 
Thanks in 
advance.
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Re: [Asterisk-Users] xlite registration fails but calls to thru

[Scott Wolfe]

Title: Re: [Asterisk-Users] xlite registration fails but calls to thru



No. They are all there as shown in your image. 


  - Original Message - 
  From: 
  Robert Keller 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Saturday, April 02, 2005 4:12 
  PM
  Subject: Re: [Asterisk-Users] xlite 
  registration fails but calls to thru
  Make 
  sure the first three codecs are not grayed out.Robert Andrew Keller 
  Ferndale School District #502[EMAIL PROTECTED]360-383-9228 
  PH.360-383-9218 FAX"Paving the way for tomorrows genius."
  
  From: "Scott Wolfe" <[EMAIL PROTECTED]>Reply-To: 
  Asterisk Users Mailing List - Non-Commercial Discussion Date: 
  Sat, 2 Apr 2005 16:03:19 -0800To: 
  Subject: 
  [Asterisk-Users] xlite regestration fails but calls to 
  thruWhile 
  on my network I can register ok with xlite but outside my firewall my Xlite 
  says that regestraion has failed but I am still able to make calls through it. 
  I have opened ports: 5060 udp/tcp and 1-2 udp/tcp  is there 
  another port Xlite needs for proper regestration? Is is this a network 
  configuation error on Astrisks part? My Asterisk server is running a IP of 
  10.0.1.x and my Cisco firewall is passing the public IP address to it from the 
  outside. Thanks for 
  any advice. -Scott
  
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[Asterisk-Users] Buying some Polycom IP300s

[Dan Morin]

I've been playing 
with Asterisk for a few weeks now, and I've gotten everything to work well with 
softphones, so I'm ready to move on to normal VoIP phones.  I've been 
looking around and reading comments that people have had, and I was convinced 
that the Polycom IP300 was a great phone for a good price.  But, then I ran 
into this page, which has been update in the last few days:
 
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Re: [Asterisk-Users] xlite registration fails but calls to thru

[Robert Keller]

Title: Re: [Asterisk-Users] xlite registration fails but calls to thru





Make sure the first three codecs are not grayed out.


Robert Andrew Keller 
Ferndale School District #502
[EMAIL PROTECTED]
360-383-9228 PH.
360-383-9218 FAX
"Paving the way for tomorrows genius."

From: "Scott Wolfe" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
Date: Sat, 2 Apr 2005 16:03:19 -0800
To: 
Subject: [Asterisk-Users] xlite regestration fails but calls to thru

While on my network I can register ok with xlite but outside my firewall my Xlite says that regestraion has failed but I am still able to make calls through it. I have opened ports: 5060 udp/tcp and 1-2 udp/tcp  is there another port Xlite needs for proper regestration? Is is this a network configuation error on Astrisks part? My Asterisk server is running a IP of 10.0.1.x and my Cisco firewall is passing the public IP address to it from the outside. 
 
Thanks for any advice.
 -Scott
 

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[Asterisk-Users] xlite regestration fails but calls to thru

[Scott Wolfe]

While on my network I can register ok with xlite 
but outside my firewall my Xlite says that regestraion has failed but I am still 
able to make calls through it. I have opened ports: 5060 udp/tcp and 1-2 
udp/tcp  is there another port Xlite needs for proper regestration? Is is 
this a network configuation error on Astrisks part? My Asterisk server is 
running a IP of 10.0.1.x and my Cisco firewall is passing the public IP address 
to it from the outside. 
 
Thanks for any advice.
 -Scott
 
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Re: [Asterisk-Users] OH323 core dump

[Yves]

You can report this here :
https://skylab.inaccessnetworks.com/mantis/main_page.php


Dipole Moment wrote:
> Hi all,
> 
> I installed and configured OH323 driver and have been using it for a
> week now, it's been working great but it also seems to be crashing
> Asterisk once in a while.  I wasn't sure until I started asterisk with
> safe_asterisk script and found a core dump in /tmp today.  Asterisk
> version is 1.0.6 and OH323 driver version is 0.6.5.  Below is the
> backtrace of the core dump.  Does anyone know if 0.7.1 with CVS-HEAD
> fixes this problem?  As usual in such cases, my excuse is that I don't
> have enough time otherwise I'd have traced the problem and fixed it :)
> 
> #0  0x0188c3fd in RTP_JitterBuffer::Main ()
>from /usr/lib/asterisk/modules/chan_oh323.so
> (gdb) bt
> #0  0x0188c3fd in RTP_JitterBuffer::Main ()
>from /usr/lib/asterisk/modules/chan_oh323.so
> #1  0x01950e6a in PThread::PX_ThreadStart ()
>from /usr/lib/asterisk/modules/chan_oh323.so
> #2  0x00c8298c in start_thread () from /lib/tls/libpthread.so.0
> #3  0x00bdd7da in clone () from /lib/tls/libc.so.6
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> 

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[Asterisk-Users] Sipura - GSM or iLBC?

[Matt]

Hi,
Does anyone know... does Sipura have any plans to support GSM or iLBC on any of their devices?  Specifically the ATA-2000?
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[Asterisk-Users] Re: Polycom sound quality problems

[Noah Miller]

There's no transcoding going on.  It's ulaw on IAX with Sixtel and ulaw
on SIP to the phone.  I considered that as a possibility originally, 
and
even tried using GSM with Sixtel to force it to do transcoding, but had
the exact same problem.

The Asterisk box is a 2.4ghz P4 with 512MB RAM, doing nothing but
Asterisk.  I have only 9 extensions.
I would think there's a possibility of packet loss on the IAX channel,
except the other SIP phones (SJPhone softphone) work flawlessly.  Also,
OUTBOUND calls are just fine on the Polycoms.  Only incoming calls are
messed up.
Just to cover all the bases, have you tried any other IAX providers or 
connections?

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[Asterisk-Users] OH323 core dump

[Dipole Moment]

Hi all,

I installed and configured OH323 driver and have been using it for a
week now, it's been working great but it also seems to be crashing
Asterisk once in a while.  I wasn't sure until I started asterisk with
safe_asterisk script and found a core dump in /tmp today.  Asterisk
version is 1.0.6 and OH323 driver version is 0.6.5.  Below is the
backtrace of the core dump.  Does anyone know if 0.7.1 with CVS-HEAD
fixes this problem?  As usual in such cases, my excuse is that I don't
have enough time otherwise I'd have traced the problem and fixed it :)

#0  0x0188c3fd in RTP_JitterBuffer::Main ()
   from /usr/lib/asterisk/modules/chan_oh323.so
(gdb) bt
#0  0x0188c3fd in RTP_JitterBuffer::Main ()
   from /usr/lib/asterisk/modules/chan_oh323.so
#1  0x01950e6a in PThread::PX_ThreadStart ()
   from /usr/lib/asterisk/modules/chan_oh323.so
#2  0x00c8298c in start_thread () from /lib/tls/libpthread.so.0
#3  0x00bdd7da in clone () from /lib/tls/libc.so.6
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Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now

[John Novack]

Brian Capouch wrote:
James Gardiner wrote:
Asterisk 2.0 on Windows..  This is all very much a bit of a Joke, but 
one
does beg to ask.
When will version 2.0 be released???

2.0 is just now really being talked about in earnest.
I think a better question would be when 1.2 is going to be out.
An even BETTER question is: When will what is already out and more or 
less working have enough accurate documentation to make it acceptable to 
a wider audience?

As one small example: the recent postings regarding wctdm. If all the 
options are at the end of the driver source, how long does it take to 
put into a more accessible form?

JMO
John N

That one has more narrow bounds.
B.
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Re: [Asterisk-Users] Book Review: VoIP Telephony with Asterisk

[John Novack]

Jean-Michel Hiver wrote:
Kerry Garrison wrote:
The book bills itself as a beginner's guide to Asterisk and Voice 
over IP (VoIP). Even with over 270 pages, it isn't possible to go 
through every single feature that Asterisk has to offer but the book 
does give enough information to get you started and even apply a few 
advanced features to your phone system. For those of you not familiar 
with Asterisk, VoIP, or PBX's we will need a little bit of background 
for you to know if this book is for you.
http://www.geekgazette.com/index.php?option=com_content&task=view&id=23&Itemid=26 
 


I don't know if you've read the book. I have it on my desk right now 
and it suffers from out of date information, lack of structure, lack 
of progression. I felt that the book is mostly just a clunky bunch of 
recipies patched together.
Agreed It seems to mostly be a rehash of the early Asterisk 
documentation, with only a few tidbits that I had not found elsewhere.
Certainly not worth it's high pricetag either.

