[asterisk-users] Hi

[Hossein moradi fard]

Hi all
this for subscribe
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] hi list need your help

[Антон Сацкий]

facing problem with  originating  webrtc calls


1-when iam  doing call from webrtc iget ice working
<--- SIP read from WS:91.196.158.205:1466 --->
INVITE sip:0669197533@77.91.132.9 SIP/2.0
Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315
Max-Forwards: 69
To: 
From: "Anton" ;tag=5i21qaop43
Call-ID: ocq4hu8eol3kijsgvt6b
CSeq: 1465 INVITE
Authorization: Digest algorithm=MD5, username="1065", realm="77.91.132.9",
nonce="5152b137", uri="sip:0669197533@77.91.132.9",
response="446883f3c97a49ea7a9a554a1ba31b6a"
X-Can-Renegotiate: true
Contact: 
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.26
Content-Length: 2554

v=0
o=- 4785391175048354014 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
m=audio 2313 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 192.168.88.26
a=rtcp:2313 IN IP4 192.168.88.26
a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=ice-ufrag:8nMZ7w8bHdBBoY1a
a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR
a=fingerprint:sha-256
6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3696151487 cname:jXfPZ33h32Mx9liw
a=ssrc:3696151487 msid:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
8a2acec3-8511-4d36-9b51-05b8752c2ddd
a=ssrc:3696151487 mslabel:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
a=ssrc:3696151487 label:8a2acec3-8511-4d36-9b51-05b8752c2ddd
m=video 2313 RTP/SAVPF 100 116 117 96
c=IN IP4 192.168.88.26
a=rtcp:2313 IN IP4 192.168.88.26
a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=ice-ufrag:8nMZ7w8bHdBBoY1a
a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR
a=fingerprint:sha-256
6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:100 VP8/9
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/9
a=rtpmap:117 ulpfec/9
a=rtpmap:96 rtx/9
a=fmtp:96 apt=100

2-BUT when i do channel originate sip/GOROD/X extension 1065@office
-- Executing [1065@office:1] Dial("SIP/GOROD-0004", "SIP/1065") in
new stack
  == Using SIP RTP CoS mark 5
[Apr 28 14:07:47] ERROR[4006][C-0032]: netsock2.c:269
ast_sockaddr_resolve: getaddrinfo("7cvtd9ihs2e8.invalid", "(null)", ...):
Name or service not known
[Apr 28 14:07:47] WARNING[4006][C-0032]: chan_sip.c:15869
__set_address_from_contact: Invalid host name in Contact: (can't resolve in
DNS) : '7cvtd9ihs2e8.invalid'
[Apr 28 14:07:47] ERROR[4006][C-0032]: netsock2.c:98
ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
Audio is at 16476
Adding codec 13 (ulaw) to SDP
Adding codec 12 (gsm) to SDP
Adding codec 14 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 91.196.158.205:1466:
INVITE sip:0momhddj@7cvtd9ihs2e8.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 77.91.132.9:5060;branch=z9hG4bK3f293d79;rport
Max-Forwards: 70
From: "asterisk" ;tag=as78119d2b
To: 
Contact: 
Call-ID: 17a96e0848cdd7d226d3665a36c65c77@77.91.132.9:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.15.0
Date: Tue, 28 Apr 2015 11:07:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 437

v=0
o=root 1122885298 1122885298 IN IP4 77.91.132.9
s=Asterisk PBX 11.15.0
c=IN IP4 77.91.132.9
t=0 0
m=audio 16476 RTP/SAVPF 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256
CC:82:C8:04:1F:DC:FE:B7:56:27:26:FF:18:CD:BB:71:99:B8:97:F9:81:2B:

[asterisk-users] HI

[David @ULC]

hI
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Hi all

[mahesh katta]

Hi ,

telco=>asteriskserver=>eppbx

I had two 4 span cards in server , one 4span connected from telco and
another one is connected to eppbx.
and when i am run the dahdi_tool commad it was showing like spna1 to span4
then also span1 to span4 all is showin OK .but i need to show span1 to
span8.
and i am dialing the no. its going hangup like *span 1 got hangup,cause 28 .
*

please help me how can i solve this problem.



