Re: [Asterisk-Users] Hardware based DSP

[Miroslav Nachev]

   Dear Shahed,

   We are in process to done hardware DSP device for coding of G.729,
G.723, GSM and other speech codecs. We will support the drivers for
Asterisk, SER and OpenH323. The device will be available at middle of
next year. There are 6 variants of the device:
   - USB with 16, 32 and 64 simultaneous channels;
   - PCI with 64, 128 and 256 simultaneous channels.
   In the future the device will support Audio and Video Codecs and
Windows OS.
   

   Best Regards,
   Miroslav Nachev

S> Hi All,

S> Is it correct to say that by design,  asterisk wont make use of any cards
S> hardware dsp capabilities ?

S> I don't think that any of the hardware cards currently supported
S> have any dsp capabilities, but I wanted to know if for example,
S> in the future a driver was written for a card that did have dsp 
S> capabilities,
S> would asterisk be able to make any use of it ?

S> I am only just starting out with asterisk, and have not fully understood
S> the architecture yet, but it seems that in order to handle VoIP and PSTN
S> seamlessly, all dsp related functionality has to be handled by software 

S> Thanks
S> Shahed

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RE: [Asterisk-Users] Calculating required bandwidth

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> [EMAIL PROTECTED] wrote:
> 
>> [EMAIL PROTECTED] wrote:
>>> I was posed this question:
>>> 
>>> A T1 set up for voice carries 24 conversations on a circuit that is
>>> 1.544 megabits/second. Right?
>> 
>> Yes and no. If the T1 is channelized, then yes. If it's a PRI
>> circuit, then it has only 23 channels to carry voice, as the 24th
>> channel is used for the D-channel (signalling channel).
> 
> Only if you're in the US. We have 30 + 1 :-)

Nope. I'm in Canada.

And what you are referring to is an E1, not a T1.


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[Asterisk-Users] Calls arent handled by asterisk - destruction of call

[Göran Törnqvist]

Hello, I’m trying to get started with
asterisk/SIP so I was trying the demo that is provided in the extensions config
file, but the call isn’t “answered” by my server when I try
calling the number that I registered at my SIP provider.

I’ve registered with register =>
John.Doe:MyPass:[EMAIL PROTECTED]/1000
in sip.conf and if I use “sip debug” I can see the call is coming
in but then nothing more happens (see debug output below).

 

Also get these error
messages:

Scheduling destruction of
call '[EMAIL PROTECTED]' in 15000 ms


WARNING[4863]: chan_sip.c:706 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Non-critical Request)

 

Sip.conf:

[general]

context=demo

 

[my-sip-provider]

type=peer

fromuser=MyUser

secret=MyPass

fromdomain=my-sip-provider

context=demo

 

extensions.conf:

[demo]

;

; All the stuff in the demo…

;

exten =>
s,1,Wait,1
; Wait a second, just for fun

exten =>
s,n,Answer
; Answer the line

exten =>
s,n,DigitTimeout,5
; Set Digit Timeout to 5 seconds

exten =>
s,n,ResponseTimeout,10 ; Set
Response Timeout to 10 seconds

…and so on…

 

That’s all I have…have I missed
something?

 

Debug output from call:

 

192.1.1.1=my server

0123456789=my number at SIP-provider

99=the number I’m calling from

213.132.103.213, 212.112.162.50=my
SIP providers IPs

==

Sip read:

INVITE sip:[EMAIL PROTECTED] SIP/2.0

Record-Route: 

Via: SIP/2.0/UDP
213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa69cda50ac3600b

Record-Route:


Via: SIP/2.0/UDP 212.112.162.50;branch=z9hG4bK4885.ddcc862.0

Via: SIP/2.0/UDP  212.112.162.22:5060

From:
;tag=2EBE3E60-1646

To: 

Date: Wed, 15 Dec 2004 10:10:11 GMT

Call-ID:
[EMAIL PROTECTED]

Supported: timer,100rel

Min-SE: 1800

Cisco-Guid:
1458717796-1303908825-2510524757-306778262

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFO

CSeq: 101 INVITE

Max-Forwards: 9

Remote-Party-ID: ;party=calling;screen=yes;privacy=off

Timestamp: 1103105411

Contact: 

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 288

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 1486 6130 IN IP4 212.112.162.22

s=SIP Call

c=IN IP4 212.112.162.22

t=0 0

m=audio 16842 RTP/AVP 18 0 101

c=IN IP4 212.112.162.22

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

 

24 headers, 12 lines

Using latest request as basis request

Sending to 213.132.103.213 : 5060 (non-NAT)

Found RTP audio format 18

Found RTP audio format 0

Found RTP audio format 101

Peer audio RTP is at port 212.112.162.22:16842

Found description format G729

Found description format PCMU

Found description format telephone-event

Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer
- audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)

Non-codec capabilities: us - 0x1 (g723), peer - 0x1
(g723), combined - 0x1 (g723)

Found peer 'wx3.se'

Reliably Transmitting (no NAT):

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP
213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa69cda50ac3600b

Via: SIP/2.0/UDP 212.112.162.50;branch=z9hG4bK4885.ddcc862.0

Via: SIP/2.0/UDP  212.112.162.22:5060

From:
;tag=2EBE3E60-1646

To:
;tag=as3c0db481

Call-ID: [EMAIL PROTECTED]

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: 

Proxy-Authenticate: Digest
realm="asterisk", nonce="59e60c89"

Content-Length: 0

 

 

 to 213.132.103.213:5060

Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000 ms

 

Sip read:

ACK sip:[EMAIL PROTECTED] SIP/2.0

User-Agent: sapphire/1.6.2.0253

Max-Forwards: 70

Via: SIP/2.0/UDP
213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa69cda50ac3600b

To: ;tag=as3c0db481

From:
;tag=2EBE3E60-1646

Call-ID:
[EMAIL PROTECTED]

CSeq: 101 ACK

Content-Length: 0

 

 

9 headers, 0 lines

 






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RE: [Asterisk-Users] Calculating required bandwidth

[Andreas Sikkema]

[EMAIL PROTECTED] wrote:

> [EMAIL PROTECTED] wrote:
>> I was posed this question:
>> 
>> A T1 set up for voice carries 24 conversations on a circuit that is
>> 1.544 megabits/second. Right?
> 
> Yes and no. If the T1 is channelized, then yes. If it's a PRI
> circuit, then it has only 23 channels to carry voice, as the 24th
> channel is used for the D-channel (signalling channel).

Only if you're in the US. We have 30 + 1 :-)

-- 
Andreas SikkemaRits tele.com
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10 2245544f: +31 (0)10 2245540
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RE: [Asterisk-Users] How "expensive"are thedifferent codecs?(Regarding CPU time)

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> Jim Van Meggelen schrieb:
> 
>> You understand. I was kidding a little bit, but yes, I am also
>> wondering just what things can be done to get a slower machine to
>> work as well as possible.
> 
> Okay. So lets try.

What are you running in terms of a kernel or distro?

 Have you tried running Asterisk at pseudo-realtime priority?
 (asterisk -p)
>>> 
>>> That helps in one way: At the moment my system is doing its morning
>>> routine. That means it makes a tar archieve of my /home directory to
>>> my backup drive. With the -p option the "show translation"-values
>>> are equal to the values when my system is idle.
>>> 
>> 
>> And otherwise not?
> 
> Otherwise - without the -p option - the system had values of 400ms
> (and higher) converting speex when it wasn't idle. Now the value
> is constantly at about 210.

Nice. The system is now giving Asterisk the priority it needs. Don't
forget to change that in your rc.local, or wherever you're starting
Asterisk from.

>>> I guess this option could help me a lot regarding the sound
>>> problems I got sometimes.
>> 
>> Yes, it might help a lot.
> 
> I will see when doing some calls over sip (there I had the most
> problems). Maybe at the evening. Now I have to breakfast, shower and
> go to work.

And I need to go to bed!

Cheers,

Jim.


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Re: [Asterisk-Users] How to generate a SIP NOTIFY for Cisco 7960 remote reboot?

[Olle E. Johansson]

Mick Hastings wrote:
Hi Folks,
cheers for all the great info on the list.
I need to create a SIP NOTIFY message to reboot my Cisco 7960 phones but I 
dont know how.
The admin guide gives an example of the packet (attached), I have tried a 
few web searches and found some cool
little programs that generate SIP packets but none that can do NOTIFY and 
none that I could easliy script. (I want a KISS solution)
Any suggestions would be appreciated, thanks.

Check
http://bugs.digium.com/bug_view_page.php?bug_id=0002266
...and build from that patch!
/O
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Re: [Asterisk-Users] My Boss wants background music!!!!

[Wilson Pickett]

> I am searching for a new PBX for the company. My choice is Astrisk. My Boss
> wants background music via all the telephones. This is done in a
> conventional PBX that he wants, but I can use the Asterisk PBX if it can do

What a waste of resources though, like installing video games on the
asterisk server... Ther must be a powerline intercom that would handle
this (adding a speaker per music distribution point.)
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Re: [Asterisk-Users] VoIP Termination

[Wilson Pickett]

> I'm looking to change from a standard telephone line to a VoIP phone line at
> home.  I'm looking for recommendations for VoIP providers that I can use with
> Asterisk.

Don't forget about emergency services (lack of) with voIP. 
 
> One of the catches is that I often telecommute and sometimes I do some side
> business; these practices violate many provider's acceptable use policies.
> So, I need a provider who doesn't care how I use the phone, and one that
> works well with Asterisk.

I agree with those who have said you should use metered services, but
you need to measure the time you spend on the phone to see.
Fortunately, asterisk will do that for you to the second, looking at
the cdr records and totalling up the duration column for a specific
period will tell you what your bill would be at the few cents a minute
you'll be charged.

For providers I also favor IAX (most here who have the choice will)
such as nufone, voicepulse and voipjet. I've also used ICH,
calgarytelecom.
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Re: [Asterisk-Users] Easy question? Get started with the Demo

[Wilson Pickett]

> that is provided in the extensions config file, but the call isn't
> "answered" by my server when I try calling the number that I registered at
> my SIP provider. 

What version of asterisk? Have you tried replacing the 'n' syntax with
2,3,4... in the extension? What does the CLI say when calling?
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Re: [Asterisk-Users] Shorten the recognition time of rings on Wildcard X100P

[Michael Vogel]

Hi!
Adi Linden schrieb:
I connected my Wildcard X100P to the PSTN and created a context in
extensions.conf which rings a number of SIP phones on inbound calls from
the PSTN. When I compare the actual PSTN rings with Asterisk recognition
of the incoming call, Asterisk rings my SIP phones on the third ring of
the incoming call.
Add "usecallerid=no" to your zapata.conf. The caller id is detected 
between the first and second ring. If you don't detect it the system can 
put the call through immediately.

Bye!
Michael
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Re: [Asterisk-Users] How "expensive" are thedifferent codecs?(Regarding CPU time)

[Michael Vogel]

Jim Van Meggelen schrieb:
You understand. I was kidding a little bit, but yes, I am also
wondering just what things can be done to get a slower machine to
work as well as possible.
Okay. So lets try.
Have you tried running Asterisk at pseudo-realtime priority? 
(asterisk -p)
That helps in one way: At the moment my system is doing its morning
 routine. That means it makes a tar archieve of my /home directory
to my backup drive. With the -p option the "show 
translation"-values are equal to the values when my system is idle.

And otherwise not?
Otherwise - without the -p option - the system had values of 400ms (and
higher) converting speex when it wasn't idle. Now the value is
constantly at about 210.
I guess this option could help me a lot regarding the sound 
problems I got sometimes.
Yes, it might help a lot.
I will see when doing some calls over sip (there I had the most 
problems). Maybe at the evening. Now I have to breakfast, shower and go 
to work.

Bye!
Michael
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[Asterisk-Users] Shorten the recognition time of rings on Wildcard X100P

[Adi Linden]

Hi,

I connected my Wildcard X100P to the PSTN and created a context in
extensions.conf which rings a number of SIP phones on inbound calls from
the PSTN. When I compare the actual PSTN rings with Asterisk recognition
of the incoming call, Asterisk rings my SIP phones on the third ring of
the incoming call.

Here is some log info:

On the first ring:
-- Starting simple switch on 'Zap/1-1'

After a second ring:
Dec 17 00:06:05 NOTICE[21609]: chan_zap.c:5363 ss_thread: Got event 2
(Ring/Answered)...
-- Executing Dial("Zap/1-1", "SIP/201&SIP/211||tr") in new stack

The third ring finally rings my SIP phones
-- Called 201
-- Called 211
-- SIP/201-98cb is ringing
-- SIP/211-11fb is ringing

Thanks,
Adi
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[Asterisk-Users] Problems with app_realtime

[Brian Wilkins]

It seems that when setting qualify = 200 or qualify = yes in the database for 
a sip friend/peer, RealTime does not update the registration status like it 
should. 

I also have several peers which have been offline and Asterisk still reports 
them as registered, even though the registration seconds are only 200. 


Asterisk Ver: CVS HEAD 12/1/2004

Layout of sip_buddies: 


mysql> describe sip_buddies;
++---+--+-+-++
| Field  | Type  | Null | Key | Default | Extra  |
++---+--+-+-++
| uniqueid   | int(11)   |  | PRI | NULL| auto_increment |
| name   | varchar(30)   |  | UNI | ||
| accountcode| varchar(30)   | YES  | | NULL||
| amaflags   | char(1)   | YES  | | NULL||
| callgroup  | varchar(30)   | YES  | | NULL||
| callerid   | varchar(50)   | YES  | | NULL||
| canreinvite| char(1)   | YES  | | NULL||
| context| varchar(30)   | YES  | | NULL||
| defaultip  | varchar(15)   | YES  | | NULL||
| dtmfmode   | varchar(7)| YES  | | NULL||
| fromuser   | varchar(50)   | YES  | | NULL||
| fromdomain | varchar(31)   | YES  | | NULL||
| host   | varchar(31)   |  | | ||
| incominglimit  | char(2)   | YES  | | NULL||
| outgoinglimit  | char(2)   | YES  | | NULL||
| insecure   | char(1)   | YES  | | NULL||
| language   | char(2)   | YES  | | NULL||
| mailbox| varchar(50)   | YES  | | NULL||
| md5secret  | varchar(32)   | YES  | | NULL||
| nat| varchar(5)| YES  | | NULL||
| permit | varchar(95)   | YES  | | NULL||
| deny   | varchar(95)   | YES  | | NULL||
| pickupgroup| varchar(10)   | YES  | | NULL||
| port   | varchar(5)|  | | ||
| qualify| varchar(4)| YES  | | NULL||
| restrictcid| char(1)   | YES  | | NULL||
| rtptimeout | char(3)   | YES  | | NULL||
| rtpholdtimeout | char(3)   | YES  | | NULL||
| secret | varchar(30)   | YES  | | NULL||
| type   | varchar(6)|  | | ||
| username   | varchar(30)   |  | | ||
| allow  | varchar(100)  | YES  | | NULL||
| disallow   | varchar(100)  | YES  | | NULL||
| regseconds | int(11)   |  | | 0   ||
| ipaddr | varchar(15)   |  | | ||
| ts | timestamp(14) | YES  | | NULL||
++---+--+-+-++
36 rows in set (0.01 sec)





-- 
Brian Wilkins
Software Engineer
[EMAIL PROTECTED]

Heritage Communications Corporation
  Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net

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[Asterisk-Users] Dialing asterisk from open phone

[amna saleem]

hi !