Keeping up with such a moving target is indeed a difficult task for ANY 
printed book.
Digium's  published work is at least worth the price!

John N

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Re: [Asterisk-Users] Shorewall firewall rules

[Remco Barende]

The exact rtp port to be used by each sip device never became a
standard in the rfc, so each vendor is allowed to chose whatever
udp port range they felt like using as their default.
Opening udp ports from 1024 to 64000 will likely help, but you might
as well dump the firewall if you're going to open everything like
that.
Also note that each line/conversation will use another udp port.
So, in the case of the xten product, the first line/conversation
may use port 8000. If you put that line on hold and start another
(second) rtp session, that line/conversation will use something
like 8002 (or whatever).
Thanks for all the replies. I did manage to get it working now but do not 
feel very comfortable will all the ports that must be opened.

Thanks again!
Remco
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Re: [Asterisk-Users] Re: Polycom sound quality problems

[Eric Mason]

There's no transcoding going on.  It's ulaw on IAX with Sixtel and ulaw 
on SIP to the phone.  I considered that as a possibility originally, and 
even tried using GSM with Sixtel to force it to do transcoding, but had 
the exact same problem. 

The Asterisk box is a 2.4ghz P4 with 512MB RAM, doing nothing but 
Asterisk.  I have only 9 extensions.

I would think there's a possibility of packet loss on the IAX channel, 
except the other SIP phones (SJPhone softphone) work flawlessly.  Also, 
OUTBOUND calls are just fine on the Polycoms.  Only incoming calls are 
messed up.


Max W Blackmer Jr wrote:
I don't see any way to tell the Polycom to "ignore" QoS.  It's mainly
routers and switches that pay attention to QoS, the phone would just set
QoS on its outgoing packets.  Anyway, here's what's in the QoS section-
it all seems to be related to sending packets:
   

It is not in the transport if it is sounding bad look and see if
there is any transcoding occuring from the IAX to the SIP. What codecs
are accepted on the AIX should be the Same codecs accepted on the SIP
channel ... and what codects are being used on each phone. This sounds
like a transcoding issue.
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Re: [Asterisk-Users] Book Review: VoIP Telephony with Asterisk

[Jean-Michel Hiver]

Kerry Garrison wrote:
The book bills itself as a beginner's guide to Asterisk and Voice over 
IP (VoIP). Even with over 270 pages, it isn't possible to go through 
every single feature that Asterisk has to offer but the book does give 
enough information to get you started and even apply a few advanced 
features to your phone system. For those of you not familiar with 
Asterisk, VoIP, or PBX's we will need a little bit of background for 
you to know if this book is for you.
http://www.geekgazette.com/index.php?option=com_content&task=view&id=23&Itemid=26 

I don't know if you've read the book. I have it on my desk right now and 
it suffers from out of date information, lack of structure, lack of 
progression. I felt that the book is mostly just a clunky bunch of 
recipies patched together.

Of course I appreciate how hard it must be to write a book such as this. 
Yet, I think Digium's PDF handbook has better value, regardless of the 
book's price.

Cheers,
Jean-Michel.
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
---> http://ykoz.net/voip/max <---
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Re: [Asterisk-Users] problem detecting answer on pri card

[Trevor Peirce]

Richard wrote:
A debug on the pri shows,
Ext: 1  Progress Description: Call is not end-to-end ISDN; further call
progress information may be available inband. (1) ]
So maybe the inband information is not detected by *?
 

I can't help you debug, but I see this same progress message and can 
hear their system fine.  No more messages appear until I hangup and a 
DISCONNECT goes out.


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Re: [Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian

[Josh Alberts]

By default, synaptic puts init scripts in all runlevel folders.  All are
exactly the same, except for init 6, which is supposed to kill the
process.

On Sat, 02 Apr 2005 12:20:40 -0600,
ron at wellsted.org.uk said:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Josh Alberts wrote:
> I'm having trouble getting asterisk to run at startup using Ubuntu. 
> I've checked, and the asterisk dameon is set to run at init 5.  However,
> I'm not seeing anything that says that asterisk has been started during
> the boot process.  Oddly, when I shut the machine down/run init6, it
> says Starting Asterisk PBX.  Odd.  I'm using the default scripts that
> came with asterisk (I installed using synaptic and the debian universe
> repositories).  I've edited /etc/default/asterisk, uncommented the first
> line and changed start asterisk to "yes".  Anybody know what might be
> wrong?

Try starting atserisk at run level 2 or 3 (debian/ubuntu does this a
little differently)


- --
Ron Wellsted
http://www.wellsted.org.uk
ron at wellsted.org.uk
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
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[Asterisk-Users] Cisco Skinny Call Control Protocol on Asterisk

[Alexandre Otto Durr]

Hi for all!

I saw it on http://signate.com/features.php an Open Source PBX Features with
support Cisco Skinny Call Control Protocol.

Is it possible in Asterisk or I need a license for this?

Has anyone using Asterisk with Cisco Skinny?

TIA

Alexandre 


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[Asterisk-Users] wctdm module parameters (Was: Issues with ringing on FXS ports)

[Richard Scobie]

[EMAIL PROTECTED] wrote:
Is there a list of these anywhere?  This is now the third one I've heard 
of, with no documentation:  lowpower (IIRC), robust and now boostringer. 
Do I have to go diving in the source, or is there a Wiki I can't find?
I have only ever found the information in the driver source of on the 
CVS list as they have been added.

There is a list of them at the end of wctdm.c.
The non obvious ones I know about:
opermode=COUNTRY
Where COUNTRY is one from the list near the top of wctdm.c This will set 
the A.C. and D.C. line impedance on the FXO modules to suit the telecom 
standard used in that country. Default if not set, is FCC (US/Canada).

fxshonormode=1
If used, it must be in conjuction with the above. This will set the A.C. 
and D.C. line impedance on the FXS modules to match COUNTRY. Default if 
not set, is FCC (US/Canada).

lowpower=1
Reduces ringing volts on FXS to 50V peak.
boostringer=1
Boosts ringing volts on FXS to 89V peak.
Regards,
Richard
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[Asterisk-Users] Re: how to tell what ${DIALSTATUS} is being set

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Randy Bush <[EMAIL PROTECTED]> wrote:
> i often have nufone problems, e.g.
> 
> -- Executing Dial("SIP/konaa0p-4b88", "IAX2/[EMAIL 
> PROTECTED]/14086661234") in new stack
> -- Called [EMAIL PROTECTED]/14086661234
> -- Call accepted by 66.225.202.72 (format ulaw)
> -- Format for call is ulaw
> -- Hungup 'IAX2/NuFone/5'
> 
> 
> 
>   == No one is available to answer at this time
> -- Executing Hangup("SIP/konaa0p-4b88", "") in new stack
>   == Spawn extension (dial-gateways, 14086661234, 5) exited non-zero on 
> 'SIP/konaa0p-4b88'
> -- Executing Hangup("SIP/konaa0p-4b88", "") in new stack
>   == Spawn extension (dial-gateways, h, 1) exited non-zero on 
> 'SIP/konaa0p-4b88'
> 
> i would like to detect this (and many other things) in ${DIALSTATUS}
> conditions so that i can then GotoIf() them.  the problem is that the
> log does not tell me explicitly which ${DIALSTATUS} has been returned,
> leaving me guessing.  with BUSY vs CONGESTION this is even more of an
> issue.
> 
> is it reasonable to ask that the log contain the value being set in
> ${DIALSTATUS}?