-- 
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
303, Gagangiri Apts, Parleshwar Road, Ville Parle East, Mumbai - 400057.
GSM +91.97029.70779 | Phone +91.22.2663.1811 | Fax +91.22.2663.1811
Web http://www.buzzworks.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi, agent intro-speech for outside caller

[Steve Edwards]

On Thu, 20 Jan 2011, Danny Nicholas wrote:

All Asterisk prompts are configurable with a little legwork.  Simply use 
the CLI to see what is playing at the point you want to change, then set 
up this little ditty to override it.  Say you wanted to record the 
“canned” tt-weasels prompt (“Weasels have eaten our phone system”).  
This 3-liner lets you re-record it.


-  exten => 999,1,answer
-  exten => 999,n,record(tt-weasels.gsm)
-  exten => 999,n,hangup

If you are using a codec other than gsm you would replace gsm with wav, 
slin, etc.


Another technique is to fiddle with the LANGUAGE channel variable. For 
example (off the top of my head):


exten = s,n,set(CHANNEL(language)=mike)
exten = s,n,playback(agent-intro)
...

Then, record /var/lib/asterisk/sounds/agent-intro.wav to be something 
generic like "Hi. I don't know what my name is, but your call is 
exceedingly valuable to us" and 
/var/lib/asterisk/sounds/mike/agent-intro.wav to be something more 
specific like "Hi. My name is Mike and I specialize in sounding sincere 
even when I really don't care. I am extremely sorry you have to wait these 
brief moments until I am able to provide you with exceptional service 
today or tomorrow depending on the length of our call queue, you know?"


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi, agent intro-speech for outside caller

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart
Swedrowski
Sent: Thursday, January 20, 2011 6:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hi, agent intro-speech for outside caller

 

 

On 20 January 2011 11:54, Tom Rymes  wrote:

I don't know of a way to do that, but I can say that, as a caller, it is
highly annoying. Your agents ought to be able to do that themselves, no?

 

Exactly, otherwise you are losing first chance to make the call "different"
from the other ones where caller feel like they are talking with machines.
Simple "Hello, it's X, how is your day today sir" (and given it's a bit
different every day) can change they way the call is going to go...

 

All Asterisk prompts are configurable with a little legwork.  Simply use the
CLI to see what is playing at the point you want to change, then set up this
little ditty to override it.  Say you wanted to record the "canned"
tt-weasels prompt ("Weasels have eaten our phone system").  This 3-liner
lets you re-record it.

-  exten => 999,1,answer

-  exten => 999,n,record(tt-weasels.gsm)

-  exten => 999,n,hangup

 

If you are using a codec other than gsm you would replace gsm with wav,
slin, etc.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi, agent intro-speech for outside caller

[Bart Swedrowski]

On 20 January 2011 11:54, Tom Rymes  wrote:

> I don't know of a way to do that, but I can say that, as a caller, it is
> highly annoying. Your agents ought to be able to do that themselves, no?
>

Exactly, otherwise you are losing first chance to make the call "different"
from the other ones where caller feel like they are talking with machines.
 Simple "Hello, it's X, how is your day today sir" (and given it's a bit
different every day) can change they way the call is going to go...
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi, agent intro-speech for outside caller

[Tom Rymes]

On Jan 19, 2011, at 11:08 PM, DSR wrote:

> Is there anyway to play prerecorded agent intro-speech (like "Hello, my name 
> is ") to outside caller when agent picks up?

I don't know of a way to do that, but I can say that, as a caller, it is highly 
annoying. Your agents ought to be able to do that themselves, no?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Hi, agent intro-speech for outside caller

[DSR]

Hello,

I'm using AsteriskNow. Asterisk version is 1.6.2.15 and FreePBX 2.7.0.0

Is there anyway to play prerecorded agent intro-speech (like "Hello, my name
is ") to outside caller when agent picks up?

thank you
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] hi

[Beebob007]

Hi, the following configuration:

The number 0 will be forwarded to the Ring-Group 25 in which the numbers 
are 71 and 73. If you call the 0 so the office is ringing at the 71 and 
73 .

At the terminal stations are Snom 320.
In the evening the 71 to make call forwarding via web interface to the 
99 (voice mail).


Problem:
When I ring the phonenumber 0 after call forwarding .the 73 ringing 
and the Voice Mail (99) didn?t take the call ..
But when I call the 71 after call forwarding all works fine and the 
Voicemail take the Call



How can the Number 99 prioritization so that they take the Call in any 
case ... even if the 73 are still ringing?

Or there other options?

Regards
Beebob

in Deutsch: (hoffe das lesen hier welche die Deutsch sprechen...da das 
oben nur Googletranslate war)



Hallo, folgende Konfiguration:

Die Rufnummer 0 wird auf die Ring Group 25 weitergeleitet in der sich 
die Rufnummern 71 und 73 befinden. Ruft man also die 0 an über Amt 
klingelt es bei der 71 und 73..