I have compiled asterisk-oh323 successfully, is there any way that the
open phone can register with asterisk??? i need my open phone to dial
to asterisk ..but it gives the message

"no phone running on IP $192.168.19.206".

I want to call asterisk from open phone and then direct calls to iax
phones  or other IP addresses  using asterisk.

Can anyone help me in this regard

Thanks
Amna Saleem
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[Asterisk-Users] How to generate a SIP NOTIFY for Cisco 7960 remote reboot?

[Mick Hastings]

Hi Folks,

cheers for all the great info on the list.
I need to create a SIP NOTIFY message to reboot my Cisco 7960 phones but I 
dont know how.
The admin guide gives an example of the packet (attached), I have tried a 
few web searches and found some cool
little programs that generate SIP packets but none that can do NOTIFY and 
none that I could easliy script. (I want a KISS solution)
Any suggestions would be appreciated, thanks.

Sample NOTIFY Message
NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP ipaddress:5060;branch=1
Via: SIP/2.0/UDP ipaddress
From: 
To: 
Event: check-sync <<>> Event header.
Date: Mon, 10 Jul 2000 16:28:53 -0700
Call-ID: [EMAIL PROTECTED]
CSeq: 1300 NOTIFY
Contact: 
Content-Length: 0


Also, im still in the testing stages but our end solution will look like 
this:
20 x Cisco 7960
Asterisk
Cisco 2611XM w/ T1 PRI (8 active timeslots, INS1500(JAP))
If anyone can foresee any problems with this setup Id like to know. 



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RE: [Asterisk-Users] How "expensive" are thedifferent codecs?(Regarding CPU time)

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> Jim Van Meggelen schrieb:
> 
>> Hmmm. I propose that we make your system the list guinea pig!  If we
>> can get that one tweaked, there's no telling what alse we can do with
>> Asterisk!
> 
> I'm not really sure that I completely get the meaning of this
> sentence. (Which can be because of the fact that its 06:47 AM or that
> I'm no native speaker or both ;-)) But I guess you meant that we
> could try to
> test everything that is known to work - including some voodo - and to
> see if it works? ;-) 

You understand. I was kidding a little bit, but yes, I am also wondering
just what things can be done to get a slower machine to work as well as
possible.

>> Have you tried running Asterisk at pseudo-realtime priority?
>> (asterisk -p)
> 
> That helps in one way: At the moment my system is doing its morning
> routine. That means it makes a tar archieve of my /home
> directory to my
> backup drive. With the -p option the "show
> translation"-values are equal
> to the values when my system is idle.

And otherwise not?

> I guess this option could help me a lot regarding the sound
> problems I
> got sometimes.

Yes, it might help a lot.

Cheers,

Jim.

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[Asterisk-Users] How to debug? - SIP calls not coming thru

[Göran Törnqvist]

Hello, I’ve just set up SIP with asterisk using
this how-to: http://www.automated.it/guidetoasterisk.htm#_Toc49248757
but when I try calling the number at my SIP provider (Wx3) it doesn’t
come thru. I THINK I registered to my SIP provider without any problem, in
sip.conf I do:

register => My.name:passwd:[EMAIL PROTECTED]/12345678

If I change the password to something else I get an
registration error when doing asterisk / reload so I guess the registration
went ok the first time then when there’s no error message? I also added 2
phones to asterisk: Cisco IP-phone 7960. I GUESS it was registered successfully
because I got errors at first because username was wrong and when I changed it
to correct values – no errors.

“sip show peers” is showing my phones:

 

Name/username   
Host    Dyn Nat
ACL
Mask
Port Status

cisco2/cisco2    213.1.1.1 
D  255.255.255.255 
5060 Unmonitored

cisco1/cisco1    213.1.1.1 
D  255.255.255.255 
5060 Unmonitored

wx3.se   
   213.1.1.1
255.255.255.255  5060 Unmonitored

3 sip peers loaded [3 online , 0 offline]

 

Though host for wx3.se is showing the wrong IP above.

 

How can I debug this?

 

Below is what I’ve added to my config-files.

 

Sip.conf

 

[general]

context=mycontext

 

register => My.name:passwd:[EMAIL PROTECTED]/X

 

 

[wx3.se]

type=peer

fromuser=username-here

secret=pass

fromdomain=wx3.se

 

[cisco1]

type=friend

host=dynamic

defaultip=192.111.111.111

username=cisco1

secret=mypass

dtmfmode=rfc2833 ; Choices are inband, rfc2833, or
info

;mailbox=1000 ; Mailbox for message waiting indicator

context=mycontext

callerid="MyUser 1" <9053>

 

[cisco2]

(almost identical)

 

in extensions.conf:

 

[mycontext]

exten => 1,1,Dial(SIP/cisco1,20,tr)

exten => 2,1,Dial(SIP/cisco2,20,tr)

exten =>
12345678,1,Dial(SIP/cisco1&SIP/cisco2,20,tr

 






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Re: [Asterisk-Users] How "expensive" are the different codecs?(Regarding CPU time)

[Michael Vogel]

Jim Van Meggelen schrieb:
Hmmm. I propose that we make your system the list guinea pig!  If we can
get that one tweaked, there's no telling what alse we can do with
Asterisk!
I'm not really sure that I completely get the meaning of this sentence. 
(Which can be because of the fact that its 06:47 AM or that I'm no 
native speaker or both ;-)) But I guess you meant that we could try to 
test everything that is known to work - including some voodo - and to 
see if it works? ;-)

Have you tried running Asterisk at pseudo-realtime priority? (asterisk
-p)
That helps in one way: At the moment my system is doing its morning 
routine. That means it makes a tar archieve of my /home directory to my 
backup drive. With the -p option the "show translation"-values are equal 
to the values when my system is idle.

I guess this option could help me a lot regarding the sound problems I 
got sometimes.

Bye!
Michael
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Re: [Asterisk-Users] Get asterisk out of the RTP stream?

[C F]

On Thu, 16 Dec 2004 14:51:53 -0600, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> Here is the setup:
> 
> Phone A (in NYC) on own bandwidth.
> Phone B (in LA) on own bandwidth.
> Asterisk box in Houston,TX on own bandwidth.
> 
> Both phones contact asterisk to register. Not much bandwidth used for this
> as it is a few packets every hour or so.
> 
> Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk
> calls phone B. Both phones are connected and both people are talking.
> 
> Is all of the data/voice comming from phone A going into asterisk box and
> then from asterisk box to phone B? If so, then using g711, phone A would
> send/recieve 64Kbps to/from asterisk and phone B would also send/recieve
> 64Kbps to/from asterisk. Asterisk would then be sending/recieving 128Kbps
> for this one call right? So with 1 T1 you could only get 12 calls going
> right?
> 
> If I use canreinvite=yes on both phones, will phone A connect to phone B
> directly therefore lowering the bandwidth usage in/out of the asterisk box
> right?
> 
> If so, what is the "signalling" bandwidth usage in/out of asterisk in this
> case? Even if the phones are connected directly to eachother, they still
> have to pass some data to asterisk so asterisk still knows that the call is
> up and has to know when the call goes away. We need to know this bandwidth
> usage on a T1 because lets say it was 10Kbps, you could actually do a bunch
> of calls on 1 T1 provided that all phones use canreinvite right?
> 
> Thanks,
> Matthew
> 
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> 
The answer is yes. If the a reinvite is issued then * is out of it but
stays in there for the signaling.
look at the following:
http://www.voip-info.org/wiki-Asterisk+SIP+media+path
http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%20clients%20connect%20directly
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[Asterisk-Users] Call confirmation on NON Zap channels

[Me]

I would like to setup call confirmation so that the called party has to 
press a key to accept the call. There seems to be an Asterisk feature to do 
this with Zap channels where you place a "c" in the dial string. I want to 
do the same thing without re-inventing the wheel with IAX and SIP channels.

Right now the best I can do is play a sound to the called party, I have also 
figured out how to run a Macro when the called party answers but not sure 
what to dump into the Macro to make the conformation work.

Any help would be appreciated!
Thanks..
--
Start Your Own ISP!
http://www.YourOwnISP.com 

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Re: [Asterisk-Users] Hardware based DSP

[Lubomir Christov]

Hi Shahed,
yes you are right if you use Digium hardware "all dsp related 
functionality has to be handled by software".
Asterisk supports also other hardware and one of them have DSP on it - 
Quicknet Phone/LineJACK cards ... but I'm NOT recommending you to use 
this hardware!
 Quicknet's driver is too old, it haven't been updated recently (I'm 
talking about the original ixj driver) and Quicknet don't offer good 
Linux support.

If you prefer to use hardware based DSP I think that it will be better 
to use not a PC cards, but to find some SIP based VoIP Gateways (they 
have g729 and g723 codecs included) and are working very well with Asterisk.

I'm recommending you this web site
http://www.voip-info.org
it's a very good VoIP information  resource.
Lubo
--
-
Appradius Project: RADIUS authentication and accounting support for 
Asterisk PBX
http://appradius.minitelecom.org/
-

Shahed wrote:
Hi All,
Is it correct to say that by design,  asterisk wont make use of any cards
hardware dsp capabilities ?
I don't think that any of the hardware cards currently supported
have any dsp capabilities, but I wanted to know if for example,
in the future a driver was written for a card that did have dsp 
capabilities,
would asterisk be able to make any use of it ?

I am only just starting out with asterisk, and have not fully understood
the architecture yet, but it seems that in order to handle VoIP and PSTN
seamlessly, all dsp related functionality has to be handled by software 


Thanks
Shahed
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[Asterisk-Users] Public Thanks

[Voip Business]

Hello List.

A public thanks for:

Mark
Digium
Asterisk List

And Litnimax (an asterisk Consultant) with very good result in his
work helping me with an installation, very recomendable service, price
and knowledge.

[EMAIL PROTECTED]

Note: no comission in this ;)

regards

HA
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[Asterisk-Users] Easy question? Get started with the Demo

[Göran Törnqvist]

Hello, I’m trying to get started with
asterisk/SIP so I was trying the demo that is provided in the extensions config
file, but the call isn’t “answered” by my server when I try
calling the number that I registered at my SIP provider.

I’ve registered with register => John.Doe:MyPass:[EMAIL PROTECTED]
in sip.conf and if I use “sip debug” I can see the call is coming
in but then nothing more happens (see debug output below).

 

Sip.conf:

[general]

context=demo

 

[wx3.se]

type=peer

fromuser=MyUser

secret=MyPass

fromdomain=my-sip-provider

context=demo

 

extensions.conf:

[demo]

;

; All the stuff in the demo…

;

exten => s,1,Wait,1 ; Wait a
second, just for fun

exten => s,n,Answer ; Answer
the line

exten => s,n,DigitTimeout,5 ; Set
Digit Timeout to 5 seconds

exten => s,n,ResponseTimeout,10 ; Set
Response Timeout to 10 seconds

…and so on…

 

That’s all I have…have I missed
something?

 

Debug output from call:

 

192.1.1.1=my server

0123456789=my number at SIP-provider

99=the number I’m calling from

213.132.103.213, 212.112.162.50=my
SIP providers IPs

==

Sip read:

INVITE sip:[EMAIL PROTECTED] SIP/2.0

Record-Route:


Via: SIP/2.0/UDP
213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa69cda50ac3600b

Record-Route: 

Via: SIP/2.0/UDP 212.112.162.50;branch=z9hG4bK4885.ddcc862.0

Via: SIP/2.0/UDP  212.112.162.22:5060

From: ;tag=2EBE3E60-1646

To: 

Date: Wed, 15 Dec 2004 10:10:11 GMT

Call-ID:
[EMAIL PROTECTED]

Supported: timer,100rel

Min-SE: 1800

Cisco-Guid:
1458717796-1303908825-2510524757-306778262

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFO

CSeq: 101 INVITE

Max-Forwards: 9

Remote-Party-ID: ;party=calling;screen=yes;privacy=off

Timestamp: 1103105411

Contact: 

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 288

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 1486 6130 IN IP4 212.112.162.22

s=SIP Call

c=IN IP4 212.112.162.22

t=0 0

m=audio 16842 RTP/AVP 18 0 101

c=IN IP4 212.112.162.22

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

 

24 headers, 12 lines

Using latest request as basis request

Sending to 213.132.103.213 : 5060 (non-NAT)

Found RTP audio format 18

Found RTP audio format 0

Found RTP audio format 101

Peer audio RTP is at port 212.112.162.22:16842

Found description format G729

Found description format PCMU

Found description format telephone-event

Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer
- audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)

Non-codec capabilities: us - 0x1 (g723), peer - 0x1
(g723), combined - 0x1 (g723)

Found peer 'wx3.se'

Reliably Transmitting (no NAT):

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa69cda50ac3600b

Via: SIP/2.0/UDP 212.112.162.50;branch=z9hG4bK4885.ddcc862.0

Via: SIP/2.0/UDP  212.112.162.22:5060

From: ;tag=2EBE3E60-1646

To: ;tag=as3c0db481

Call-ID:
[EMAIL PROTECTED]

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: 

Proxy-Authenticate: Digest
realm="asterisk", nonce="59e60c89"

Content-Length: 0

 

 

 to 213.132.103.213:5060

Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000 ms

 

Sip read:

ACK sip:[EMAIL PROTECTED] SIP/2.0

User-Agent: sapphire/1.6.2.0253

Max-Forwards: 70

Via: SIP/2.0/UDP
213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa69cda50ac3600b

To: ;tag=as3c0db481

From: ;tag=2EBE3E60-1646

Call-ID:
[EMAIL PROTECTED]

CSeq: 101 ACK

Content-Length: 0

 

 

9 headers, 0 lines

 






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[Asterisk-Users] 191st simultaneous call fails

[Jim Gottlieb]

I've been testing both T400P and TE405P boards and I'm running into
some kind of hard limit on the number of simultaneous calls.  This is
on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1.