I find it useful to follow the Dial command with a NoOp command as follows:
 NoOp(DIALSTATUS=${DIALSTATUS})

Then it shows up in the log. It also provides something for my Manager event
parser to see, in order to discover the reason for a failed call.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Users] how to tell what ${DIALSTATUS} is being set

[[EMAIL PROTECTED]]

Have you tried a simple hangup NoOp to output ${DIALSTATUS} to the CLI?

exten => h,1,NoOp("${DIALSTATUS}")

-josiah


Original Message:
-
From: Randy Bush [EMAIL PROTECTED]
Date: Sat, 2 Apr 2005 10:44:46 -0800
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] how to tell what ${DIALSTATUS} is being set


i often have nufone problems, e.g.

-- Executing Dial("SIP/konaa0p-4b88", "IAX2/[EMAIL PROTECTED]/14086661234")
in new stack
-- Called [EMAIL PROTECTED]/14086661234
-- Call accepted by 66.225.202.72 (format ulaw)
-- Format for call is ulaw
-- Hungup 'IAX2/NuFone/5'



  == No one is available to answer at this time
-- Executing Hangup("SIP/konaa0p-4b88", "") in new stack
  == Spawn extension (dial-gateways, 14086661234, 5) exited non-zero on
'SIP/konaa0p-4b88'
-- Executing Hangup("SIP/konaa0p-4b88", "") in new stack
  == Spawn extension (dial-gateways, h, 1) exited non-zero on
'SIP/konaa0p-4b88'

i would like to detect this (and many other things) in ${DIALSTATUS}
conditions so that i can then GotoIf() them.  the problem is that the
log does not tell me explicitly which ${DIALSTATUS} has been returned,
leaving me guessing.  with BUSY vs CONGESTION this is even more of an
issue.

is it reasonable to ask that the log contain the value being set in
${DIALSTATUS}?

randy

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Re: [Asterisk-Users] Zaptel Anti-MMX Optimizations

[Trevor Peirce]

Michael D Schelin wrote:
Hay Trevor, what would be the problem if you were using AMD processors?
/*
* Define if you want MMX optimizations in zaptel
*
* Note: CONFIG_ZAPTEL_MMX is generally incompatible with AMD
* processors and can cause system instability!
*
*/
/* #define CONFIG_ZAPTEL_MMX */
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Re: [Asterisk-Users] Zaptel Anti-MMX Optimizations

[Michael D Schelin]

Hay Trevor, what would be the problem if you were using AMD processors?
Trevor Peirce wrote:
Hello,
Running Fedora Core 2 with a Celeron processor I'm seeing a 
significant problem when enabling MMX optimizations.

I'll gladly submit a bug report, but I don't know what information is 
useful.

First, with -no- MMX optimizations enabled, here is what I see
CLI> show translation
   Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
   g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
 g723 - - - - - - - - - - -
  gsm - - 2 2 3 2 11118 -24
 ulaw - 5 - 1 3 2 11118 -24
 alaw - 5 1 - 3 2 11118 -24
 g726 - 6 3 3 - 3 21219 -25
adpcm - 5 2 2 3 - 11118 -24
 slin - 4 1 1 2 1 -1017 -23
lpc10 - 7 4 4 5 4 3 -20 -26
 g729 - 7 4 4 5 4 313 - -26
speex - - - - - - - - - - -
 ilbc - 8 5 5 6 5 41421 - -
If I enable MMX_OPTIMIZATIONS (and change nothing else), it gets quite 
worse

CLI> show translation
   Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
   g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
 g723 - - - - - - - - - - -
  gsm - - 2 2 3 2 14820 - 446270
 ulaw - 5 - 1 3 2 14820 - 446270
 alaw - 5 1 - 3 2 14820 - 446270
 g726 - 6 3 3 - 3 24921 - 446271
adpcm - 5 2 2 3 - 14820 - 446270
 slin - 4 1 1 2 1 -4719 - 446269
lpc10 -   368   365   365   366   365   364 -   383 - 446633
 g729 -23202021201966 - - 446288
speex - - - - - - - - - - -
 ilbc -   266   263   263   264   263   262   309   281 - -
And just incase you think I've got an AMD, here you go...
processor   : 0
vendor_id   : GenuineIntel
cpu family  : 15
model   : 1
model name  : Intel(R) Celeron(R) CPU 1.70GHz
stepping: 3
cpu MHz : 1716.114
cache size  : 128 KB
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 2
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca 
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm
bogomips: 3399.68

If this isn't supposed to work with Celerons I'd like to update the 
documentation. On the other hand if it's a problem with my system I'd 
like to resolve it :)

TIA!
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Re: [Asterisk-Users] Integrating door intercom?

[Jeff Noxon]

On Fri, Apr 01, 2005 at 04:07:30PM -0500, C F wrote:
> Well, depends how you set it up. If you leave it as is, it will only
> ring 3 times. You can't just call up the box (without the chip), b/c
> it will just throw you to the other end of the doorbell fon (the co
> port). So no this is one of the more cheaper one and I wouldn't
> recommend it with Asterisk, try Vikingelectronics instead (the c2000
> from them even support callerid). Or you can try Valcom.

In my configuration, the Doorbell Fon has a dedicated FXO port.  I would
not recommend using it any other way.  Asterisk answers immediately
when the user hits the button on the intercom and indicates ringing to
the intercom.  Asterisk sets the Caller ID, and rings my house phones
(and cell phone) with distinctive ring.

Personally I have no need or desire to make calls to the intercom box.
Unless someone pushed the button, I'd be unaware anyone was there in
the first place.

> The distinctive ring doesn't really work with asterisk, since it is
> never (well, almost never 1 out of 5 might repeat, but then again it
> might switch the pattaren with the other box) exactly the same
> pattaren. Caller ID just simply doesn't work with this box, it does'nt
> send callerid, the only thing you acomplish by turning it off, is to
> ring the phones imediatly.

There's no need for the distinctive ring to work with Asterisk unless you
are trying to get by with one FXO port for both the doorbell and a POTS
line.  I have not tested that configuration and would not recommend it.
With a dedicated FXO interface for the doorbell, those issues go away.

Regards,

Jeff
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[Asterisk-Users] Zaptel Anti-MMX Optimizations

[Trevor Peirce]

Hello,
Running Fedora Core 2 with a Celeron processor I'm seeing a significant 
problem when enabling MMX optimizations.

I'll gladly submit a bug report, but I don't know what information is 
useful.

First, with -no- MMX optimizations enabled, here is what I see
CLI> show translation
   Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
   g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
 g723 - - - - - - - - - - -
  gsm - - 2 2 3 2 11118 -24
 ulaw - 5 - 1 3 2 11118 -24
 alaw - 5 1 - 3 2 11118 -24
 g726 - 6 3 3 - 3 21219 -25
adpcm - 5 2 2 3 - 11118 -24
 slin - 4 1 1 2 1 -1017 -23
lpc10 - 7 4 4 5 4 3 -20 -26
 g729 - 7 4 4 5 4 313 - -26
speex - - - - - - - - - - -
 ilbc - 8 5 5 6 5 41421 - -
If I enable MMX_OPTIMIZATIONS (and change nothing else), it gets quite 
worse

CLI> show translation
   Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
   g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
 g723 - - - - - - - - - - -
  gsm - - 2 2 3 2 14820 - 446270
 ulaw - 5 - 1 3 2 14820 - 446270
 alaw - 5 1 - 3 2 14820 - 446270
 g726 - 6 3 3 - 3 24921 - 446271
adpcm - 5 2 2 3 - 14820 - 446270
 slin - 4 1 1 2 1 -4719 - 446269
lpc10 -   368   365   365   366   365   364 -   383 - 446633
 g729 -23202021201966 - - 446288
speex - - - - - - - - - - -
 ilbc -   266   263   263   264   263   262   309   281 - -
And just incase you think I've got an AMD, here you go...
processor   : 0
vendor_id   : GenuineIntel
cpu family  : 15
model   : 1
model name  : Intel(R) Celeron(R) CPU 1.70GHz
stepping: 3
cpu MHz : 1716.114
cache size  : 128 KB
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 2
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca 
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm
bogomips: 3399.68

If this isn't supposed to work with Celerons I'd like to update the 
documentation. On the other hand if it's a problem with my system I'd 
like to resolve it :)

TIA!
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[Asterisk-Users] Book Review: VoIP Telephony with Asterisk

[Kerry Garrison]

The book bills itself as a beginner's guide to Asterisk and Voice over IP 
(VoIP). Even with over 270 pages, it isn't possible to go through every single 
feature that Asterisk has to offer but the book does give enough information to 
get you started and even apply a few advanced features to your phone system. For 
those of you not familiar with Asterisk, VoIP, or PBX's we will need a little 
bit of background for you to know if this book is for you. 
http://www.geekgazette.com/index.php?option=com_content&task=view&id=23&Itemid=26
 
-Kerry
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RE: [Asterisk-Users] problem detecting answer on pri card

[Richard]

Both use national as switchtype. I put a traditional pbx to the circuit.
Everything is working. Any suggestion?