An den Endstellen befinden sich Snom 320.
Abends soll die 71 per Weboberfläche eine Umleitung auf die 99 
(Voicemail) machen.


Problem:
Wenn ich die 0 nach erfolgter Rufumleitung anrufe klingelt die 73 munter 
vor sich hin und die Voicemail (99)geht nicht ran..
Rufe ich aber die 71 über Amt an funktioniert die Rufumleitung auf die 
99 wunderbar.



Wie kann ich die 99 bevorrechtigen, so dass sie auf jeden Fall 
rangeht...auch wenn die 73 noch klingelt bzw gibt es noch andere 
Möglichkeiten?


Gruß
Beebob
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Hi

[Rajkiran Reddy]

Hello everyone





---
Regards,
Raj.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[Ciprian ARSENIE]

Nothing :( only www.tainet.net have and i need partner account

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, February 26, 2010 9:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] hi

Try this link
http://www.epygi.com/forum/archive/index.php/f-14.html


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ciprian
ARSENIE
Sent: Friday, February 26, 2010 12:59 PM
To: '--[ UxBoD ]--'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: Re: [asterisk-users] hi

Anybody with a firmware with support SIP for a TAINET VENUS 2804 because I
do not find anywhere on the Internet. Thanks in advance










-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]--
Sent: Friday, February 26, 2010 8:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hi

- "C F"  wrote:

> Wouldn't some online translator do a better job? Or just plain old
> spell checking?
> 
> On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE 
> wrote:
> >
> >
> >
> >
> > If anyoane have a firmware with sip support for a tainet venus 2804
> please
> > give am feedback caz i kan-t find on internet
> >
Daca anyoane au o firmware cu gura sprijin pentru o tainet Venus 2804
oferiti sint ecoul caz I can-t gasesc pe Internet ;)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[Danny Nicholas]

Try this link
http://www.epygi.com/forum/archive/index.php/f-14.html


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ciprian
ARSENIE
Sent: Friday, February 26, 2010 12:59 PM
To: '--[ UxBoD ]--'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: Re: [asterisk-users] hi

Anybody with a firmware with support SIP for a TAINET VENUS 2804 because I
do not find anywhere on the Internet. Thanks in advance










-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]--
Sent: Friday, February 26, 2010 8:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hi

- "C F"  wrote:

> Wouldn't some online translator do a better job? Or just plain old
> spell checking?
> 
> On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE 
> wrote:
> >
> >
> >
> >
> > If anyoane have a firmware with sip support for a tainet venus 2804
> please
> > give am feedback caz i kan-t find on internet
> >
Daca anyoane au o firmware cu gura sprijin pentru o tainet Venus 2804
oferiti sint ecoul caz I can-t gasesc pe Internet ;)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[Ciprian ARSENIE]

Anybody with a firmware with support SIP for a TAINET VENUS 2804 because I
do not find anywhere on the Internet. Thanks in advance










-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]--
Sent: Friday, February 26, 2010 8:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hi

- "C F"  wrote:

> Wouldn't some online translator do a better job? Or just plain old
> spell checking?
> 
> On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE 
> wrote:
> >
> >
> >
> >
> > If anyoane have a firmware with sip support for a tainet venus 2804
> please
> > give am feedback caz i kan-t find on internet
> >
Daca anyoane au o firmware cu gura sprijin pentru o tainet Venus 2804
oferiti sint ecoul caz I can-t gasesc pe Internet ;)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[John Regal]

http://www.tainet.net/Product/venus.htm


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Friday, February 26, 2010 1:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hi

Wouldn't some online translator do a better job? Or just plain old
spell checking?

On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE 
wrote:
>
>
>
>
> If anyoane have a firmware with sip support for a tainet venus 2804 please
> give am feedback caz i kan-t find on internet
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[--[ UxBoD ]--]

- "C F"  wrote:

> Wouldn't some online translator do a better job? Or just plain old
> spell checking?
> 
> On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE 
> wrote:
> >
> >
> >
> >
> > If anyoane have a firmware with sip support for a tainet venus 2804
> please
> > give am feedback caz i kan-t find on internet
> >
Daca anyoane au o firmware cu gura sprijin pentru o tainet Venus 2804 oferiti 
sint ecoul caz I can-t gasesc pe Internet ;)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[C F]

Wouldn't some online translator do a better job? Or just plain old
spell checking?