Everything is fine up to 190 channels, but the 191st call fails every
time with errors like:

Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1
Dec 14 15:44:00 WARNING[1215]: Failed to create update thread!
Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on channel 0/9, span 9
Dec 14 15:44:00 WARNING[1215]: Call specified, but not found?
Dec 14 15:44:00 WARNING[1215]: Hangup on bad channel 0/9 on span 9

It's not tied to which channel the call comes in on.  It's some
resource that's exhausted after 190 calls.  A limit on threads?

I thought it might be per-process file descriptors even though we were
only going up to 529 on that PID and I used 'ulimit -n' to increase it
before starting asterisk, but that didn't make a difference.

# cat /proc/sys/kernel/threads-max 
14336

I would think that's enough, but perhaps the per-process limit is much
lower.

Any clues?

Thanks...
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[Asterisk-Users] Call Waiting FXS and *

[Philippe Daoust]

I have a small home setup with one Cisco 7960 SIP phone, one WISIP, one 
FXO connected to Bell Canada PSTN, and four FXS connected to POTS phones 
throughout the house.  I also have an account to a SIP based DID provider.

My problem is when I'm on a call on one of the FXS connected phones and 
receive another call either via the PSTN line (assuming the call I'm on 
is using my SIP account) or SIP account I can hear my other phones ring 
but don't get the call waiting tone on the phone I am (it works great on 
the Cisco though).  I can run to another phone and answer the call or 
let it go to VM but I would really like to be able to pick it up using 
the FXS connected phone.

I did a bit of searching and it doesn't seem like it's possible...  Is 
there a way or is it on the roadmap for an eventual feature?  Is this a 
software of hardware limitation?

This is really important, it's seriously affecting the WAF for this 
project...  ;-)  I'm starting to think I should have gone with SPA's to 
support my POTS phones...  :-(

Thanks!
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Re: [Asterisk-Users] Realtime problem

[Brian Wilkins]

Clay,
Can you post your extconfig.conf and your database schema?

If you want to load your static sip configuration into a database, follow 
these instructions:
   http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20Static

I haven't loaded a static file into the database using RealTime, but used this 
method and it works great: 

http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20Sip

I placed a timestamp column so I can track when the most recent successful 
registration occurred.

--
-- Table structure for table `sip_buddies`
--

CREATE TABLE `sip_buddies` (
  `uniqueid` int(11) NOT NULL auto_increment,
  `name` varchar(30) NOT NULL default '',
  `accountcode` varchar(30) default NULL,
  `amaflags` char(1) default NULL,
  `callgroup` varchar(30) default NULL,
  `callerid` varchar(50) default NULL,
  `canreinvite` char(1) default NULL,
  `context` varchar(30) default NULL,
  `defaultip` varchar(15) default NULL,
  `dtmfmode` varchar(7) default NULL,
  `fromuser` varchar(50) default NULL,
  `fromdomain` varchar(31) default NULL,
  `host` varchar(31) NOT NULL default '',
  `incominglimit` char(2) default NULL,
  `outgoinglimit` char(2) default NULL,
  `insecure` char(1) default NULL,
  `language` char(2) default NULL,
  `mailbox` varchar(50) default NULL,
  `md5secret` varchar(32) default NULL,
  `nat` varchar(5) default NULL,
  `permit` varchar(95) default NULL,
  `deny` varchar(95) default NULL,
  `pickupgroup` varchar(10) default NULL,
  `port` varchar(5) NOT NULL default '',
  `qualify` varchar(4) default NULL,
  `restrictcid` char(1) default NULL,
  `rtptimeout` char(3) default NULL,
  `rtpholdtimeout` char(3) default NULL,
  `secret` varchar(30) default NULL,
  `type` varchar(6) NOT NULL default '',
  `username` varchar(30) NOT NULL default '',
  `allow` varchar(100) default NULL,
  `disallow` varchar(100) default NULL,
  `regseconds` int(11) NOT NULL default '0',;
  `ipaddr` varchar(15) NOT NULL default '',
  `ts` timestamp(14) NOT NULL,
  PRIMARY KEY  (`uniqueid`),
  UNIQUE KEY `name` (`name`),
  KEY `name_2` (`name`)
) TYPE=MyISAM;

-extconfig.conf-
==

; Static and realtime external configuration
; engine configuration
;
; Please read doc/README.extconfig for basic table
; formatting information.
;
[settings]
;
; Static configuration files:
;
; file.conf => driver,database[,table]
;
; maps a particular configuration file to the given
; database driver, database and table (or uses the
; name of the file as the table if not specified)
;
;uncomment to load queues.conf via the odbc engine.
;
;sip.conf => odbc,asterisk,sip

;
; Realtime configuration engine
;
; maps a particular family of realtime
; configuration to a given database driver,
; database and table (or uses the name of
; the family if the table is not specified
;
;iaxfriends => odbc,asterisk
sipfriends => mysql,asterisk,sip_buddies
;voicemail => odbc,asterisk


-res_mysql.conf-
;
; Sample configuration for res_config_mysql.c
;
; The value of dbhost may be either a hostname or an IP address.
; If dbhost is commented out or the string "localhost", a connection
; to the local host is assumed and dbsock is used instead of TCP/IP
; to connect to the server.
;
[general]
;dbhost = 127.0.0.1
dbname = asterisk
dbuser = [removed]
dbpass = [removed]
dbport = 3306
dbsock = /var/run/mysqld/mysqld.sock


On Tuesday 14 December 2004 09:50 pm, Clay Reiche wrote:
> I'm having trouble with the Realtime setup. I've followed the instructions
> on voip-info using odbc but I get this message during asterisk boot:
>
>
>
> Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory)
>
> Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load
> config sip.conf, SIP disabled
>
>   == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
>
>   == Registered application 'SIPDtmfMode'
>
>
>
> And my device(s) won't register. I don't even see them attempt the
> registration...(from the CLI in ery verbose.)
>
>
>
> Maybe I'm not using the right version of asterisk??? Is that possible and
> how would I know? My "show version" gives me this:
>
>
>
> *CLI> show version
>
> Asterisk CVS-v1-0-12/08/04-16:50:05 built by [EMAIL PROTECTED] on a i686
> running Linux
>
> *CLI>
>
>
>
> Any help would be appreciated. Thanks!
>
>
>
> Clay Reiche

-- 
Brian Wilkins
Software Engineer
[EMAIL PROTECTED]

Heritage Communications Corporation
  Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net

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RE: [Asterisk-Users] VoIP Termination

[Jay Milk]

These or very similar policies are found with all providers of
"unlimited" service.  Fact is, unless you're paying for every one of the
43,000 minutes possible each month, they can't guarantee you "unlimited"
service... so they sneak sanity clauses into their contracts to prevent
you from using the phone above and beyond their profit margin.  The only
way to get around this is to get metered services, and those are, in my
experience, quite competitive with "unlimited" plans.  Termination can
be had for 1.3c/minute to anywhere in the US, origination can be found
for between $7.50/unlimited to $1.50/month+0.014/minute.  Do the math
and see how many minutes you *actually* use.  In our case, we found it
easier and *cheaper* to simply get a toll-free number rather than
multiple DIDs to be reached locally.

All that said, I would avoid Packet8 for other reasons, most importantly
the fact that they won't let you terminate into * directly.

> -Original Message-
> From: Mike Diehl (Encrypted email preferred) 
> [mailto:[EMAIL PROTECTED] 
> Sent: Thursday, December 16, 2004 2:54 PM
> To: [EMAIL PROTECTED]
> Cc: Andrew Kohlsmith
> Subject: Re: [Asterisk-Users] VoIP Termination
> 
> 
> On Thursday 16 December 2004 05:17 am, Andrew Kohlsmith wrote:
> > On December 16, 2004 12:03 am, Mike Diehl (Encrypted email 
> preferred) 
> > wrote:
> > > One of the catches is that I often telecommute and sometimes I do 
> > > some side business; these practices violate many provider's 
> > > acceptable use policies. So, I need a provider who 
> doesn't care how 
> > > I use the phone, and one that works well with Asterisk.
> >
> > You've gotta be kidding, VOIP providers are trying to 
> regulate who you 
> > can call?  Go with Nufone or iax.cc or even voicepulse 
> connect -- use 
> > IAX2 over SIP, IMO it's just better.
> 
> Thanx, I will look into these providers.
> 
> This is an exerpt from Packet8's Terms of Use statement.  
> I've edited it for 
> space, but I've tried to retain the context:
> --
> 
> PERSONAL USE. 8x8's Service Plans for residential subscribers 
> that offer 
> unlimited minutes of PSTN calls ("Unlimited PSTN Plans") are for the 
> reasonable personal residential use of End User only. End 
> Users of Unlimited 
> PSTN Plans shall not use the Services for commercial or 
> governmental purposes 
> or for profit or non-profit activities, including, but not 
> limited to, home 
> office, business, sales, tele-commuting, autodialing, 
> continuous or extensive 
> call forwarding, continuous connectivity, fax broadcast, fax 
> blasting, 
> telemarketing or any other activity that would be 
> inconsistent with personal 
> and residential usage. 8x8 reserves the right to immediately 
> terminate or 
> modify the Services of any End User using Unlimited PSTN Plans if 8x8 
> determines, in its sole discretion, that End User is not 
> using the Unlimited 
> PSTN Plans for End User's reasonable personal residential use.
> --
> 
> 
> Now I agree with their policy on fax-blasting, etc.  But 
> according to them, I 
> can't use my own phone for charity work?  I work at a 
> national lab; would my 
> wife be alowed to call me at work?  Or would the be a 
> "governmental purpose?"
> 
> It gets better... If Packet8 decides, in THEIR SOLE 
> DISCRETION, that I'm 
> conducting a business with my phone, they can terminate my 
> service, or 
> increase the price of it.
> 
> I'm trying to make an issue out of this because I think it 
> needs to change and 
> I'm hoping people who are affiliated with these providers are 
> reading this.  
> I was going to go with Packet8.  I was going through the 
> "final checklist" 
> before subscribing when I came accross this fascist policy.
> 
> Sure, I can go with a business plan, but that would cost me 
> $39.95.  That's $5 
> more than I'm spending for an analog phone line!  Part of the 
> reason for me 
> to go with VoIP is to become "Quest Free."  But suddenly, 
> Quest is starting 
> to resemble the Boy Scouts when compared to the types of 
> usage policies I'm 
> seeing from some of the VoIP providers.
> 
> Sorry for the rant, but I hope you understand.
> 
> -- 
> Mike
> gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc
> 83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB
> 
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RE: [Asterisk-Users] MusicOnHold. not getting it.

[Ferguson, Michael]

Mark, Thanks for the pointer. 'preciate it.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Thursday, December 16, 2004 6:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MusicOnHold. not getting it.


This is well documented in the WIKI.

And, it's not configured from the extensions.conf file. Look for
musiconhold.conf in /etc/asterisk. Genreally one doesn't have to mess
with it but you can do all sorts of neat tricks with it. I have our
office one doing different hold music for different departments so that
they can have their own messages etc played when someone is on hold.

If you want to her your hold music ad a line like this to your
extensions.conf

exten => ,1,Musiconhold(default)

Which will play all the hold music until you hangup.

If you want to play the hold music to peeps while your phone is ringing
do this;

exten => ,1,Dial(SIP/|20|m)

which will play the music for 20 seconds whilst ringing the phone.

Mark

On Thu, 2004-12-16 at 16:57, Ferguson, Michael wrote:
>  
>  
> G'Day All;
>  
> I am a little unsure on how to get Music On Hold to work. Please 
> critique my extensions.conf.  ? Thanks
>  
> 
> ; SIP 5001
> 
> exten => 5001,1,Dial(SIP/5001)
> 
> exten => 5001,2,Voicemail(u${EXTEN})
> 
> exten => 5001,3,Hangup
> 
> exten => 5001,102,Voicemail(b${EXTEN})
> 
> exten => 5001,103,Hangup
> 
>  
> 
>  
> 
> Thanks
> 
> 
> 
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-- 

Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
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Re: [Asterisk-Users] Calculating required bandwidth

[Ed Greenberg]

--On Thursday, December 16, 2004 9:45 AM -0800 Ed Greenberg 
<[EMAIL PROTECTED]> wrote:

I was posed this question:
I've learned a ton, in the discussion that followed this question. Thanks, 
all.
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Re: [Asterisk-Users] No Caller ID Name PRI NI2.

[Kevin P. Fleming]

Chris Modesitt wrote:
Your right, after three hours the decided that they both (Qwest and ELI)
were not sending the name via facility.  I don't know how they were sending
it but it is now working.  :)
Most likely they weren't. In my experience, ordering a PRI with NI-2 
signaling is not enough, you have to specifically ask for Calling Name 
delivery, even though they don't charge for it.

I suspect this is because if you don't ask, then they don't have to do 
the SS7 database dips to get the CNAM for your incoming calls, and that 
saves them $$$.
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Re: [Asterisk-Users] Hardware based DSP

[Steve Underwood]

Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
 

I don't think that any of the hardware cards currently
supported have any dsp capabilities, but I wanted to know if
for example, in the future a driver was written for a card that did
have dsp capabilities,
would asterisk be able to make any use of it ?
   

Cards like that already exist. Audiotronix cards have on board DSPs, and
they work with Asterisk (I am not endorsing those cards -- I've never
used one -- just using them as an example of existing,
hardware-based-DSP cards that work with Asterisk).
 

Did you means Voicetronix? I think Audiotronix only do DSP consulting.
Regards,
Steve
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[Asterisk-Users] Compile issues: * 1.02 + FreeBSD 5.3

[Jason Lixfeld]

New 5.3 install.  Saw some stuff around about problems with pwlib but  
those were in 0.9 and have long since been fixed and I haven't found  
anything else out there to explain this stuff.