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Alex Vishnev
> Sent: Saturday, April 02, 2005 9:26 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] problem detecting answer on pri card
> 
> I have seen that before when you mismatch the type of pri flavor. For
> example, if you carrier gives you 4ess and you put 5ess in your config.
> There are subtle differences in packets. I would check the configuration
> on
> your carrier side and * side.
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Richard
> Sent: Saturday, April 02, 2005 1:20 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] problem detecting answer on pri card
> 
> Hi,
> 
> I have a digium PRI T1 card connecting to my carrier. However it has
> problems to detect the answer signal on some numbers. For example,
> 1-800-225-2525 is KLM airline's reservation line. It should answer right
> away. But * can't detect it is answered and keeps ringing the ip phone. I
> put a monitor on the channel, and get the answer messages in the channels.
> So somehow the line is answered but * doesn't know. I don't have a problem
> to most numbers. The problem only got my attention after one customer
> reported it.
> 
> A debug on the pri shows,
> Ext: 1  Progress Description: Call is not end-to-end ISDN; further call
> progress information may be available inband. (1) ]
> 
> So maybe the inband information is not detected by *?
> 
> Anyone has the same setup, i.e. PRI to your carrier? Can you please dial
> the
> number 1-800-225-2525 and have 'pri debug'? I'd like to compare the
> results.
> I am not sure if it is * or just my * configuration.
> 
> Your help is highly appreciated. I am really stuck here.
> 
> Thanks,
> Richard
> 
> 
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RE: [Asterisk-Users] problem detecting answer on pri card

[Alex Vishnev]

I have seen that before when you mismatch the type of pri flavor. For
example, if you carrier gives you 4ess and you put 5ess in your config.
There are subtle differences in packets. I would check the configuration on
your carrier side and * side. 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Sent: Saturday, April 02, 2005 1:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] problem detecting answer on pri card

Hi,

I have a digium PRI T1 card connecting to my carrier. However it has
problems to detect the answer signal on some numbers. For example,
1-800-225-2525 is KLM airline's reservation line. It should answer right
away. But * can't detect it is answered and keeps ringing the ip phone. I
put a monitor on the channel, and get the answer messages in the channels.
So somehow the line is answered but * doesn't know. I don't have a problem
to most numbers. The problem only got my attention after one customer
reported it.

A debug on the pri shows,
Ext: 1  Progress Description: Call is not end-to-end ISDN; further call
progress information may be available inband. (1) ]

So maybe the inband information is not detected by *?

Anyone has the same setup, i.e. PRI to your carrier? Can you please dial the
number 1-800-225-2525 and have 'pri debug'? I'd like to compare the results.
I am not sure if it is * or just my * configuration.

Your help is highly appreciated. I am really stuck here.

Thanks,
Richard


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Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now

[Brian Capouch]

James Gardiner wrote:
Asterisk 2.0 on Windows..  This is all very much a bit of a Joke, but one
does beg to ask.
When will version 2.0 be released???
2.0 is just now really being talked about in earnest.
I think a better question would be when 1.2 is going to be out.
That one has more narrow bounds.
B.
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RE: [Asterisk-Users] *** Asterisk 2.0 Stable release out now

[James Gardiner]

Asterisk 2.0 on Windows..  This is all very much a bit of a Joke, but one
does beg to ask.

When will version 2.0 be released???


James



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Re: [Asterisk-Users] echo paid support

[James Taylor]

Thanks guys,
The echo is there when T1 is connected directly to Asterisk box.
If connected to TNT first then to Asterisk via SIP, then Asterisk echo  
training kicks in (you can hear it 1st second or so) then only a little.

I haven't tried the steps below, but will when I build another box this  
next week.

Thanks,
James
On Fri, 1 Apr 2005 17:09:46 -0700, Damon Estep  
<[EMAIL PROTECTED]> wrote:

Brian,
By the description of James config there is no zaptel in hus box, looks
like the TDM to SIP conversion is happening on the TNT, is it still your
opinion that the steps below will help?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian M. Arlinghaus
Sent: Wednesday, March 23, 2005 2:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; James
Taylor
Subject: Re: [Asterisk-Users] echo paid support
James,
After three months of echo, I finally got mine to go away.
I am using a Dell Optiplex GX280 with a P4 3.2GHz processor and 512MB
RAM
and Asterisk version stable 1.07.
Here's what I came up with after reading many of the posts and much
trial
and error.
1. Make sure that ther is no GUI loaded on the asterisk box.
2. Recompile Zaptel with MMX enabled (I think this applies to Intel
Processors only, but someone may correct me if I'm wrong.)
To enable MMX in zaptel, before you compile zaptel, uncomment the line
in
/usr/src/zaptel/zconfig.h file that says:
/* #define CONFIG_ZAPTEL_MMX */
and change it to:
#define CONFIG_ZAPTEL_MMX
3. Recompile Zaptel with the "Aggressive Suppressor" enabled.  I have
never
read anything about this, but saw it while I was enabling the MMX
support.
>From reading zconfig.h, there are different versions of the echo
canceller,
but the comments say that the "aggressive suppressor" works with
"MARK2"
which is what was enabled by default in stable 1.07.
To enable the MARK2 AGGRESSIVE SUPPRESSOR in zaptel, before you
compile
zaptel, uncomment the line in /usr/src/zaptel/zconfig.h file that
says:
/* #define AGGRESSIVE_SUPPRESSOR */
and change it to:
#define AGGRESSIVE_SUPPRESSOR
4. Recompile Zaptel with the instructions "reordered."  I don't know
what
this does, but it was recommended in these posts for fixing echo.
To "reorder" the instructions in zaptel, before you compile zaptel,
add
the
following in /usr/src/zaptel/Makefile underneath the comment in the
Makefile
talking about all the config settings being in zconfig.h.  From the
looks
of
it, it might only effect Pentium 4s???
CFLAGS+=-march=pentium4
5. Make sure the Zaptel card is not sharing an IRQ with othe hardware.
In
my case, this involved moving my T100P to another slot and disabling
all
USB
ports.  (I login remotely to administer the asterisk box since the
GX280
doesn't have a PS/2 keyboard port.)
6. Make sure that you have a sufficient processor and sufficient RAM.
I
didn't make any additions to my configuration, but I did remove one
256MB
RAM chip that seemed to be bad leaving 256MB.
Hope this helps.  Again, most of this is not from me, but others here
with
much more knowledge.
Brian Arlinghaus
- Original Message -
From: "James Taylor" <[EMAIL PROTECTED]>
To: 
Sent: Thursday, March 17, 2005 10:40 AM
Subject: [Asterisk-Users] echo paid support
> I've got echo problems.
> *** I'm looking for paid support. ***
> I'll accept free support, but don't mind paying if someone really
knows
> what they are doing.
>
>
> I've read the wiki, etc.
> Played with the settings in zapata.conf
> Using V400P
>
> PSTN->_T1->_ASTERISK->_BROADVOICE->_PSTNECHO ON CALLED PHONE
> PSTN->_T1->_ASTERISK->_T1->_PSTNNO ECHO
> VOIP->_ASTERISK->_T1->_PSTN ECHO ON VOIP PHONE G711
>
> I have another trunk group and different T1's that go to a MAX TNT
first:
> PSTN->_T1->_MAX_TNT->_VOIP->_ASTERISK->_VOIP_PHONE  ECHO ON VOIP
PHONE
> g711
>
>
PSTN->_T1->_MAX_TNT->_VOIP_G711->_ASTERISK_IAX_GSM->_ASTERISK_IAX_GSM-
>_VOIP_PHONE_g711
> NO ECHO
> --
> James Taylor
> MetroTel
> 3505 Summerihll Road
> Suite 11
> Texarkana, Texas  75503
> 903-793-1956
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--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas  75503
903-793-1956
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Re: [Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Josh Alberts wrote:
> I'm having trouble getting asterisk to run at startup using Ubuntu. 
> I've checked, and the asterisk dameon is set to run at init 5.  However,
> I'm not seeing anything that says that asterisk has been started during
> the boot process.  Oddly, when I shut the machine down/run init6, it
> says Starting Asterisk PBX.  Odd.  I'm using the default scripts that
> came with asterisk (I installed using synaptic and the debian universe
> repositories).  I've edited /etc/default/asterisk, uncommented the first
> line and changed start asterisk to "yes".  Anybody know what might be
> wrong?