On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE  wrote:
>
>
>
>
> If anyoane have a firmware with sip support for a tainet venus 2804 please
> give am feedback caz i kan-t find on internet
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] hi

[Ciprian ARSENIE]

 

 

If anyoane have a firmware with sip support for a tainet venus 2804 please
give am feedback caz i kan-t find on internet



smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi friend

[Alex Balashov]

Thanks.  In that case, do me a favour in return and start using the  
mailing list as it is intended, instead of mailing people privately.

--
Sent from mobile device

On Nov 14, 2009, at 1:58 PM,  wrote:

> hi alex done with music on hold. n  thanks a lot for the reply. u r  
> the only person who always reply to my query. from u only i am  
> learning

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[giancarlo lombardo]

Ciao,
try

/etc/asterisk/asterisk -vr

2009/11/8 Alex Balashov 

> Try /usr/sbin/asterisk.
>
> Also, copy the list.  Don't email me privately.
>
> aster...@opensourcesolution.in wrote:
>
> > hi friend,
> >
> > i gave that command which u told i.e asterisk -V. the output is below
> >
> >
> >
> > [r...@localhost ~]# cd /etc/
> > [r...@localhost etc]# asterisk -v
> > bash: asterisk: command not found
> > [r...@localhost etc]# asterisk -V
> > bash: asterisk: command not found
> >
> > thx
> >
>
>
> --
> Alex Balashov - Principal
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct  : (+1) (678) 954-0671
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Giancarlo Lombardo
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[Tzafrir Cohen]

Hi,

Un-top-posting,
> > aster...@opensourcesolution.in wrote:

On Sun, Nov 08, 2009 at 01:04:31PM +0100, giancarlo lombardo wrote:

> > > i gave that command which u told i.e asterisk -V. the output is below
> > >
> > > [r...@localhost ~]# cd /etc/
> > > [r...@localhost etc]# asterisk -v
> > > bash: asterisk: command not found
> > > [r...@localhost etc]# asterisk -V
> > > bash: asterisk: command not found
> try as below (in bold the command)
> 
> *[r...@dhcppc0 asterisk]# pwd
> /etc/asterisk*
> [r...@dhcppc0 asterisk]# *asterisk -vr*

My guess for the output of that command:

  bash: asterisk: command not found

But if I strictly followed your advice:

$ *asterisk -vr*
bash: *asterisk: command not found


If this is installed from source: 

  ./main/asterisk -V

in the source directory. Just in case it was not installed. In this
case, ask yourself why wasn't it installed.

What is the output of:

  echo $PATH

What Linux distributribution is it? If not Linux: What OS is it? What
version?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[giancarlo lombardo]

Ciao,
try as below (in bold the command)

*[r...@dhcppc0 asterisk]# pwd
/etc/asterisk*
[r...@dhcppc0 asterisk]# *asterisk -vr*
*Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <**marks...@digium.com* *>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=
Connected to Asterisk 1.4.24 currently running on dhcppc0 (pid = 2711)
Verbosity is at least 8
dhcppc0*CLI>*


2009/11/8 Alex Balashov 

> Try /usr/sbin/asterisk.
>
> Also, copy the list.  Don't email me privately.
>
> aster...@opensourcesolution.in wrote:
>
> > hi friend,
> >
> > i gave that command which u told i.e asterisk -V. the output is below
> >
> >
> >
> > [r...@localhost ~]# cd /etc/
> > [r...@localhost etc]# asterisk -v
> > bash: asterisk: command not found
> > [r...@localhost etc]# asterisk -V
> > bash: asterisk: command not found
> >
> > thx
> >
>
>
> --
> Alex Balashov - Principal
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct  : (+1) (678) 954-0671
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Giancarlo Lombardo
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[Alex Balashov]

Try /usr/sbin/asterisk.