Anyone have any ideas?
gmake[2]: Entering directory  
`/usr/ports/net/asterisk/work/asterisk-1.0.2/channels/h323'
c++ -O -pipe -c -fno-rtti -o ast_h323.o -O -pipe  -Wall  
-Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations   
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE   -march=i386  
-I/usr/local/include -DZAPTEL_OPTIMIZATIONS  
-DASTERISK_VERSION=\"1.0.2\" -DINSTALL_PREFIX=\"/usr/local\"  
-DASTETCDIR=\"/usr/local/etc/asterisk\"  
-DASTLIBDIR=\"/usr/local/lib/asterisk\"  
-DASTVARLIBDIR=\"/usr/local/share/asterisk\"  
-DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\"  
-DASTLOGDIR=\"/var/log/asterisk\"  
-DASTCONFPATH=\"/usr/local/etc/asterisk/asterisk.conf\"  
-DASTMODDIR=\"/usr/local/lib/asterisk/modules\"  
-DASTAGIDIR=\"/usr/local/share/asterisk/agi-bin\"  
-DBUSYDETECT_MARTIN  -Wno-missing-prototypes  
-Wno-missing-declarations  -DOLD_SANGOMA_API -DIAX_TRUNKING  
-I/usr/local/include  -DCRYPTO -fPIC  -DPBYTE_ORDER=PLITTLE_ENDIAN  
-DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -pthread  
-D_REENTRANT -D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS  
-DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I../../include  
-I/usr/ports/devel/pwlib/work/pwlib/include/ptlib/unix  
-I/usr/ports/devel/pwlib/work/pwlib/include  
-I/usr/ports/net/openh323/work/openh323/include -Wno-missing-prototypes  
-Wno-missing-declarations ast_h323.cpp
cc1plus: warning: command line option "-Wstrict-prototypes" is valid  
for C/ObjC but not for C++
cc1plus: warning: command line option "-Wmissing-prototypes" is valid  
for C/ObjC but not for C++
cc1plus: warning: command line option "-Wmissing-declarations" is valid  
for C/ObjC but not for C++
cc1plus: warning: command line option "-Wno-missing-prototypes" is  
valid for C/ObjC but not for C++
cc1plus: warning: command line option "-Wno-missing-declarations" is  
valid for C/ObjC but not for C++
cc1plus: warning: command line option "-Wno-missing-prototypes" is  
valid for C/ObjC but not for C++
cc1plus: warning: command line option "-Wno-missing-declarations" is  
valid for C/ObjC but not for C++
In file included from  
/usr/ports/devel/pwlib/work/pwlib/include/ptlib/unix/ptlib/../../ 
contain.h:776,
 from  
/usr/ports/devel/pwlib/work/pwlib/include/ptlib/unix/ptlib/contain.h: 
120,
 from  
/usr/ports/devel/pwlib/work/pwlib/include/ptlib.h:139,
 from ast_h323.cpp:34:
/usr/ports/devel/pwlib/work/pwlib/include/ptlib/contain.inl: In  
constructor `PAbstractList::PAbstractList()':
/usr/ports/devel/pwlib/work/pwlib/include/ptlib/contain.inl:419:  
warning: right-hand operand of comma has no effect
ast_h323.cpp: In member function `void  
MyH323Connection::SendUserInputTone(char, unsigned int)':
ast_h323.cpp:725: error: invalid conversion from `char' to `const char*'
ast_h323.cpp: In member function `virtual void  
MyH323Connection::OnUserInputTone(char, unsigned int, unsigned int,  
unsigned int)':
ast_h323.cpp:735: error: invalid conversion from `char' to `const char*'
ast_h323.cpp: In member function `virtual void  
MyH323Connection::OnUserInputString(const PString&)':
ast_h323.cpp:746: error: invalid conversion from `char' to `const char*'
/usr/ports/devel/pwlib/work/pwlib/include/ptlib/unix/ptlib/../../ 
pdirect.h: At global scope:
/usr/ports/devel/pwlib/work/pwlib/include/ptlib/unix/ptlib/../../ 
pdirect.h:458: warning: inline function `static BOOL  
PDirectory::Remove(const PString&)' used but never defined
chan_h323.h:31: warning: 'bindaddr' defined but not used
gmake[2]: *** [ast_h323.o] Error 1
gmake[2]: Leaving directory  
`/usr/ports/net/asterisk/work/asterisk-1.0.2/channels/h323'
gmake[1]: *** [h323/ast_h323.o] Error 2
gmake[1]: Leaving directory  
`/usr/ports/net/asterisk/work/asterisk-1.0.2/channels'
gmake: *** [subdirs] Error 1
*** Error code 2

Stop in /usr/ports/net/asterisk.
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Re: [Asterisk-Users] Rapid DTMF entry failure

[Steve Underwood]

Greg,
How fast is fast? The DTMF detector is tested at the maximum specified 
rate - 10 digits per second - and gives no problems. Some systems do not 
control the rate properly, and allow the gaps between tone pulses to be 
completely eliminated. This contravenes the spec., and does not work. 
The detector is lifted from my spandsp library. You can download and try 
that yourself. It contains a full test suite for the DTMF decoder :-)

Steve
Greg Smith wrote:
 

 

Using inband DTMF signaling, our Asterisk server fails to pass DTMF 
when it is entered rapidly by a caller.For example, when a phone 
number is entered, if it is entered slowly (a brief pause between each 
number) Asterisk is fine.  However, if the ten digits are entered 
quickly, Asterisk drops digits.

 

Thanks.
Greg
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Re: [Asterisk-Users] Bugtracker Karma Hall Of Fame

[Kevin P. Fleming]

Paul Crick wrote:
Maybe we need a karma arbitration committee..
Sorry, silly mood today ;-)
Yeah, let's organize one! Oh wait, this is unimportant... never mind 
But seriously, if you think you're owed karma for something and haven't
received it, flag it to a bug marshall. I'm not one, I just did the web
stuff.
No, it's not that. It's just general inconsistency; I have seen it 
awarded twice for the same bug, awarded for bugs that never got merged, 
etc. I don't make the rules, don't really care about the rules, just 
want to see them consistently applied... But I doubt I'll pay that much 
attention anyway, I just want to make Asterisk better!
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RE: [Asterisk-Users] No Caller ID Name PRI NI2.

[Chris Modesitt]

> You are definitely not getting calling name over facility information
element
>from your telco.  I do not see calling name information anywhere in that
>dump.
>Unless that's not a complete dump, or there's some other problem, I'd talk
>to your telco about it.

>Matthew Fredrickson

Your right, after three hours the decided that they both (Qwest and ELI)
were not sending the name via facility.  I don't know how they were sending
it but it is now working.  :)


Chris.

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Re: [Asterisk-Users] Steps to configure D/41EPCI card

[Steve Underwood]

[EMAIL PROTECTED] wrote:
Hi,
 

Somebody can give me the necessary steps for configuring a D/41EPCI in 
Asterisk.

 

Thanks in advance,
 

Jorge
You can't. It isn't supported.
Regards,
Steve
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Re: [Asterisk-Users] native MOH with Asterisk 1.0.3

[Kevin P. Fleming]

Kristian Kielhofner wrote:
What is the bug ID for moh stop?
It's 3035.
The current native moh patch in Mantis definitely will not apply to 
1.0.x. I believe that anthm removed my posted patch from the bug, so 
it's no longer available there. If you want it I can try to scrounge up 
a patch that will work for you, but I don't run 1.0.x here so I can't 
make any promises.
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Re: [Asterisk-Users] VoIP Termination

[William Suffill]

1 port so easier w/ nat + it can trunk(lowering overhead) for multiple
calls to 1 provider.
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Re: [Asterisk-Users] VoIP Termination

[Andrew Kohlsmith]

On December 16, 2004 08:47 pm, Gary Carr wrote:
> Why is IAX termination better?

Becuase it's lean and mean and doesn't rely on the "too many cooks in the 
kitchen" implementation that is SIP.  IAX2 works seamlessly behind NAT and 
doesn't have any of the weird issues that SIP tends to have (STUN, SER, SIP 
Proxies, etc., etc.).

I'm biased though, I just hate SIP.  :-)

-A.
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Re: [Asterisk-Users] native MOH with Asterisk 1.0.3

[Kevin P. Fleming]

Brian West wrote:
That moh_stop is for the MusicOnHold application ONLY.
How is that true? It's just a safety net to ensure that any moh that 
might be running on the channel is stopped before the channel is 
destroyed. This could be due to the MusicOnHold application, or because 
the call was placed on hold, or because the caller is in a queue waiting 
service.
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Re: [Asterisk-Users] Get asterisk out of the RTP stream?

[Kevin P. Fleming]

Matthew Boehm wrote:
If so, what is the "signalling" bandwidth usage in/out of asterisk in this
case? Even if the phones are connected directly to eachother, they still
have to pass some data to asterisk so asterisk still knows that the call is
up and has to know when the call goes away. We need to know this bandwidth
usage on a T1 because lets say it was 10Kbps, you could actually do a bunch
of calls on 1 T1 provided that all phones use canreinvite right?
Nope, you've misunderstood. If these phones are connecting via SIP or 
IAX, and Asterisk is allowed to reinvite them to talk to each other, 
then Asterisk will be _completely_ out of the conversation. The only way 
that Asterisk would become involved again is if one of the phone users 
decided to transfer their end of the call.

Given this, if you allow reinvites, you _cannot_ have accurate and 
complete CDR information. Many of us would like to see Asterisk support 
this mode of operation (reinvite only the media stream, not the control 
stream), and some of those many think it's actually possible... but 
there are others who feel that since Asterisk is not a SIP proxy it 
cannot be done. I am not in that group, I just don't have the time to 
try to implement it myself :-(

In any case, you have two choices: avoid the bandwidth consumption on 
the Asterisk server's link, or have accurate CDR.
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Re: [Asterisk-Users] VoIP Termination

[Gary Carr]

Why is IAX termination better?

Gary


So they offer termination via SIP for $0.013/minute?
Even better-- IAX termination :)
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RE: [Asterisk-Users] Polycom SIP Phones

[Jared Armstrong]

I found IP 500's for $170.

Jared Armstrong
-Original Message-
From: Kristian Kielhofner [mailto:[EMAIL PROTECTED] 
Sent: Thursday, December 16, 2004 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom SIP Phones

Adam Robins wrote:

> Could someone please direct me (via personal email) to a provider with

> good prices on Polycom Soundpoint IP 500's with POE cables?  I need 14

> of them.
>  
> Thanks,
> Adam

Adam,

Why get the IP 500 when you can get the IP 600 for less?  Check
out 
www.tritechcoa.com.  They have the IP 600 for $255.  But, I think that 
this stuff should go to the -biz lists.

--
Kristian Kielhofner


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RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems

[Matt Schulte]

Anyone???

-Original Message-
From: Matt Schulte 
Sent: Thursday, December 16, 2004 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems


The first example wasn't even touching SER.. 

7960sip --> asterisk --> IAX2 --> PRI

:/

-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED] 
Sent: Thursday, December 16, 2004 9:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 7960 (SIP) hold problems


> Has anyone had problems with using hold on a 7960 SIP firmware? The
> problem is when the 7960 puts a call on hold and you take it off hold 
> again, the 7960 outbound audio is delayed on the other end. Sometimes 
> up to a few seconds. I've tried a couple different things, making the 
> "other end" a diff type of trunk ie:
> 
> 7960sip --> asterisk --> IAX2 --> PRI
> 
> 7960sip --> asterisk --> SER --> SIP proxy
> 
> Anyone have a clue? The 7960 has the latest firmware, 7.3 or
> something. Could this be a (the?) problem? Thanks!

I'm not aware of any issues. One remote internet based with g729 and
nat, another with g711, and several local. If its happening here, no one
knows about it. We're not using SER though.



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Re: [Asterisk-Users] Has anyone connected to 7960 with consolecablefor setup?

[Daniel Chester]

One thing that I noted is that you mentioned you don't see anything on
the tftp server. If tftp isn't seeing a request, that usually means you
have a switch or router that is preventing broadcasts.  What does the
top right of the phone says, if there is a SIP firmware on the phone it
should say sip.  If you have tftp running in verbose logging mode, you
should see every file request and what the server response was.  You
can use a packet sniffer and make sure that the phone is transmitting
tftp/dhcp requests.

Randy MacKay wrote:

  When I push the settings button, nothing happens.  I never get a chance to
put in the password.

I think the previous owner may have messed up a firmware upgrade.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Sean Cook
Sent: Thursday, December 16, 2004 11:40 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Has anyone connected to 7960 with
consolecablefor setup?


Why can't you use the settings button?  If
you know the password (or using the default
password) you should be able to unlock the
phone and do a hard reset...

Sean

  
  
-Original Message-
From:

  
  [EMAIL PROTECTED]
  
  [mailto:[EMAIL PROTECTED]
om] On Behalf Of
  
  
Randy MacKay
Sent: Thursday, December 16, 2004 1:35 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Has anyone

  
  connected to 7960 with
  
  
console cablefor setup?

I have a Cisco 7960 phone.  I cannot seem

  
  to use the settings
  
  
button to get into the phone to change the

  
  TFTP server.  I've
  
  
set up a DHCP Server, TFTP Server with the

  
  same address, and
  
  
the phone requests the address of 0.0.0.0,

  
  the server offers
  
  
the  address of 192.168.2.2, but the phone

  
  seems not to take it.
  
  
I have no action on the TFTP side.

So, since I can't seem to server the phone

  
  anything by TFTP,
  
  
and I can't use the settings button, then I

  
  thought I might
  
  
make a console cable (see below).  I tried

  
  to use
  
  
hyperTerminal, but got no response from the

  
  phone.
  
  
Anyone have any ideas?