Try starting atserisk at run level 2 or 3 (debian/ubuntu does this a
little differently)


- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
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Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

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=ARpy
-END PGP SIGNATURE-
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[Asterisk-Users] how to tell what ${DIALSTATUS} is being set

[Randy Bush]

i often have nufone problems, e.g.

-- Executing Dial("SIP/konaa0p-4b88", "IAX2/[EMAIL PROTECTED]/14086661234") 
in new stack
-- Called [EMAIL PROTECTED]/14086661234
-- Call accepted by 66.225.202.72 (format ulaw)
-- Format for call is ulaw
-- Hungup 'IAX2/NuFone/5'



  == No one is available to answer at this time
-- Executing Hangup("SIP/konaa0p-4b88", "") in new stack
  == Spawn extension (dial-gateways, 14086661234, 5) exited non-zero on 
'SIP/konaa0p-4b88'
-- Executing Hangup("SIP/konaa0p-4b88", "") in new stack
  == Spawn extension (dial-gateways, h, 1) exited non-zero on 'SIP/konaa0p-4b88'

i would like to detect this (and many other things) in ${DIALSTATUS}
conditions so that i can then GotoIf() them.  the problem is that the
log does not tell me explicitly which ${DIALSTATUS} has been returned,
leaving me guessing.  with BUSY vs CONGESTION this is even more of an
issue.

is it reasonable to ask that the log contain the value being set in
${DIALSTATUS}?

randy

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[Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian

[Josh Alberts]

I'm having trouble getting asterisk to run at startup using Ubuntu. 
I've checked, and the asterisk dameon is set to run at init 5.  However,
I'm not seeing anything that says that asterisk has been started during
the boot process.  Oddly, when I shut the machine down/run init6, it
says Starting Asterisk PBX.  Odd.  I'm using the default scripts that
came with asterisk (I installed using synaptic and the debian universe
repositories).  I've edited /etc/default/asterisk, uncommented the first
line and changed start asterisk to "yes".  Anybody know what might be
wrong?
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[Asterisk-Users] problem detecting answer on pri card

[Richard]

Hi,

I have a digium PRI T1 card connecting to my carrier. However it has
problems to detect the answer signal on some numbers. For example,
1-800-225-2525 is KLM airline's reservation line. It should answer right
away. But * can't detect it is answered and keeps ringing the ip phone. I
put a monitor on the channel, and get the answer messages in the channels.
So somehow the line is answered but * doesn't know. I don't have a problem
to most numbers. The problem only got my attention after one customer
reported it.

A debug on the pri shows,
Ext: 1  Progress Description: Call is not end-to-end ISDN; further call
progress information may be available inband. (1) ]

So maybe the inband information is not detected by *?

Anyone has the same setup, i.e. PRI to your carrier? Can you please dial the
number 1-800-225-2525 and have 'pri debug'? I'd like to compare the results.
I am not sure if it is * or just my * configuration.

Your help is highly appreciated. I am really stuck here.

Thanks,
Richard


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[Asterisk-Users] Outbound calls with xlite and Xpro PocketPC

[Robert Keller]

My issue is dialing out to a local prefix does not always connect. The telco
"were sorry, your call does not come thru..." message is received. If I dial
my cell phone (a 201 prefix vs. a 758 prefix) then my cell phone rings every
time. My clients are Xlite on a Mac and Xpro on a pocketpc.

I have an Asterisk (AMP) server running with an X100P clone card connected
to an analog line. 

What could I be missing?




Robert Andrew Keller
Ferndale School District #502
[EMAIL PROTECTED]
360-383-9228 PH.
360-383-9218 FAX
"Paving the way for tomorrows genius."

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Re: [Asterisk-Users] Re: Are there online forums instead of this email

[Joseph]

Que? You told someone to "grow up"?
YE HAA.. We 'be out har' in rural internet space, mamma
Stepping in cow pats .   Way out past the fancy, hi falutin',
city-slicker, collaborative tools of 2005   ..
YE... HAA.   SMTP MAIL!W.
G  I'm really impresses out har'.  Goooll moses.
woo
Um, yeah, check that. Sorry, this "horse-and-buggy" system gives a lot 
more flexibility and lets tools be built off of it. If you don't like it 
we got that, but please, to repeat you, "Grow up and stop posting to 
this tread."

Thanks,
--Joseph
Tim Bass wrote:
Tom Ivar Helbekkmo.   Grow up and stop posting to this tread.
Nobody cares about your bullying insults.
 

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Re: [Asterisk-Users] Asterisk Voice mail with CCM

[Nathan Alberti]

I'm currently in the process of getting it to work for a CCME install, I 
have it all working except for one thing.. I think it was calling a 
phone from the asterisk server the call transfer back to asterisk would 
fail with an authentication issue and die. I'm pretty sure this issue 
can be resolved I just have not had the time recently wo work on it, I 
can provide more info when I'm back in the office next week.

Nathan Reeves wrote:
Anyone running Cisco Call Manager and using Asterisk for voice mail
services?  Things working well or is the concept a bit of a hassle to
implement?
TIA
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[Asterisk-Users] Passing varibles *out* of macros

[Joe Presto]

Hi, I have a dialplan that has a called party indicate whether they want to
accept a call by pressing "1". I'm using a feature found in CVS head.

It works great, except that if the call is connected, and the called party
hangs up first, the caller goes into voicemail.

I tried to work around this by passing a variable out of the macro, but that
doesn't appear to work.  Here's my dialplan - what's the best way to
accomplish this?

[dg-extensions]
exten => 1,1,Playback(dg-connect-to-sales)
exten => 1,2,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) 
exten => 1,3,Monitor(wav,${CALLFILENAME},m) 
exten => 1,4,SetMusicOnHold,sales
exten => 1,5,Dial(SIP/sales|30|gmM(screen))
exten => 1,6,GotoIf($[${screenresult} = accept ] ?8:7)
exten => 1,7,VoiceMail,su1
exten => 1,8,Wait(0)
exten => 1,107,VoiceMail,su1

[macro-screen]
exten => s,1,Wait(0.2)
exten => s,2,Read(ACCEPT|all-your-base|1)
exten => s,3,GotoIf($[${ACCEPT} = 1 ] ?7:4)  ;5:4
exten => s,4,SetVar(MACRO_RESULT=CONTINUE) ;do not connect call
exten => s,5,SetVar(screenresult=deny)
exten => s,6,Goto(s,8) 
exten => s,7,SetVar(screenresult=accept)  ;connect call
exten => s,8,Wait(0)

Thanks in advance - Joe

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asterisk-users@lists.digium.com

[flavio patria]

Can I use SJphone like a H.323 phone in order to dial and receive call
through Asterisk?Can I consider Asterisk just like a sort of H323
Gateway?

Thanks 4 all!

flx
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Re: [Asterisk-Users] Snom and Multiple calls

[Karl Brose]

Did you enable call waiting?
call_waiting: on

Josh Dady wrote:
I've got an issue on the snoms, and I'm wondering if anyone has some 
recent experience with it; I've contacted the one specific reference I 
found to it in the list archives, and the person in question didn't 
seem to find an answer (and snom doesn't appear to be finished moving 
their offices yet).

On the snom (I've tested this on the 220 and 360), the phone will 
immediately reject any new INVITE that arrives with 486 BUSY HERE if 
there's already a call on the phone opening (i.e., either the phone is 
already ringing or you've dialed a call that hasn't been answered 
yet).  If we were to give one of these phones to our receptionist, 
obviously, that wouldn't be acceptable.  Is there a way to disable 
this behavior?

--
Joshua P. Dady

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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

[Etienne Pretorius]

But what happens when the channel is busy etc?

It does not seem to drop the "Attempt a naitive bridge". I saw this in
the commented out section and thought that it'll work but well it still
hangs at "Attempting a naitive bridge".

[outgoing]  ;Dial
"0" on the phone for external line

exten => _0,1,Dial(Zap/g2/$EXTEN)
exten => _0,2,Goto(_0-${DIALSTATUS},1) 
;Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => _0-ANSWER,1,Goto(_0,102)
exten => _0-.,1,Goto(_0,1)  ;Try
another line

exten => _0,102,Congestion
exten => _0,103,Hangup


Hints?
Kind Regards
Etienne

Technical Support
Kingsley Technologies



Etienne Pretorius wrote:

  
Thank you very much, that sorted out the problem.
  Kind Regards
Etienne
  
  
Steven Critchfield wrote:
  
On Sat, 2005-04-02 at 18:39 +0200, Etienne Pretorius wrote:
  

  Never Mind. oops.

I just needed to play around with some syntax.

Zap/1,2,3,4/$EXTEN

Ps: Is there a better santax because 1-4 doesn't work.



Look at groups in the /etc/asterisk/zaptel.conf

Once you define your groups, you can just 
exten => _0,1,Dial(Zap/g1/$EXTEN)
And asterisk will pick some available channel out of the channels
defined in group 1 to use for dialing out.

  

  
Hi All * users...

Question:
   In extensions.conf - I am awaire that you can use macro's but what 
I am wondering about.. is that can you create a macro to do dynamic 
Zap channel allocation for a out going call?
   I don't want to reserve a channel/port in the TDM400P card for "Out 
break" calls, so i was just wandering if some1 could help me a bit 
over here.

[outgoing]  ;Dial 
"0" on the phone for external line

exten => _0,1,Dial(Zap/4/$EXTEN)   ;<=== statically 
allocated to Zap/4 needs to be dynamic
exten => _0,2,Goto(102)
exten => _0,102,Congestion
exten => _0,103,Hangup

I'll apreciate any help in this regard.

  
  

  
  

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asterisk-users@lists.digium.com

[flavio patria]

 Can I use SJphone like a H.323 phone in order to dial and receive call
through Asterisk?Can I consider Asterisk just like a sort of H323
Gateway?

Thanks 4 all!

flx
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RE: [Asterisk-Users] Starting with Asterisk-SIP

[Damon Estep]

Start here to get it running, plan on burning a couple days playing with it. 
There is no fast way to get comfortable with it other than hands on and 
research on the list and wiki.

http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ruben cuevas 
rumin
Sent: Saturday, April 02, 2005 9:24 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Starting with Asterisk-SIP

Hi all,

I'm a Telecomunication Engeenering student. I have to develop a VoIP
apliccation using SIP protocol. I have to develop the SIP Server, and
the SIP clients.

I think I can use Asterisk for this issue. I have installed it and I
have run it, but I don't know how I have to configure it.

I have read the documentation, but It's so much big and I don't know
what I have to do.

Someone could tell me what configuration files have I to use, and what
have I to put in this files?. If is it posible, I would like someone
send me some simple examples of this files.

It would be wonderful if someone could help me.

Thanks in advance.

Best Regards,

  Rubén.
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[Asterisk-Users] Re: Starting with Asterisk-SIP

[flavio patria]

At the URL http://www.voip-info.org may find some examples.

Gettin' started
First of all you must define a "possible" dialplan that you can
configure in the file extensions.conf. Dialplan may include several
options, just like a simple comunication between two softphone(for
example Sjphone) using SIP through the Asterisk PBX.
After this, you must define setting about the other configuration
files (.conf, like sip.conf.. etc..)related to the dialplan defined..
and so on...

However you must easily find several interesting examples over
Internet if you search them^_^

I am an Electronic Engineer student too ^_^

bye
flx


On Sat, 2 Apr 2005 18:24:17 +0200, ruben cuevas rumin
<[EMAIL PROTECTED]> wrote:
> Hi all,
> 
> I'm a Telecomunication Engeenering student. I have to develop a VoIP
> apliccation using SIP protocol. I have to develop the SIP Server, and
> the SIP clients.
> 
> I think I can use Asterisk for this issue. I have installed it and I
> have run it, but I don't know how I have to configure it.
> 
> I have read the documentation, but It's so much big and I don't know
> what I have to do.
> 
> Someone could tell me what configuration files have I to use, and what
> have I to put in this files?. If is it posible, I would like someone
> send me some simple examples of this files.
> 
> It would be wonderful if someone could help me.
> 
> Thanks in advance.
> 
> Best Regards,
> 
>   Rubén.
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Re: [Asterisk-Users] Re: Are there online forums instead of this email

[Jean-Michel Hiver]

You are the bully. So far the majority wish the email list to
continue and yet you still continue to demand that Digium
convert to a forum.
 

Looks like M. Bass likes to troll about mailing lists, see this post:
http://info.ccone.at/INFO/Mail-Archives/procmail/Feb-2003/msg00230.html
As usual, the pro-censorship moralizing zealots don't apply their own 
rules to themselves. Nothing new here...

Best Regards,
Jean-Michel.
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
---> http://ykoz.net/voip/max <---
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Re: [Asterisk-Users] Preserve g729 registration over reinstall??

[Mike Matthews]

For the record here, I am quoting from an email from Digium on the subject:
 
"You will need to backup /var/lib/asterisk/licenses.  You will also needto backup the codec_g729 and format_g729 in your/usr/lib/asterisk/modules/ directory.   The ethernet cards in yourmachine cannot be changed.  Otherwise you will have to reregister yourcodec.
If required you may reregister your codec.  If you run into any problemsreregistering, we will assist you on with that problem.
Please refer to http://www.digium.com/index.php?menu=asterisk_g729 foradditional instructions."
 
Thanks to Digium Support for the prompt and thorough response.Mike Matthews <[EMAIL PROTECTED]> wrote:

I purchased the g729 codec from Digium.  Every time I reinstall Asterisk (or Linux) I naturally lose the registration. Digium only allows one reinstallation without calling them which is a nusance for both them and me. Is there any way to preserve the registration across a reinstall?  Perhaps by backing up a directory or a file? Any help appreciated.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users"Put down that coffee...coffee is for Closers!"Phone: 918-770-4503Fax: 206-666-1720email: [EMAIL PROTECTED]sip: [EMAIL PROTECTED]___
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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

[Etienne Pretorius]

Thank you very much, that sorted out the problem.
Kind Regards
Etienne


Steven Critchfield wrote:

  On Sat, 2005-04-02 at 18:39 +0200, Etienne Pretorius wrote:
  
  
Never Mind. oops.

I just needed to play around with some syntax.

Zap/1,2,3,4/$EXTEN

Ps: Is there a better santax because 1-4 doesn't work.

  
  
Look at groups in the /etc/asterisk/zaptel.conf

Once you define your groups, you can just 
exten => _0,1,Dial(Zap/g1/$EXTEN)
And asterisk will pick some available channel out of the channels
defined in group 1 to use for dialing out.

  
  

  Hi All * users...

Question:
   In extensions.conf - I am awaire that you can use macro's but what 
I am wondering about.. is that can you create a macro to do dynamic 
Zap channel allocation for a out going call?
   I don't want to reserve a channel/port in the TDM400P card for "Out 
break" calls, so i was just wandering if some1 could help me a bit 
over here.

[outgoing]  ;Dial 
"0" on the phone for external line

exten => _0,1,Dial(Zap/4/$EXTEN)   ;<=== statically 
allocated to Zap/4 needs to be dynamic
exten => _0,2,Goto(102)
exten => _0,102,Congestion
exten => _0,103,Hangup

I'll apreciate any help in this regard.