Thanks,

Randy



I found a link to make a Cisco Console

  
  Cable for RJ-45.
  
  http://www.hardwarebook.net/cable/serial/cisc
oconsole9.html
  
  
  DB9F RJ45
Receive Data 		2 	3
Transmit Data 		3 	6
Data Terminal Ready 	4 	7
Ground 			5 	4
Ground 			5 	5
Data Set Ready 		6 	2
Request to Send 		7 	8
Clear to Send 		8 	1



The Console Access Manual, give the

  
  following cable information:
  
  
Console Cable Requirements
You use a serial cable with a connector to

  
  connect a PC and a
  
  
phone. The cable uses an RJ-11 connector

  
  for the phone and an
  
  
RJ-45 connector to an
RJ-45-to-DB9 converter for the PC. Table

  
  D-1 shows the pinout
  
  
requirements for the console cable.

Table D-1 Console Cable Pinouts
RJ-11 Connector 	RJ-45 Connector
Pin 2 ==	Pin 6
Pin 3 ==	Pin 4
Pin 4 ==	Pin 3

So, I thought I would go right from DB9F to

  
  RJ-11
  
  
DB9F		RJ-45		RJ-11
Pin 2		Pin 3		Pin 4
Pin 5		Pin 4		Pin 3
Pin 3		Pin 6		Pin 2
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[Asterisk-Users] zap, agents, ackcall

[Jeremy Jones]

Hi,

With "ackcall=no" set in agents.conf, agents on sip and iax channels are
not required to press # to answer.  However, this setting does not seem
to do anything for agents on zap channels.  This is with asterisk-1.0.2
and zaptel-1.0.2.  I saw one post a while back about this, but no
answer.  Anyone have insight?
-- 
Jeremy Jones <[EMAIL PROTECTED]>

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[Asterisk-Users] are there any tips/tricks to get the uip200 to register?

[Charles S. Antrim]

I 
am getting the #3 Register Error??

 

TIA

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Re: [Asterisk-Users] VoIP Termination

[Julio Arruda]

Mike Diehl (Encrypted email preferred) wrote:

I'm trying to make an issue out of this because I think it needs to change and 
I'm hoping people who are affiliated with these providers are reading this.  
I was going to go with Packet8.  I was going through the "final checklist" 
before subscribing when I came accross this fascist policy.

Sure, I can go with a business plan, but that would cost me $39.95.  That's $5 
more than I'm spending for an analog phone line!  Part of the reason for me 
to go with VoIP is to become "Quest Free."  But suddenly, Quest is starting 
to resemble the Boy Scouts when compared to the types of usage policies I'm 
seeing from some of the VoIP providers.
Sorry for the rant, but I hope you understand.

Mike,
I was a happy P8 user for 18 months, just keep in mind that theirs (and 
most of the "consumer market" voip providers) unlimited plans could be 
in theory abused by telemarketers and etc.
I just cancelled P8 because their rates to Campinas/Brazil went too 
high, but is after all, consumer VOIP.
You will notice that almost all the "unlimited VOIP plans" have some 
kind of safeguard/policy like that. The good part is, IMHO, with VOIP, 
you are really free to choose :-) and running asterisk just give you 
even more flexibility.
You may want to take also a look at http://www.dslreports.com/forum/voip
They have quite few users of each of the consumer VOIP providers, and 
you should be able to have a feeling about how good/bad are their 
experiences.
[], 

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[Asterisk-Users] Get asterisk out of the RTP stream?

[Matthew Boehm]

Here is the setup:

Phone A (in NYC) on own bandwidth.
Phone B (in LA) on own bandwidth.
Asterisk box in Houston,TX on own bandwidth.

Both phones contact asterisk to register. Not much bandwidth used for this
as it is a few packets every hour or so.

Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk
calls phone B. Both phones are connected and both people are talking.

Is all of the data/voice comming from phone A going into asterisk box and
then from asterisk box to phone B? If so, then using g711, phone A would
send/recieve 64Kbps to/from asterisk and phone B would also send/recieve
64Kbps to/from asterisk. Asterisk would then be sending/recieving 128Kbps
for this one call right? So with 1 T1 you could only get 12 calls going
right?

If I use canreinvite=yes on both phones, will phone A connect to phone B
directly therefore lowering the bandwidth usage in/out of the asterisk box
right?

If so, what is the "signalling" bandwidth usage in/out of asterisk in this
case? Even if the phones are connected directly to eachother, they still
have to pass some data to asterisk so asterisk still knows that the call is
up and has to know when the call goes away. We need to know this bandwidth
usage on a T1 because lets say it was 10Kbps, you could actually do a bunch
of calls on 1 T1 provided that all phones use canreinvite right?

Thanks,
Matthew

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Re: [Asterisk-Users] How to tell "Who's Online"?

[Matt Riddell]

Brent Goran wrote:
I've been trying to find a simple way to check "who's online", meaning
who is reachable at the moment, without actually going through and
dialing everybody. Is there a way to do this with Asterisk? I am sorry
if I'm missing something obvious, but I couldn't find any console
command to show users online.
I have developed a few apps for windows to show this information 
(including latency in ms).

You can download one of the five (depending on what you want in it) from 
http://www.sineapps.com/downloads.php

--
Cheers,
Matt Riddell
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[Asterisk-Users] Has anyone connected to 7960 with console cable for setup?

[Randy MacKay]

I have a Cisco 7960 phone.  I cannot seem to use the settings button to get
into the phone to change the TFTP server.  I've set up a DHCP Server, TFTP
Server with the same address, and the phone requests the address of 0.0.0.0,
the server offers the  address of 192.168.2.2, but the phone seems not to
take it.

I have no action on the TFTP side.

So, since I can't seem to server the phone anything by TFTP, and I can't use
the settings button, then I thought I might make a console cable (see
below).  I tried to use hyperTerminal, but got no response from the phone.

Anyone have any ideas?

Thanks,

Randy



I found a link to make a Cisco Console Cable for RJ-45.
http://www.hardwarebook.net/cable/serial/ciscoconsole9.html

DB9F RJ45
Receive Data2   3
Transmit Data   3   6
Data Terminal Ready 4   7
Ground  5   4
Ground  5   5
Data Set Ready  6   2
Request to Send 7   8
Clear to Send   8   1



The Console Access Manual, give the following cable information:

Console Cable Requirements
You use a serial cable with a connector to connect a PC and a phone. The
cable
uses an RJ-11 connector for the phone and an RJ-45 connector to an
RJ-45-to-DB9 converter for the PC. Table D-1 shows the pinout requirements
for
the console cable.

Table D-1 Console Cable Pinouts
RJ-11 Connector RJ-45 Connector
Pin 2 ==Pin 6
Pin 3 ==Pin 4
Pin 4 ==Pin 3

So, I thought I would go right from DB9F to RJ-11
DB9FRJ-45   RJ-11
Pin 2   Pin 3   Pin 4
Pin 5   Pin 4   Pin 3
Pin 3   Pin 6   Pin 2
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RE: [Asterisk-Users] How "expensive" are the different codecs?(Regarding CPU time)

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> Antony Stone schrieb:
>> On Wednesday 15 December 2004 21:26, Michael Vogel wrote:
>> 
>>> Is it a little bit too much for such a machine? What could be the
>>> bottleneck? CPU? Memory? Interrupts?
>> 
>> My advice would be to whack in a load more RAM - basically, try to
>> get the poor little thing so it doesn't need to use swap.  That will
>> make a big difference to performance.
> 
> I just doubled the memory, now I have 256mb and I am using - by now -
> zero bytes for swap ;-) 
> 
> The values at "show translation" doesn't change. And they
> change only a
> little bit when I unload the "baycom_ser_hdx"-module that generates
> three times more interrupts than the wcfxo-module.

Hmmm. I propose that we make your system the list guinea pig! If we can
get that one tweaked, there's no telling what alse we can do with
Asterisk!

Have you tried running Asterisk at pseudo-realtime priority? (asterisk
-p)


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RE: [Asterisk-Users] Has anyone connected to 7960 with consolecablefor setup?

[Randy MacKay]

The phone is a used one I picked up from ebay.  **# doesn't seem to unlock
anything.

The display of the phone says; Configuring VLAN, Configuring IP, 
(requesting ??? flashes), TFTP ??.cfg.xml , Protocol Application
Invalid.

If I could just somehow get to the TFTP Settings?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paul Brock
Sent: Thursday, December 16, 2004 11:20 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Has anyone connected to 7960 with
consolecablefor setup?


Randy,

Is it a new unit? The only reason I ask is that hitting the settings button
should let you straight in.

There is an Rs232 port on the bottom - however not oversure what it's used
for on the 7960's.

The reason I as wether it's new or not is that it might need firmware
resetting as per the cisco information (not immediately to hand).

If you can see the menu's and just chance change the setting, I think it's
something like *# or **# to allow change.

Sorry if that's "suck egg" territory - just trying to cover anything obvious
which is easily missed!!

Paul

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Randy MacKay
Sent: 16 December 2004 18:35
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Has anyone connected to 7960 with console cablefor
setup?

I have a Cisco 7960 phone.  I cannot seem to use the settings button to get
into the phone to change the TFTP server.  I've set up a DHCP Server, TFTP
Server with the same address, and the phone requests the address of 0.0.0.0,
the server offers the  address of 192.168.2.2, but the phone seems not to
take it.

I have no action on the TFTP side.

So, since I can't seem to server the phone anything by TFTP, and I can't use
the settings button, then I thought I might make a console cable (see
below).  I tried to use hyperTerminal, but got no response from the phone.

Anyone have any ideas?

Thanks,

Randy



I found a link to make a Cisco Console Cable for RJ-45.
http://www.hardwarebook.net/cable/serial/ciscoconsole9.html

DB9F RJ45
Receive Data2   3
Transmit Data   3   6
Data Terminal Ready 4   7
Ground  5   4
Ground  5   5
Data Set Ready  6   2
Request to Send 7   8
Clear to Send   8   1



The Console Access Manual, give the following cable information:

Console Cable Requirements
You use a serial cable with a connector to connect a PC and a phone. The
cable
uses an RJ-11 connector for the phone and an RJ-45 connector to an
RJ-45-to-DB9 converter for the PC. Table D-1 shows the pinout requirements
for
the console cable.

Table D-1 Console Cable Pinouts
RJ-11 Connector RJ-45 Connector
Pin 2 ==Pin 6
Pin 3 ==Pin 4
Pin 4 ==Pin 3

So, I thought I would go right from DB9F to RJ-11
DB9FRJ-45   RJ-11
Pin 2   Pin 3   Pin 4
Pin 5   Pin 4   Pin 3
Pin 3   Pin 6   Pin 2
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[Asterisk-Users] problem with freebsd 4.9 port

[Andrea Riela]

Hi folks,
when I try to compile the freebsd port (1.0.2) I see:
ast_h323.cpp: In method `void MyH323Connection::SendUserInputTone(char, 
unsigned int)':
ast_h323.cpp:725: argument passing to `const char *' from `char' lacks 
a cast
ast_h323.cpp: In method `void MyH323Connection::OnUserInputTone(char, 
unsigned int, unsigned int, unsigned int)':
ast_h323.cpp:735: argument passing to `const char *' from `char' lacks 
a cast
ast_h323.cpp: In method `void MyH323Connection::OnUserInputString(const 
PString &)':
ast_h323.cpp:746: argument passing to `const char *' from `char' lacks 
a cast
ast_h323.cpp: At top level:
chan_h323.h:31: warning: `struct sockaddr_in bindaddr' defined but not 
used
gmake[2]: *** [ast_h323.o] Error 1
gmake[2]: Leaving directory 
`/usr/ports/net/asterisk/work/asterisk-1.0.2/channels/h323'
gmake[1]: *** [h323/ast_h323.o] Error 2
gmake[1]: Leaving directory 
`/usr/ports/net/asterisk/work/asterisk-1.0.2/channels'
gmake: *** [subdirs] Error 1
*** Error code 2

Stop in /usr/ports/net/asterisk.
observe#
what could I do?
thanks for your support
Regards
Andrea
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RE: [Asterisk-Users] Calculating required bandwidth

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption
> 
> A T1 set up for voice uses G711 at 64Kbps which at 1.544Mbps
> equals 24 channels. This is also known as a PRI.

No, that is incorrect. PRI is a type of ISDN service that is typically
carried across one (or more) T1s. The PRI-ISDN service uses one of the
24 channels in the T1 as a signalling channel (or D-channel), the other
23 channels become the bearer-channels (or B-channels).

PRI is sometimes referred to as 23B+D, but in reality can be built far
more creatively than that. Many large PBXs will have a single PRI
circuit that is delivered on two T1s, with a primary and backup
D-channel in each T1 (46B+2D). If you wanted, you could save the backup
D-channel and have a 47B+D (bad idea, but technically possible). It is
even possible to have a D-channel running across a serial link,
completely out of the T1 altogether; although I haven't seen one of
these in a long time . . .

To summarize:
A T1 is a *circuit*, that can be used to carry all kinds of different
services.
PRI is one kind of *service* that is commonly carried over a T1.

Cheers,

Jim.



> 
> -Matthew
> 
> - Original Message -
> From: "Ed Greenberg" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <[EMAIL PROTECTED]>
> Sent: Thursday, December 16, 2004 11:45 AM
> Subject: [Asterisk-Users] Calculating required bandwidth
> 
> 
>> I was posed this question:
>> 
>> A T1 set up for voice carries 24 conversations on a circuit that is
>> 1.544 megabits/second. Right?
>> 
>> Well, if you set that T1 up to carry data and run a link between two
>> IP networks over it, how many SIP conversations could it be expected
>> to
> carry?
>> How about IAX?
>> 
>> How would one extend this calculation to varying bandwidth circuits
>> and various VOIP protocols (MGCP, SCCP and H323 come to mind)?
>> 
>> Rather than asking for a full education here, can somebody point me
>> at a suitable practical reference? Of course, if somebody wants to
>> actually
> post
>> the answer that'd be fine too :)
>> 
>> THanks,
>> 
>> 
>> 
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Re: [Asterisk-Users] Queueueueuueue position

[Brian Roy]

On Thu, 16 Dec 2004 15:18:10 +0100, E. Versaevel <[EMAIL PROTECTED]> wrote:

> When I call in (with an agent logged in) I get to hear the MOH on the client
> side, hover no matter how high the hold time is, I NEVER get an announcement
> over my queue position or my estimated wait time?
> Both the incoming call and the agent are on SIP channels.
> 
> What is wrong ?
> 
> Kind regards,
> 
> E. Versaevel

Would that be because this is the only call in queue? Try putting
another call in queue and see what you get.

-Chuji
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Re: [Asterisk-Users] VoIP Termination

[Michael Graves]

On Thu, 16 Dec 2004 17:05:54 -0500, Dorn Hetzel wrote:

>
>This is the sort of language you are likely to find from anyone offering
>"unlimited" plans.  It's just the reality of the fact that there is 
>no such thing as unlimited, really...  Everything has it's limits :)
>
>If you go with a per-minute plan, like from NuFone, Voipjet, etc., you
>will not find any of this sort of hoo-hah, since they WANT you to use
>more minutes, since that's how they get paid :)  
>
>Any since the per minute rates can be as low as $0.013/minute last time
>I looked, you have to use a LOT of minutes before you spend as much as
>you would have with that "unlimited" plan... 
>

This too has been  my experience. Also, you can buy DIDs from one
provider but setup outbound termination from the same, or another, or
both. I use VoipJet and SixTel. I recently dumped VoicePulse Connect
since their IAX termination has been unreliable in recent months.

Michael
--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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RE: [Asterisk-Users] Bugtracker Karma Hall Of Fame

[Brian West]

Haha but kram doesn't post ANY bugs... he's the one that awards it... and
its not really for him its for the people that commit... we all know kram
should be at +20 karma.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brian Roy
> Sent: Thursday, December 16, 2004 6:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Bugtracker Karma Hall Of Fame
> 
> On Wed, 15 Dec 2004 17:25:59 -0800, Paul Crick
> <[EMAIL PROTECTED]> wrote:
> > The Karma Hall Of Fame is now available at:
> >
> > http://bugs.digium.com/karma_halloffame.php
> >
> 
> Well the fact that Kram sits at +14 tells me this system if flawed.
> Not that I'm unappreciative of Anthony's work, but come on "14"?
> 
> -Chuji
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Re: [Asterisk-Users] Bugtracker Karma Hall Of Fame

[Brian Roy]

On Wed, 15 Dec 2004 17:25:59 -0800, Paul Crick
<[EMAIL PROTECTED]> wrote:
> The Karma Hall Of Fame is now available at:
> 
> http://bugs.digium.com/karma_halloffame.php
> 

Well the fact that Kram sits at +14 tells me this system if flawed.
Not that I'm unappreciative of Anthony's work, but come on "14"?

-Chuji
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RE: [Asterisk-Users] Hardware based DSP

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> Hi All,
> 
> Is it correct to say that by design,  asterisk wont make use
> of any cards hardware dsp capabilities ?

I think it would be more accurate to say that Asterisk will be able to
connect to ANY type of media device, as long as an Asterisk-compatible
channel has been written for it.

> I don't think that any of the hardware cards currently
> supported have any dsp capabilities, but I wanted to know if
> for example, in the future a driver was written for a card that did
> have dsp capabilities,
> would asterisk be able to make any use of it ?

Cards like that already exist. Audiotronix cards have on board DSPs, and
they work with Asterisk (I am not endorsing those cards -- I've never
used one -- just using them as an example of existing,
hardware-based-DSP cards that work with Asterisk).

> I am only just starting out with asterisk, and have not fully
> understood the architecture yet, but it seems that in order
> to handle VoIP and PSTN seamlessly, all dsp related
> functionality has to be handled by software 

A lot of DSP work happens in software, but it is not a requirement or
design limitation.



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Re: [Asterisk-Users] Voice Prompt Info

[Jeff Pratt]

[EMAIL PROTECTED] wrote:
I am trying to put together a list of 'departments' to request as 
voice prompts.  I have the biggies (sales, accounting, shipping, 
etc...) but I want to make sure I do not miss any. If anyone anyone 
has some suggestions (Ha... that is like going to an NRA meeting ans 
asking if anybody has a gun  :-)  ) please forward them to me (and / 
or post here although, with the volume of this list I do not always 
have time to read every digest so the 'and' option may be best.) so 
that I can compile a single list, verify that they are not already 
available, group them, and send them on.  Please put 'voice prompt' in 
the subject line of anything you forward me so that I am less likely 
to miss it.
I am looking for titles that fit into the string:
"press 1 for the DEPT department" or  "press 1 for DEPT"
but if you have other suggestions, let me know.
I will be collecting these for about a week so please try to get them 
to me in that time frame.
I am hopeful that, with these prompts, it will be possible to make a 
complete (albeit fairly generic) tree, all with the same voice.

Thanks;
James
alspachfam at charter dot net
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RE: [Asterisk-Users] VoIP Termination

[Jay Milk]

> So they offer termination via SIP for $0.013/minute?

Even better-- IAX termination :)

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Re: [Asterisk-Users] sox-12.17.6

[Wayne Sheppard]

TELUX wrote:
does this version work? after the asterisk MIXING of files i have a 
file of dead air.
Hmmm, in a twisted sort of way I'm glad to hear you're seeing this. I am 
too and have been losing hair with it.

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RE: [Asterisk-Users] Calculating required bandwidth

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> --On Thursday, December 16, 2004 3:59 PM -0500 Jim Van Meggelen
> <[EMAIL PROTECTED]> wrote: 
> 
>> I've always found Newton's Telecom Dictionary to be a great
>> reference. It's not too technical, packed with humour, and very
>> comprehensive. 
> 
> I have a very old copy of this, so went off to Amazon to see about a
> new one. I discovered that that a 2005 edition (21st edition)
> will be available
> in February, so I'm going to wait.

That reminds me of last January, when I said to myself "The 2004 version
is coing out in a few weeks, I'll wait".

I just picked mine up last week :-)

Regards,

Jim.

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[Asterisk-Users] ilbc and asterisk 1.0.3 - strange noises.

[Alessandro Ren]

Title: OpSign





    Have someone experienced any strange noises using the ilbc codec
after upgrading to asterisk 1.0.3?
I had to change the codec do gsm to fix this problem. The noise is very
loud, like saturation of the echo ro something, seems like the echo
cancelation is amplifying itself.
    I'be been using ilbs since asterisl 0.70 and have never had any
problem like this.
    Thanks.

    

-- 

__

  

   Alessandro Ren
  
   OpServices
  Luciana de Abreu, 471 - Sala 403
  Porto Alegre, RS - CEP 90570-060
  

  


  

   (   phone 55(51)3061-3588
  4fax 55(51)3061-3588
  
   Q   mobile 55(51)9807-3255
  :   email [EMAIL PROTECTED]
  

  

__



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[Asterisk-Users] Rapid DTMF entry failure

[Greg Smith]

 

 

Using inband DTMF signaling, our Asterisk server fails to
pass DTMF when it is entered rapidly by a caller.    For example, when a phone
number is entered, if it is entered slowly (a brief pause between each number)
Asterisk is fine.  However, if the ten digits are entered quickly, Asterisk
drops digits.

 

Thanks.


Greg






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RE: [Asterisk-Users] Has anyone connected to 7960 with console cablefor setup?

[Shane Young]

If the phone has not been converted to SIP, the console may not work.  I was 
never able to get the 
console to work on a skinny phone, but it does work on a SIP phone.


Quoting Paul Brock <[EMAIL PROTECTED]>:

> Randy,
> 
> Is it a new unit? The only reason I ask is that hitting the settings button
> should let you straight in.
> 
> There is an Rs232 port on the bottom - however not oversure what it's used
> for on the 7960's.
> 
> The reason I as wether it's new or not is that it might need firmware
> resetting as per the cisco information (not immediately to hand).
> 
> If you can see the menu's and just chance change the setting, I think it's
> something like *# or **# to allow change.
> 
> Sorry if that's "suck egg" territory - just trying to cover anything obvious
> which is easily missed!!
> 
> Paul
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Randy MacKay
> Sent: 16 December 2004 18:35
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Has anyone connected to 7960 with console cablefor
> setup?
> 
> I have a Cisco 7960 phone.  I cannot seem to use the settings button to get
> into the phone to change the TFTP server.  I've set up a DHCP Server, TFTP
> Server with the same address, and the phone requests the address of 0.0.0.0,
> the server offers the  address of 192.168.2.2, but the phone seems not to
> take it.
> 
> I have no action on the TFTP side.
> 
> So, since I can't seem to server the phone anything by TFTP, and I can't use
> the settings button, then I thought I might make a console cable (see
> below).  I tried to use hyperTerminal, but got no response from the phone.
> 
> Anyone have any ideas?
> 
> Thanks,
> 
> Randy
> 
> 
> 
> I found a link to make a Cisco Console Cable for RJ-45.
> http://www.hardwarebook.net/cable/serial/ciscoconsole9.html
> 
>   DB9F RJ45
> Receive Data  2   3
> Transmit Data 3   6
> Data Terminal Ready   4   7
> Ground5   4
> Ground5   5
> Data Set Ready6   2
> Request to Send   7   8
> Clear to Send 8   1
> 
> 
> 
> The Console Access Manual, give the following cable information:
> 
> Console Cable Requirements
> You use a serial cable with a connector to connect a PC and a phone. The
> cable
> uses an RJ-11 connector for the phone and an RJ-45 connector to an
> RJ-45-to-DB9 converter for the PC. Table D-1 shows the pinout requirements
> for
> the console cable.
> 
> Table D-1 Console Cable Pinouts
> RJ-11 Connector   RJ-45 Connector
> Pin 2 ==  Pin 6
> Pin 3 ==  Pin 4
> Pin 4 ==  Pin 3
> 
> So, I thought I would go right from DB9F to RJ-11
> DB9F  RJ-45   RJ-11
> Pin 2 Pin 3   Pin 4
> Pin 5 Pin 4   Pin 3
> Pin 3 Pin 6   Pin 2
> ---
> Outgoing mail is certified Virus Free.
> Checked by AVG anti-virus system (http://www.grisoft.com).
> Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004
> 
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RE: [Asterisk-Users] Has anyone connected to 7960 with console cablefor setup?

[Sean Cook]

Why can't you use the settings button?  If
you know the password (or using the default
password) you should be able to unlock the
phone and do a hard reset...

Sean 

> -Original Message-
> From:
[EMAIL PROTECTED] 
>
[mailto:[EMAIL PROTECTED]
om] On Behalf Of 
> Randy MacKay
> Sent: Thursday, December 16, 2004 1:35 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Has anyone
connected to 7960 with 
> console cablefor setup?
> 
> I have a Cisco 7960 phone.  I cannot seem
to use the settings 
> button to get into the phone to change the
TFTP server.  I've 
> set up a DHCP Server, TFTP Server with the
same address, and 
> the phone requests the address of 0.0.0.0,
the server offers 
> the  address of 192.168.2.2, but the phone
seems not to take it.
> 
> I have no action on the TFTP side.
> 
> So, since I can't seem to server the phone
anything by TFTP, 
> and I can't use the settings button, then I
thought I might 
> make a console cable (see below).  I tried
to use 
> hyperTerminal, but got no response from the
phone.
> 
> Anyone have any ideas?
> 
> Thanks,
> 
> Randy
> 
> 
> 
> I found a link to make a Cisco Console
Cable for RJ-45.
>
http://www.hardwarebook.net/cable/serial/cisc
oconsole9.html
> 
>   DB9F RJ45
> Receive Data  2   3
> Transmit Data 3   6
> Data Terminal Ready   4   7
> Ground5   4
> Ground5   5
> Data Set Ready6   2
> Request to Send   7   8
> Clear to Send 8   1
> 
> 
> 
> The Console Access Manual, give the
following cable information:
> 
> Console Cable Requirements
> You use a serial cable with a connector to
connect a PC and a 
> phone. The cable uses an RJ-11 connector
for the phone and an 
> RJ-45 connector to an
> RJ-45-to-DB9 converter for the PC. Table
D-1 shows the pinout 
> requirements for the console cable.
> 
> Table D-1 Console Cable Pinouts
> RJ-11 Connector   RJ-45 Connector
> Pin 2 ==  Pin 6
> Pin 3 ==  Pin 4
> Pin 4 ==  Pin 3
> 
> So, I thought I would go right from DB9F to
RJ-11
> DB9F  RJ-45   RJ-11
> Pin 2 Pin 3   Pin 4
> Pin 5 Pin 4   Pin 3
> Pin 3 Pin 6   Pin 2
> ---
> Outgoing mail is certified Virus Free.
> Checked by AVG anti-virus system
(http://www.grisoft.com).
> Version: 6.0.817 / Virus Database: 555 -
Release Date: 12/15/2004
> 
>
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[Asterisk-Users] Can I read more than 7 numbers from capi ?

[Robert Rozman]

Hi,

we have 7 numbers in out country phone system. If I dial 123456766 and
1234567 is my number - is there any way to get those two extra numbers in
Asterisk for transfer to local extension ?



I have AVM Fritz with CAPI driver.

Thanks in advance,

regards,

Rob.

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Re: [Asterisk-Users] Which Primary ISDN card to use in Europe ?

[Peter Svensson]

On Thu, 16 Dec 2004, Robert Rozman wrote:

> I'd kindly ask for any advices, experiences and opinions on what Primary
> (and also several BRI) cards should be used on EuroISDN and Asterisk ?
> 
> I'm also curious what features do these cards offer regarding echo
> cancellation (mostly when calling from analog lines) ?

For pri the conventional choice for use with Asterisk are the 
TE405P/TE410P cards. Only software echo canceling is available. I do not 
know how efficient it is, since we have it diabled.

For several BRI channels in one computer I think the quadbri/octobri from 
Junghanns offer the best features. They have an internal TDM bus for 
native bridging.

Peter


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Re: [Asterisk-Users] Connecting Asterisk to GSM

[jurgen]

Hi Martin,

I've looked at a few different options, including 2N's 4-channel
SIP-to-GSM gateways, and the cheapest and most reliable I've been able
to find (at least for people who need fewer than 8 ports) is a
combination of Digium's 4-port FXO card and four Telular PhoneCell
SE5e units. When you get into larger needs than that, there are a
bunch of other options available, including PRI as Matteo suggested.
Check into the Wiki
(http://www.voip-info.org/wiki-Asterisk+Connecting+to+the+Cellular+Network
) for lots more products and information, including an interesting
option called CellSocket, which might be cheaper for you.