  

  



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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Etienne Pretorius wrote:
> Nope - I jumped to conclusions.
> 
> It just tries channel 1 the whole time.
> 
> Any ideas any1
> 
> Kind Regards
> Etienne
> 
> 
> 
> Etienne Pretorius wrote:
> 
>> Never Mind. oops.
>>
>> I just needed to play around with some syntax.
>>
>> Zap/1,2,3,4/$EXTEN
>>
>> Ps: Is there a better santax because 1-4 doesn't work.
>>
>> Kind Regards
>> Etienne
>>
>>
>> Etienne Pretorius wrote:
>>
>>> Hi All * users...
>>>
>>> Question:
>>>In extensions.conf - I am awaire that you can use macro's but what
>>> I am wondering about.. is that can you create a macro to do dynamic
>>> Zap channel allocation for a out going call?
>>>I don't want to reserve a channel/port in the TDM400P card for
>>> "Out break" calls, so i was just wandering if some1 could help me a
>>> bit over here.
>>>
>>> [outgoing]  ;Dial
>>> "0" on the phone for external line
>>>
>>> exten => _0,1,Dial(Zap/4/$EXTEN)   ;<=== statically
>>> allocated to Zap/4 needs to be dynamic
>>> exten => _0,2,Goto(102)
>>> exten => _0,102,Congestion
>>> exten => _0,103,Hangup
>>>
>>> I'll apreciate any help in this regard.
>>>

Allocate the channels to a group in zaptel.conf (group=1)

then dial with Zap/g1 which will take the lowets available channel.

- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
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Re: [Asterisk-Users] Starting with Asterisk-SIP

[Jeff Heath]

This may not work for you on a student budget unless you're willing to
cut into your beer budget ... :-)

I recently got the book and cd from Signate and got a system up pretty
quickly that can make calls back and forth between a couple of SIP
phones directly attached to the server.  

take a look at www.signate.com

Jeff Heath


On Sat, 2005-04-02 at 11:24, ruben cuevas rumin wrote:
> Hi all,
> 
> I'm a Telecomunication Engeenering student. I have to develop a VoIP
> apliccation using SIP protocol. I have to develop the SIP Server, and
> the SIP clients.
> 
> I think I can use Asterisk for this issue. I have installed it and I
> have run it, but I don't know how I have to configure it.
> 
> I have read the documentation, but It's so much big and I don't know
> what I have to do.
> 
> Someone could tell me what configuration files have I to use, and what
> have I to put in this files?. If is it posible, I would like someone
> send me some simple examples of this files.
> 
> It would be wonderful if someone could help me.
> 
> Thanks in advance.
> 
> Best Regards,
> 
>   Rubén.
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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

[Steven Critchfield]

On Sat, 2005-04-02 at 18:39 +0200, Etienne Pretorius wrote:
> Never Mind. oops.
> 
> I just needed to play around with some syntax.
> 
> Zap/1,2,3,4/$EXTEN
> 
> Ps: Is there a better santax because 1-4 doesn't work.

Look at groups in the /etc/asterisk/zaptel.conf

Once you define your groups, you can just 
exten => _0,1,Dial(Zap/g1/$EXTEN)
And asterisk will pick some available channel out of the channels
defined in group 1 to use for dialing out.

> > Hi All * users...
> >
> > Question:
> >In extensions.conf - I am awaire that you can use macro's but what 
> > I am wondering about.. is that can you create a macro to do dynamic 
> > Zap channel allocation for a out going call?
> >I don't want to reserve a channel/port in the TDM400P card for "Out 
> > break" calls, so i was just wandering if some1 could help me a bit 
> > over here.
> >
> > [outgoing]  ;Dial 
> > "0" on the phone for external line
> >
> > exten => _0,1,Dial(Zap/4/$EXTEN)   ;<=== statically 
> > allocated to Zap/4 needs to be dynamic
> > exten => _0,2,Goto(102)
> > exten => _0,102,Congestion
> > exten => _0,103,Hangup
> >
> > I'll apreciate any help in this regard.
> >
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

[Etienne Pretorius]

Nope - I jumped to conclusions.
It just tries channel 1 the whole time.
Any ideas any1
Kind Regards
Etienne

Etienne Pretorius wrote:
Never Mind. oops.
I just needed to play around with some syntax.
Zap/1,2,3,4/$EXTEN
Ps: Is there a better santax because 1-4 doesn't work.
Kind Regards
Etienne
Etienne Pretorius wrote:
Hi All * users...
Question:
   In extensions.conf - I am awaire that you can use macro's but what 
I am wondering about.. is that can you create a macro to do dynamic 
Zap channel allocation for a out going call?
   I don't want to reserve a channel/port in the TDM400P card for 
"Out break" calls, so i was just wandering if some1 could help me a 
bit over here.

[outgoing]  ;Dial 
"0" on the phone for external line

exten => _0,1,Dial(Zap/4/$EXTEN)   ;<=== statically 
allocated to Zap/4 needs to be dynamic
exten => _0,2,Goto(102)
exten => _0,102,Congestion
exten => _0,103,Hangup

I'll apreciate any help in this regard.
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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

[Etienne Pretorius]

Never Mind. oops.
I just needed to play around with some syntax.
Zap/1,2,3,4/$EXTEN
Ps: Is there a better santax because 1-4 doesn't work.
Kind Regards
Etienne
Etienne Pretorius wrote:
Hi All * users...
Question:
   In extensions.conf - I am awaire that you can use macro's but what 
I am wondering about.. is that can you create a macro to do dynamic 
Zap channel allocation for a out going call?
   I don't want to reserve a channel/port in the TDM400P card for "Out 
break" calls, so i was just wandering if some1 could help me a bit 
over here.

[outgoing]  ;Dial 
"0" on the phone for external line

exten => _0,1,Dial(Zap/4/$EXTEN)   ;<=== statically 
allocated to Zap/4 needs to be dynamic
exten => _0,2,Goto(102)
exten => _0,102,Congestion
exten => _0,103,Hangup

I'll apreciate any help in this regard.
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RE: [Asterisk-Users] Starting with Asterisk-SIP

[Kerry Garrison]

If you ar really lost I would suggest getting [EMAIL PROTECTED] and getting your
system up and running fairly easily. Then you can look at the config files
in case you want to do it yourself later.

Kerry Garrison
http://www.geekgazette.com
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ruben cuevas
rumin
Sent: Saturday, April 02, 2005 8:24 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Starting with Asterisk-SIP

Hi all,

I'm a Telecomunication Engeenering student. I have to develop a VoIP
apliccation using SIP protocol. I have to develop the SIP Server, and the
SIP clients.

I think I can use Asterisk for this issue. I have installed it and I have
run it, but I don't know how I have to configure it.

I have read the documentation, but It's so much big and I don't know what I
have to do.

Someone could tell me what configuration files have I to use, and what have
I to put in this files?. If is it posible, I would like someone send me some
simple examples of this files.

It would be wonderful if someone could help me.

Thanks in advance.

Best Regards,

  Rubén.
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Re: [Asterisk-Users] Starting with Asterisk-SIP

[Jean-Michel Hiver]

I think I can use Asterisk for this issue. I have installed it and I
have run it, but I don't know how I have to configure it.
 

You should go and read some docs:
http://www.digium.com/handbook-draft.pdf
I have read the documentation, but It's so much big and I don't know
what I have to do.
 

The handbook isn't that big. You should be fine.
Someone could tell me what configuration files have I to use, and what
have I to put in this files?. If is it posible, I would like someone
send me some simple examples of this files.
 

Well I can't tell you what to do but I can tell you the following:
- Entries sip.conf, iax.conf have nothing to do with contexts (they 
refer to them though, through context => "blah" directives
)
- Entries in extensions.conf are all contexts.

This was the only thing I had to realize before I could use the software.
It would be wonderful if someone could help me.
 

Unfortunately, I can't do for you the research work you're supposed to 
be doing. But do take a look at voip-info.org. There's a lot of good 
stuff there - it's a wonderful resource and will help you get going.