.jurgen


On Thu, 16 Dec 2004 19:27:06 +, Martin List-Petersen
<[EMAIL PROTECTED]> wrote:
> On Thu, 2004-12-16 at 21:47, Jean-Michel Hiver wrote:
> > Hi List,
> >
> > I was wondering if there was any device I could use to connect * to GSM
> > networks. I don't need much capacity, maybe 2-4 GSM channels. As usual,
> > cheap is better :-)
> >
> 
> What you are looking for is something like the Ateus GSM to PSTN or ISDN
> gateways
> (http://www.mobilecomms-technology.com/contractors/gsm/2n_tele/)
> 
> Cheaper would be some gsm to pstn adapter, that you can connect to the
> cellphone. Check the archives of the asterisk-users for that, because
> it's something, that commonly has been asked before.
> 
> Another alternative would be chan_bluetooth,
> (http://www.crazygreek.co.uk/content/chan_bluetooth), but that is in a
> state far from working.
> 
> Slán leat,
> Martin List-Petersen
> Dublin, Eire
> (contact info on --> http://www.marlow.dk/)
> 
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-- 
[EMAIL PROTECTED] is jurgen's gmail address.
Visit http://jurgen.ca/ for more yummy goodness.
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RE: [Asterisk-Users] sox-12.17.6

[Brian West]

I noticed this too.  Thought I was just crazy.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of TELUX
> Sent: Thursday, December 16, 2004 12:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] sox-12.17.6
> 
> does this version work? after the asterisk MIXING of files i have a file
> of dead air.
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[Asterisk-Users] Low-latency kernel?

[Rich Adamson]

While trying to apply the low-latency kernel patches to our RHv9
Linux 2.4.20-31.9, the patches would not apply. In comparing one
of the first patch files (lowlatency.h) to that already on the system,
it would appear the low latency patches were already applied by RH.

The original RHv9 file (lowlatency.h) even had the patch author's
name/credit in it.

Does anyone know whether RH made an effort to incorporate the patches,
and if so, about what kernel version?

Rich



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[Asterisk-Users] Asterisk Cisco CallManager Integration

[Adi Linden]

Hi,

Where can I find information on H.323 for Asterisk and/or integration with
Cisco CallManager in particular?


I have oh323 working on Asterisk. Since the CallManger I am working with
is running 3.3.3 I cannot use SIP...

Thanks,
Adi
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Re: [Asterisk-Users] MusicOnHold. not getting it.

[Mark Phillips]

This is well documented in the WIKI.

And, it's not configured from the extensions.conf file. Look for
musiconhold.conf in /etc/asterisk. Genreally one doesn't have to mess
with it but you can do all sorts of neat tricks with it. I have our
office one doing different hold music for different departments so that
they can have their own messages etc played when someone is on hold.

If you want to her your hold music ad a line like this to your
extensions.conf

exten => ,1,Musiconhold(default)

Which will play all the hold music until you hangup.

If you want to play the hold music to peeps while your phone is ringing
do this;

exten => ,1,Dial(SIP/|20|m)

which will play the music for 20 seconds whilst ringing the phone.

Mark

On Thu, 2004-12-16 at 16:57, Ferguson, Michael wrote:
>  
>  
> G'Day All;
>  
> I am a little unsure on how to get Music On Hold to work. Please
> critique my extensions.conf.  ? Thanks
>  
> 
> ; SIP 5001
> 
> exten => 5001,1,Dial(SIP/5001)
> 
> exten => 5001,2,Voicemail(u${EXTEN})
> 
> exten => 5001,3,Hangup
> 
> exten => 5001,102,Voicemail(b${EXTEN})
> 
> exten => 5001,103,Hangup
> 
>  
> 
>  
> 
> Thanks
> 
> 
> 
> __
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Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
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Re: [Asterisk-Users] Polycom SIP Phones

[Antony Stone]

On Thursday 16 December 2004 22:57, Jared Armstrong wrote:

> I found IP 500's for $170.

Where?

Antony

-- 
The truth is rarely pure, and never simple.

 - Oscar Wilde

 Please reply to the list;
   please don't CC me.
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Re: [Asterisk-Users] How "expensive" are the different codecs? (Regarding CPU time)

[Michael Vogel]

Antony Stone schrieb:
On Wednesday 15 December 2004 21:26, Michael Vogel wrote:
Is it a little bit too much for such a machine? What could be the
bottleneck? CPU? Memory? Interrupts?
My advice would be to whack in a load more RAM - basically, try to get the 
poor little thing so it doesn't need to use swap.  That will make a big 
difference to performance.
I just doubled the memory, now I have 256mb and I am using - by now - 
zero bytes for swap ;-)

The values at "show translation" doesn't change. And they change only a 
little bit when I unload the "baycom_ser_hdx"-module that generates 
three times more interrupts than the wcfxo-module.

Bye!
Michael
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Re: [Asterisk-Users] Polycom SIP Phones

[Kristian Kielhofner]

Jared Armstrong wrote:
I found IP 500's for $170.
Froogle can find them for that, just make sure that you don't get raked 
over the coals on shipping ($27 to ship a phone some places)!  Yikes! 
Anyways, by the time you but PoE cables you are already talking $200. 
I, personally, would rather have an IP 600 with six line appearances, 
XHTML browser, and PoE built in for a little extra (not having to worry 
about the PoE shenanigans is alone worth it)...

Just my opinion...
--
Kristian Kielhofner
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Re: [Asterisk-Users] VoIP Termination

[Gary Carr]

So they offer termination via SIP for $0.013/minute?

Gary

Any since the per minute rates can be as low as $0.013/minute last time
I looked, you have to use a LOT of minutes before you spend as much as
you would have with that "unlimited" plan...
Regards,
-Dorn
p.s. I use both NuFone and VoipJet and am reasonably happy with both.

On Thu, Dec 16, 2004 at 01:53:44PM -0700, Mike Diehl (Encrypted email 
preferred) wrote:
On Thursday 16 December 2004 05:17 am, Andrew Kohlsmith wrote:
> On December 16, 2004 12:03 am, Mike Diehl (Encrypted email preferred) 
> wrote:
> > One of the catches is that I often telecommute and sometimes I do 
> > some
> > side business; these practices violate many provider's acceptable use
> > policies. So, I need a provider who doesn't care how I use the phone, 
> > and
> > one that works well with Asterisk.
>
> You've gotta be kidding, VOIP providers are trying to regulate who you 
> can
> call?  Go with Nufone or iax.cc or even voicepulse connect -- use IAX2 
> over
> SIP, IMO it's just better.

Thanx, I will look into these providers.
This is an exerpt from Packet8's Terms of Use statement.  I've edited it 
for
space, but I've tried to retain the context:
--
PERSONAL USE. 8x8's Service Plans for residential subscribers that offer
unlimited minutes of PSTN calls ("Unlimited PSTN Plans") are for the
reasonable personal residential use of End User only. End Users of 
Unlimited
PSTN Plans shall not use the Services for commercial or governmental 
purposes
or for profit or non-profit activities, including, but not limited to, 
home
office, business, sales, tele-commuting, autodialing, continuous or 
extensive
call forwarding, continuous connectivity, fax broadcast, fax blasting,
telemarketing or any other activity that would be inconsistent with 
personal
and residential usage. 8x8 reserves the right to immediately terminate or
modify the Services of any End User using Unlimited PSTN Plans if 8x8
determines, in its sole discretion, that End User is not using the 
Unlimited
PSTN Plans for End User's reasonable personal residential use.
--

Now I agree with their policy on fax-blasting, etc.  But according to 
them, I
can't use my own phone for charity work?  I work at a national lab; would 
my
wife be alowed to call me at work?  Or would the be a "governmental 
purpose?"

It gets better... If Packet8 decides, in THEIR SOLE DISCRETION, that I'm
conducting a business with my phone, they can terminate my service, or
increase the price of it.
I'm trying to make an issue out of this because I think it needs to 
change and
I'm hoping people who are affiliated with these providers are reading 
this.
I was going to go with Packet8.  I was going through the "final 
checklist"
before subscribing when I came accross this fascist policy.

Sure, I can go with a business plan, but that would cost me $39.95. 
That's $5
more than I'm spending for an analog phone line!  Part of the reason for 
me
to go with VoIP is to become "Quest Free."  But suddenly, Quest is 
starting
to resemble the Boy Scouts when compared to the types of usage policies 
I'm
seeing from some of the VoIP providers.

Sorry for the rant, but I hope you understand.
--
Mike
gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc
83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB
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Re: [Asterisk-Users] Multiple IAX client behind a NAT

[Michael Graves]

I believe that the parameter is notransfer=no in the specific case of
IAX2.

Michael

On Thu, 16 Dec 2004 10:01:39 -0600, Justin Carlson wrote:

>I wonder if you can put can reinvite=yes in the iax2.conf file like we
>use in our sip.conf file to do what you are requesting.
>
>I believe it should tell the phones to do what you wish
>
>On Thu, 2004-12-16 at 17:40 +0200, CuPoTKa wrote:
>> Hello!
>> 
>> I have a number of IAX clients behind a NAT (on the same LAN) and 
>> asterisk server on the Internet. And that clients doesn't speak directly 
>> to each other, it goes through the asterisk server.
>> What should I configure to make IAX clients on the same LAN to speak 
>> directly, please?
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--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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[Asterisk-Users] Hardware based DSP

[Shahed]

Hi All,
Is it correct to say that by design,  asterisk wont make use of any cards
hardware dsp capabilities ?
I don't think that any of the hardware cards currently supported
have any dsp capabilities, but I wanted to know if for example,
in the future a driver was written for a card that did have dsp 
capabilities,
would asterisk be able to make any use of it ?

I am only just starting out with asterisk, and have not fully understood
the architecture yet, but it seems that in order to handle VoIP and PSTN
seamlessly, all dsp related functionality has to be handled by software 
Thanks
Shahed
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RE: [Asterisk-Users] Polycom FX Video Unit - asterisk-oh323

[Tim Courcy]

I had this setup and working fine on *
simply setup and extension number in exten.conf and
point it to the IP of the Viewstation.

 

 

 

-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, December 16, 2004
1:36 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Polycom
FX Video Unit - asterisk-oh323

 


I'm installing an office in a couple of weeks that will
have some nice Polycom FX video units in the conference rooms. I'm thinking
that with asterisk-oh323 
http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/#section2

I
should hopefully get the ability for phone users to dial an extension and
participate in video conferences, or just simply phone conference with users in
the room (would be able to use the multiple high quality mics that the Polycom
has, and avoid purchasing a separate conference phone). 

Any
tips / suggestions? I'm unfamiliar with the asterisk-oh323 stuff.


Regards
and TIA, 
-Ron






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[Asterisk-Users] Which Primary ISDN card to use in Europe ?

[Robert Rozman]

Hi,

I'd kindly ask for any advices, experiences and opinions on what Primary
(and also several BRI) cards should be used on EuroISDN and Asterisk ?

I'm also curious what features do these cards offer regarding echo
cancellation (mostly when calling from analog lines) ?

Thanks in advance,

regards,

Rob.

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Re: [Asterisk-Users] Dynamically Choose Codec for Bandwidth Management

[Stefan de Konink]

[EMAIL PROTECTED] wrote:
Is there any way to set Asterisk to choose what codec to allow for a new 
call based on current usage?
I think there is a way. Since I'm not in the stage yet to configure my 
extensions.conf on that deep level I found some clues.

http://www.voip-info.org/wiki-Asterisk+variables
${SIP_CODEC}: Used to set the SIP codec for a call
Probably if you make the call go thru an extension which checks current 
bandwidth consumption via an external program. (Something AGI) You could 
make the call jump to an low/normal/high bandwidth setting by set the 
SIP_CODEC for the to be used codec. With a bit of magic you probably can 
 check the amount of free G729 licences too.

Greetings,
Stefan de Konink
ps. The idea is neat... I'm definately going to try to work out some code.
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Re: [Asterisk-Users] VoIP Termination

[Dorn Hetzel]

This is the sort of language you are likely to find from anyone offering
"unlimited" plans.  It's just the reality of the fact that there is 
no such thing as unlimited, really...  Everything has it's limits :)

If you go with a per-minute plan, like from NuFone, Voipjet, etc., you
will not find any of this sort of hoo-hah, since they WANT you to use
more minutes, since that's how they get paid :)  

Any since the per minute rates can be as low as $0.013/minute last time
I looked, you have to use a LOT of minutes before you spend as much as
you would have with that "unlimited" plan... 

Regards,

-Dorn

p.s. I use both NuFone and VoipJet and am reasonably happy with both.



On Thu, Dec 16, 2004 at 01:53:44PM -0700, Mike Diehl (Encrypted email 
preferred) wrote:
> On Thursday 16 December 2004 05:17 am, Andrew Kohlsmith wrote:
> > On December 16, 2004 12:03 am, Mike Diehl (Encrypted email preferred) wrote:
> > > One of the catches is that I often telecommute and sometimes I do some
> > > side business; these practices violate many provider's acceptable use
> > > policies. So, I need a provider who doesn't care how I use the phone, and
> > > one that works well with Asterisk.
> >
> > You've gotta be kidding, VOIP providers are trying to regulate who you can
> > call?  Go with Nufone or iax.cc or even voicepulse connect -- use IAX2 over
> > SIP, IMO it's just better.
> 
> Thanx, I will look into these providers.
> 
> This is an exerpt from Packet8's Terms of Use statement.  I've edited it for 
> space, but I've tried to retain the context:
> --
> PERSONAL USE. 8x8's Service Plans for residential subscribers that offer 
> unlimited minutes of PSTN calls ("Unlimited PSTN Plans") are for the 
> reasonable personal residential use of End User only. End Users of Unlimited 
> PSTN Plans shall not use the Services for commercial or governmental purposes 
> or for profit or non-profit activities, including, but not limited to, home 
> office, business, sales, tele-commuting, autodialing, continuous or extensive 
> call forwarding, continuous connectivity, fax broadcast, fax blasting, 
> telemarketing or any other activity that would be inconsistent with personal 
> and residential usage. 8x8 reserves the right to immediately terminate or 
> modify the Services of any End User using Unlimited PSTN Plans if 8x8 
> determines, in its sole discretion, that End User is not using the Unlimited 
> PSTN Plans for End User's reasonable personal residential use.
> --
> 
> Now I agree with their policy on fax-blasting, etc.  But according to them, I 
> can't use my own phone for charity work?  I work at a national lab; would my 
> wife be alowed to call me at work?  Or would the be a "governmental purpose?"
> 
> It gets better... If Packet8 decides, in THEIR SOLE DISCRETION, that I'm 
> conducting a business with my phone, they can terminate my service, or 
> increase the price of it.
> 
> I'm trying to make an issue out of this because I think it needs to change 
> and 
> I'm hoping people who are affiliated with these providers are reading this.  
> I was going to go with Packet8.  I was going through the "final checklist" 
> before subscribing when I came accross this fascist policy.
> 
> Sure, I can go with a business plan, but that would cost me $39.95.  That's 
> $5 
> more than I'm spending for an analog phone line!  Part of the reason for me 
> to go with VoIP is to become "Quest Free."  But suddenly, Quest is starting 
> to resemble the Boy Scouts when compared to the types of usage policies I'm 
> seeing from some of the VoIP providers.
> 
> Sorry for the rant, but I hope you understand.
> 
> -- 
> Mike
> gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc
> 83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB
> 
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Re: [Asterisk-Users] VoIP Termination

[Mike Diehl (Encrypted email preferred)]

On Thursday 16 December 2004 02:45 pm, Julio Arruda wrote:
>
> Mike,
> I was a happy P8 user for 18 months, just keep in mind that theirs (and
> most of the "consumer market" voip providers) unlimited plans could be
> in theory abused by telemarketers and etc.