Cheers,
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
---> http://ykoz.net/voip/max <---
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Re: [Asterisk-Users] Starting with Asterisk-SIP

[Mike]

On Sat, 2 Apr 2005, ruben cuevas rumin wrote:
Hi all,
I'm a Telecomunication Engeenering student. I have to develop a VoIP
apliccation using SIP protocol. I have to develop the SIP Server, and
the SIP clients.
I think I can use Asterisk for this issue. I have installed it and I
have run it, but I don't know how I have to configure it.
I have read the documentation, but It's so much big and I don't know
what I have to do.
Someone could tell me what configuration files have I to use, and what
have I to put in this files?. If is it posible, I would like someone
send me some simple examples of this files.
It would be wonderful if someone could help me.
Thanks in advance.
Best Regards,
 Rubén.
Do this:
in
/usr/src/asterisk
run make samples
now you will have the sample configs in /etc/asterisk
then take a look at voip-info.org
Michael


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[Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

[Etienne Pretorius]

Hi All * users...
Question:
   In extensions.conf - I am awaire that you can use macro's but what I 
am wondering about.. is that can you create a macro to do dynamic Zap 
channel allocation for a out going call?
   I don't want to reserve a channel/port in the TDM400P card for "Out 
break" calls, so i was just wandering if some1 could help me a bit over 
here.

[outgoing]  ;Dial 
"0" on the phone for external line

exten => _0,1,Dial(Zap/4/$EXTEN)   ;<=== statically 
allocated to Zap/4 needs to be dynamic
exten => _0,2,Goto(102)
exten => _0,102,Congestion
exten => _0,103,Hangup

I'll apreciate any help in this regard.
--
Kind Regards
Etienne

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[Asterisk-Users] Starting with Asterisk-SIP

[ruben cuevas rumin]

Hi all,

I'm a Telecomunication Engeenering student. I have to develop a VoIP
apliccation using SIP protocol. I have to develop the SIP Server, and
the SIP clients.

I think I can use Asterisk for this issue. I have installed it and I
have run it, but I don't know how I have to configure it.

I have read the documentation, but It's so much big and I don't know
what I have to do.

Someone could tell me what configuration files have I to use, and what
have I to put in this files?. If is it posible, I would like someone
send me some simple examples of this files.

It would be wonderful if someone could help me.

Thanks in advance.

Best Regards,

  Rubén.
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Re: [Asterisk-Users] VoIP Provider problems

[Johnathan Corgan]

Rich Adamson wrote:
In other words, as the ttl value is increased and additional icmps
are sent, you might see what you believe is congestion, but you still
don't have any clue as to whether hop #2, #5, or #10 actually was
involved with that congestion.
Sure.  But there is a way around this.
The traceroute-style statistics gathering technique that PingPlotter 
uses tries all the hops at the same time and plots the return rate for 
each one.  So a 10 hop path has 10 packets go out, with individual 
packet's TTL set to expire at each hop.  Done over and over again and 
averaged over many probes, you get a very clear picture.  Packet loss at 
one node affects all the probes to that node and further ones, resulting 
in an increasing loss rate as you go down the path. For example:

Hop Loss
1   0%
2   1%
3   1%
4   5%
5   5%
6   6%
7   15%
8   15%
9   16%
10  16%
It's easy to see there is a big problem between hops 6 and 7 and a 
smaller problem between hops 3 and 4.

With the broadvoice router I was seeing (at first) a jump from 0% to 9% 
at my local ISP, then small increments over the next 10 hops until it 
was at about 14%, then a big jump to 29% at the last hop.

The data has varied cyclically between as high as the above and as low 
as <1% all the way across.  Right this very moment, it is 2% within my 
ISP, still 2% all the way to PNAP, then a jump to 14% at the broadvoice 
ingress router at PNAP.

Again, temper the above with the fact that the packet loss may be 
intentional, and these statistics not representative of real RTP 
traffic, as per my previous message.  But I can predict with high 
accuracy what the caller on the other end of my broadvoice call will say 
about my voice quality based on the last number I see for the broadvoice 
ingress router.

-Johnathan
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[Asterisk-Users] Asterisk Voice mail with CCM

[Nathan Reeves]

Anyone running Cisco Call Manager and using Asterisk for voice mail
services?  Things working well or is the concept a bit of a hassle to
implement?

TIA
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Re: [Asterisk-Users] Two accounts at one provider and a 302 redirect problem

[Rich Adamson]

> I've got a problem with my incoming calls (SIP). First I tried to route
> different providers to different extensions in which ._ matched the call
> and called the internal phones and so on.
> 
> Then I got this Nikotel Account. I managed to get it working. Small hint
> for the people trying Nikotel and having problems with the internal 99er
> numbers. Nikotel redirects with the 302 message to a user called
> user at 63.214.186.6. Asterisk can't redirect unless you define a sip.conf
> section with [63.214.186.6]. See
> http://bugs.digium.com/bug_view_page.php?bug_id=0001974 and read
> marksters comment at the bottom.
> 
> So now it worked.
> 
> Then I tried two Nikotel Accounts on the same asterisk machine. Problem
> is that now every internal 99er call goes to the extension defined in
> [63.214.186.6]. My idea is now to route every incoming call to one
> extension. I tried to recognize the called number int the extension like
> this:
> 
> 9978389389,1,Answer
> 9978389389,2,...ring phone 1
> 
> 9948389390,1,Answer
> 9948389390,2,...ring phone 2
> 
> (these are fake numbers)
> 
> But I did not manage to route the call based on the called numbers. Only
> 
> _.,1,Answer
> _.,2,...ring a phone
> 
> did work.
> 
> Does anyone have a hint? I would appreciate any comments!

Not enough info to guess with any reasonableness.

Assuming you are using a register statement for each account, that
statement should look something like:
 register=myuserid:[EMAIL PROTECTED]/1234

The "1234" at the end of that statement tells your provider what
digits to dial when contacting your asterisk. So, in this example,
an entry in extensions.conf like:
 exten => 1234,1,Dial(SIP/3000,15,r)
would cause that incoming call to ring sip phone x3000. For 

If you use a different suffix on each provider's register statement,
you should be able to put together an associated extensions.conf
entry to handle each separately.

On the other hand, if you use a register statement like:
 register=myuserid:[EMAIL PROTECTED]
without the suffix, then an extensions.conf entry like:
 exten => s,1,Dial(SIP/3000,15,r)
would work, but it doesn't distinguish between multiple providers.

If you need a better answer, then post your register statements in
sip.conf (change the passwords) for each provider, along with the
appropriate sections of extensions.conf that handle the incoming
calls.


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Re: [Asterisk-Users] Are there online forums instead of this email

[Jean-Michel Hiver]

YE HAA.. We 'be out har' in rural internet space, mamma
Stepping in cow pats .   Way out past the fancy, hi falutin',
city-slicker, collaborative tools of 2005   ..
 

You babble all this nonsense, and *YOU ARE* the guy advocating moderation?
BWAHAHAHAHAHAHAHAHAAA!
Now please go back to your "superior" women-and-students web forum world 
and leave us in peace.

--
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Essayez gratuitement - 5 crédits offerts.
---> http://ykoz.net/voip/max <---
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[Asterisk-Users] Two accounts at one provider and a 302 redirect problem

[Christian Peter]

Hi,

I've got a problem with my incoming calls (SIP). First I tried to route
different providers to different extensions in which ._ matched the call
and called the internal phones and so on.

Then I got this Nikotel Account. I managed to get it working. Small hint
for the people trying Nikotel and having problems with the internal 99er
numbers. Nikotel redirects with the 302 message to a user called
user at 63.214.186.6. Asterisk can't redirect unless you define a sip.conf
section with [63.214.186.6]. See
http://bugs.digium.com/bug_view_page.php?bug_id=0001974 and read
marksters comment at the bottom.

So now it worked.

Then I tried two Nikotel Accounts on the same asterisk machine. Problem
is that now every internal 99er call goes to the extension defined in
[63.214.186.6]. My idea is now to route every incoming call to one
extension. I tried to recognize the called number int the extension like
this:

9978389389,1,Answer
9978389389,2,...ring phone 1

9948389390,1,Answer
9948389390,2,...ring phone 2

(these are fake numbers)

But I did not manage to route the call based on the called numbers. Only

_.,1,Answer
_.,2,...ring a phone

did work.

Does anyone have a hint? I would appreciate any comments!
Thanks in advance

Christian Peter

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