One of the other responders told me that P8 didn't allow termination to 
Asterisk.  Did you simply do it on the sly?  Or is this simply not true?

> You will notice that almost all the "unlimited VOIP plans" have some
> kind of safeguard/policy like that. The good part is, IMHO, with VOIP,
> you are really free to choose :-) and running asterisk just give you
> even more flexibility.

Exactly!  Real choice.  And it turns out that I don't have to shop "features" 
between the various providers.  Any feature that they may provide, I can 
provide with Asterisk.  That simplified my choice.  It came down to money and 
policy.

> You may want to take also a look at http://www.dslreports.com/forum/voip
> They have quite few users of each of the consumer VOIP providers, and
> you should be able to have a feeling about how good/bad are their
> experiences.

I will look at this.  Thanx,
-- 
Mike
gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc
83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB

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Re: [Asterisk-Users] Polycom SIP Phones

[Kristian Kielhofner]

Adam Robins wrote:
Could someone please direct me (via personal email) to a provider with 
good prices on Polycom Soundpoint IP 500's with POE cables?  I need 14 
of them.
 
Thanks,
Adam
Adam,
	Why get the IP 500 when you can get the IP 600 for less?  Check out 
www.tritechcoa.com.  They have the IP 600 for $255.  But, I think that 
this stuff should go to the -biz lists.

--
Kristian Kielhofner
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RE: [Asterisk-Users] Calculating required bandwidth

[Ed Greenberg]

--On Thursday, December 16, 2004 3:59 PM -0500 Jim Van Meggelen 
<[EMAIL PROTECTED]> wrote:

I've always found Newton's Telecom Dictionary to be a great reference.
It's not too technical, packed with humour, and very comprehensive.
I have a very old copy of this, so went off to Amazon to see about a new 
one. I discovered that that a 2005 edition (21st edition) will be available 
in February, so I'm going to wait.


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Re: [Asterisk-Users] asterisk on FC3

[C F]

I have asterisk on FC3 64BIT runs nice, the only issue I had was:
1. zpdummy doesn't load (I don't know if it is FC3 64 bit related or
not haven't have time to trouble shoot it yet).
2. Festival and asterisk don't seem to comunicate nicely, at least not
with the latest version, and the old festivals don't work on 64bit.


On Thu, 16 Dec 2004 15:48:19 +0500, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> Hello,
>  Since FC3 has been a very recent release
> I was just wondering if there are issues related
> to asterisk installation on FC3.
> 
> Thanks
> 
> Varun
> 
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[Asterisk-Users] MusicOnHold. not getting it.

[Ferguson, Michael]

Title: Message



 
 
G'Day 
All;
 
I am a 
little unsure on how to get Music On Hold to work. Please critique my 
extensions.conf.  ? Thanks
 

; SIP 5001
exten => 5001,1,Dial(SIP/5001)
exten => 5001,2,Voicemail(u${EXTEN})
exten => 5001,3,Hangup
exten => 5001,102,Voicemail(b${EXTEN})
exten => 5001,103,Hangup
 
 
Thanks
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RE: [Asterisk-Users] Bugtracker Karma Hall Of Fame

[Paul Crick]

> That's nice, and I'm #14 on the list so far, but it would
> be more valuable if karma-awarding policies were followed
> a little more consistently :-)
Maybe we need a karma arbitration committee..

Sorry, silly mood today ;-)

But seriously, if you think you're owed karma for something and haven't
received it, flag it to a bug marshall. I'm not one, I just did the web
stuff.

Cheers
Paul

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Re: [Asterisk-Users] Voipjet problems

[Julio Arruda]

I would guess they really had some problems yesterday.
I had some failed calls in my * at home, had it rerouted via Nufone, 
since the mother-in-law-retry timer was set too low, and I didn't want 
to hear complaints when I arrived home :-). I'll try to switch it back 
later today.

Ed Greenberg wrote:
I made a few voipjet calls today and they all went through just fine.
--On Wednesday, December 15, 2004 9:26 PM -0300 Gustavo Russo 
<[EMAIL PROTECTED]> wrote:

 Anybody is experimenting problems with Voipjet lately ?
Last 2 days we are having some intermitent problems in which, after
accepting the call, the error at the Asterisk console is :
 == No one is available to answer at this time
 Voipjet technical support at this time was not able to fix the problem.
 Regards
Gustavo
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[Asterisk-Users] Polycom SIP Phones

[Adam Robins]

Could someone please 
direct me (via personal email) to a provider with good prices on Polycom 
Soundpoint IP 500's with POE cables?  I need 14 of 
them.
 
Thanks,
Adam
 




Adam S. RobinsExecutive Vice President & CIO
PHARMACENTRA, LLP 5901B Peachtree Dunwoody 
Road, Suite 380Atlanta, GA 30328
Office:  770-395-0088 x34Fax: 
 770-395-0989Mobile: 
770-855-1360Email:  [EMAIL PROTECTED]Web:http://www.pharmacentra.com 



 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.
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RE: [Asterisk-Users] Has anyone connected to 7960 withconsolecablefor setup?

[Randy MacKay]

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Henry
> Devito
> Sent: Thursday, December 16, 2004 12:12 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Has anyone connected to 7960
> withconsolecablefor setup?
>
>
> > I have a Cisco 7960 phone.  I cannot seem to use the settings button to
> > get
> > into the phone to change the TFTP server.  I've set up a DHCP
> Server, TFTP
> > Server with the same address, and the phone requests the address of
> > 0.0.0.0,
> > the server offers the  address of 192.168.2.2, but the phone
> seems not to
> > take it.
> >
> > I have no action on the TFTP side.
> >
> > So, since I can't seem to server the phone anything by TFTP, and I can't
> > use
> > the settings button, then I thought I might make a console cable (see
> > below).  I tried to use hyperTerminal, but got no response from
> the phone.
> >
> > Anyone have any ideas?
> >
> > Thanks,
> >
> > Randy
>
> You have to enable the console with the phone programming.  2
> things unlock
> the settings by pressing **# settings then you can go into
> network setup and
> keep scrolling down until you see alternate tftp yes/no.  Make
> sure that it
> says yes.  Save and reboot phone then you should be able to set the tftp
> server by pressing **# settings then scroll down to the TFTP server and
> enter in the correct ip address, save and reboot. If this is a newer SIP
> version you can press the settings button press the digit 9 then enter the
> password "cisco"
>
>
> Henry Devito


I'm assuming that the phone has not been upgraded to SIP yet.  I'm also
assuming the software upgrade was messed up also.

I tried pressing **#, nothing
I tried pressings settings, then 9, nothing
I tried pressings settings and 9, nothing

The top of the screen shows "Universal Application Loader"
The bottom of the screen shows " Protocol Application Invalid"
Occasionally the bottom of the screen flashes something about TFTP
SEP0002B9EB0f06.cnf.xml
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004

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Re: [Asterisk-Users] Calculating required bandwidth

[Andrew Kohlsmith]

On December 16, 2004 03:49 pm, Race Vanderdecken wrote:
> Also remember that a telephone conversation is 2/3's silence. ( I speak,
> silence, then you speak. See the book at bought on Amazon 4 years ago
> but can't remember the name of the book.)IP only sends the data when
> there is noise versus the T1 which is a constant TDM stream. So I

Incorrect.  Only when VAD is active is silence supression used.  Asterisk does 
not currently support VAD in any form since the RTP stream is used as a clock 
source.

> predict in testing with good VoIP equipment you can get more then 24
> G.711 calls per T1. So take that and comment. You should be able to get
> more VoIP calls, my prediction is 40 G.711 well behaved calls with
> silence suppression per T1. Why else would the Baby Bells move to VoIP?

Because they'll transcode to GSM (minimum) and call it toll quality?  :-)  

-A.
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[Asterisk-Users] Dynamically Choose Codec for Bandwidth Management

[rsenykoff]

Is there any way to set Asterisk to
choose what codec to allow for a new call based on current usage? In other
words... be able to define a max number of ulaw calls, then after that
only allowing g729? The idea here is that in general, a T-1 should be enough
for our offices to have phone + citrix + some video (got good QoS in place
already). But for usage spikes, user experience would be kept good if we
could shift it into using g729.


-Ron

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Re: [Asterisk-Users] Connecting Asterisk to GSM

[Leandro Morgado]

Jean-Michel Hiver wrote:
Hi List,
I was wondering if there was any device I could use to connect * to 
GSM networks. I don't need much capacity, maybe 2-4 GSM channels. As 
usual, cheap is better :-)

I've used Nokia 32 with a TDM400 FXO. It works reasonably well but has 
some anoying "features", like taking 5-10 seconds to establish a call, 
hangup detection problems, cdr records always being answered, etc, etc. 
It's doable but not perfect.

Any tips on this?
I haven't tried ISDN GSM Terminals but do have fixed-land ISDN. ISDN is 
so much more reliable (faster setup, proper cdr and billing seconds, 
better voice quality) that I would consider a ISDN GSM terminal. Try 
searching google for quasar smartcell.

A Voip-GSM terminal sounds even better. No need for FXO, ISDN, etc. And 
you could manage it remotly. The Ateus Voice Blue looks good but I 
haven't had a change to try it out. Google around.. you find nice stuff 
like: http://www.thehightechstore.com/cellinterface.html

Let me know about your experiences.
Regards,
Leandro Morgado
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[Asterisk-Users] Steps to configure D/41EPCI card

[jjara]

Hi,

 

Somebody can give me the necessary steps for configuring a
D/41EPCI in Asterisk.

 

Thanks in advance,

 

Jorge

 






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RE: [Asterisk-Users] native MOH with Asterisk 1.0.3

[Brian West]

That moh_stop is for the MusicOnHold application ONLY.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kristian Kielhofner
> Sent: Thursday, December 16, 2004 2:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] native MOH with Asterisk 1.0.3
> 
> Kevin P. Fleming wrote:
> 
> > Kristian Kielhofner wrote:
> >
> >> Does anyone have a WORKING native MOH patch for Asterisk 1.0.3?
> >
> >
> > We are running the last version I posted to Mantis, coupled with
> > twisted's "moh stop" patch that's also in Mantis, and it seems to be
> > working fine.
> >
> > However, the patch I posted was not made against the 1.0 branch, so it's
> > possible it won't work there; I don't think there would be a problem
> > using it with 1.0.3, though.
> 
> 
> Kevin,
> 
>   Thanks for the quick reply.  I was actually following your and
> anthm's
> patch, but it will not apply to 1.0 code...
> 
> What is the bug ID for moh stop?
> 
> --
> Kristian Kielhofner
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Re: [Asterisk-Users] VoIP Termination

[Mike Diehl (Encrypted email preferred)]

On Thursday 16 December 2004 05:17 am, Andrew Kohlsmith wrote:
> On December 16, 2004 12:03 am, Mike Diehl (Encrypted email preferred) wrote:
> > One of the catches is that I often telecommute and sometimes I do some
> > side business; these practices violate many provider's acceptable use
> > policies. So, I need a provider who doesn't care how I use the phone, and
> > one that works well with Asterisk.
>
> You've gotta be kidding, VOIP providers are trying to regulate who you can
> call?  Go with Nufone or iax.cc or even voicepulse connect -- use IAX2 over
> SIP, IMO it's just better.

Thanx, I will look into these providers.

This is an exerpt from Packet8's Terms of Use statement.  I've edited it for 
space, but I've tried to retain the context:
--
PERSONAL USE. 8x8's Service Plans for residential subscribers that offer 
unlimited minutes of PSTN calls ("Unlimited PSTN Plans") are for the 
reasonable personal residential use of End User only. End Users of Unlimited 
PSTN Plans shall not use the Services for commercial or governmental purposes 
or for profit or non-profit activities, including, but not limited to, home 
office, business, sales, tele-commuting, autodialing, continuous or extensive 
call forwarding, continuous connectivity, fax broadcast, fax blasting, 
telemarketing or any other activity that would be inconsistent with personal 
and residential usage. 8x8 reserves the right to immediately terminate or 
modify the Services of any End User using Unlimited PSTN Plans if 8x8 
determines, in its sole discretion, that End User is not using the Unlimited 
PSTN Plans for End User's reasonable personal residential use.
--

Now I agree with their policy on fax-blasting, etc.  But according to them, I 
can't use my own phone for charity work?  I work at a national lab; would my 
wife be alowed to call me at work?  Or would the be a "governmental purpose?"

It gets better... If Packet8 decides, in THEIR SOLE DISCRETION, that I'm 
conducting a business with my phone, they can terminate my service, or 
increase the price of it.

I'm trying to make an issue out of this because I think it needs to change and 
I'm hoping people who are affiliated with these providers are reading this.  
I was going to go with Packet8.  I was going through the "final checklist" 
before subscribing when I came accross this fascist policy.

Sure, I can go with a business plan, but that would cost me $39.95.  That's $5 
more than I'm spending for an analog phone line!  Part of the reason for me 
to go with VoIP is to become "Quest Free."  But suddenly, Quest is starting 
to resemble the Boy Scouts when compared to the types of usage policies I'm 
seeing from some of the VoIP providers.

Sorry for the rant, but I hope you understand.

-- 
Mike
gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc
83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB

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