Also, copy the list.  Don't email me privately.

aster...@opensourcesolution.in wrote:

> hi friend,
> 
> i gave that command which u told i.e asterisk -V. the output is below
> 
>  
> 
> [r...@localhost ~]# cd /etc/
> [r...@localhost etc]# asterisk -v
> bash: asterisk: command not found
> [r...@localhost etc]# asterisk -V
> bash: asterisk: command not found
> 
> thx
> 


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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[Mike Bessette]

Hello. To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

On 10/15/09, as asd  wrote:
> plz do not send for me e-mail
> thanks
>
>
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] hi

[as asd]

plz do not send for me e-mail
thanks


  ___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[Vinicius Neves]

Heya, how are you doing recently ? 
I would like to introduce you a very good company which I knew. Their company 
homepage is www.vanigo.com. They can offer you all kinds of electronical 
products which you need,  laptops, mobile phones, digial cameras, TV LCD,xbox, 
ps3, gps, MP3/4, etc. Please take some time to have a look at it, there must be 
something you 'd like to purchase.
Their contact E-mail: van...@188.com
MSN : van...@live.cn




Hope you have a good mood in shopping from their company!
Regards.
_
More than messages–check out the rest of the Windows Live™.
http://www.microsoft.com/windows/windowslive/___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi from argentina

[Valentin Bud]

Hello David,

 Welcome to the world of *. :)

a great day,
v
On Tue, Dec 2, 2008 at 10:40 PM, David fire <[EMAIL PROTECTED]> wrote:
> hi
> this is mi first email and just for say hello.
>
> David
>
> --
> (\__/)
> (='.'=)This is Bunny. Copy and paste bunny into your
> (")_(")signature to help him gain world domination.
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] hi from argentina

[David fire]

hi
this is mi first email and just for say hello.

David

-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] HI ~ good friend,

[Dean Collins]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Murphy
Sent: Saturday, 2 August 2008 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] HI ~ good friend,

On Fri, 2008-08-01 at 16:08 -0400, Dean Collins wrote:
> Yep I totally agree with you that documentation is an area digium is
> dropping the ball.
> 


>Mayhaps, but as one of the digium guys, I might add that Asterisk
>content is not solely supplied by Digium. Users out there, dissatisfied
>with documentation on various parts of Asterisk are more than welcome
>to help fill in the gaps! Join #asterisk-doc, get on the mailing list,
>put a sheet of paper in your typewriter (er, word processor), and 
>have at it!


Gasp no really I would never have seen that answer coming :P

 lol you guys are so predictable sometimes.



Cheers,
Dean

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] HI ~ good friend,

[Steve Murphy]

On Fri, 2008-08-01 at 16:08 -0400, Dean Collins wrote:
> Yep I totally agree with you that documentation is an area digium is
> dropping the ball.
> 


Mayhaps, but as one of the digium guys, I might add that Asterisk
content is not solely supplied by Digium. Users out there, dissatisfied
with documentation on various parts of Asterisk are more than welcome
to help fill in the gaps! Join #asterisk-doc, get on the mailing list,
put a sheet of paper in your typewriter (er, word processor), and 
have at it!

As to dimensioning, I might add, that to do justice, and create
metrics, is a tremendous task. Factors that would affect the 
calls/sec and concurrent calls numbers would be:

1. The drivers involved
2. the codecs in use
3. the hardware used (digium vs Sangoma, etc)
4. the CPU speed
5. the memory speed
6. the memory amount
7. the size of CPU caches
8. bus speeds
9. disk speeds
10. network effective bandwidth
11. Call logging (to console & CDR backends)
12. Asterisk software version
13. Hardware (dahdi, etc) driver versions
...
and so on.. I cannot even begin to enumerate all the factors.

(And, on top of the above, I'm willing to bet that for each
factor, the speed will probably NOT be a simple linear relationship)

Now, the Cisco guys and others can chop the list down because they
can set the hardware the software runs on. They can run a few
tests and come up with sets of metrics that give you an idea about
how fast the darn thing is.

But we just do the software. Digium sells cards, but they run
in a very widely different range of machines.

Coming up with a formula that you can plug numbers into would
be a big, expensive task, and by the time it was produced, it
would be wrong because of software fixes, speedups, slowdowns,
etc.

No, really, the only practical approach is for a user to freeze
all the factors he can, like software version and most of the
hardware numbers, and run load tests to see how well things 
work out. In the end, the individual implementor is responsible
to know what his implementation can do. If the combo doesn't
meet his needs, he can get a faster cpu, more mem, etc.

murf


> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Al Baker
> Sent: Friday, 1 August 2008 3:51 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] HI ~ good friend,
> 
> I must disagree.
> "Dimensioning" of Asterisk is a very sorely lacking area and is one of 
> the main area CISCO
> and such eats its lunch. There simply no a base of solid metric that 
> allow for true "provisioning" .
> Yes, there are INVALUABLE anecdotal reports from people who have been 
> kind, and sharing of their
> experiences and for which are all very very grateful.
> BUT
> That that just is not the same as as solid, vendor based Metrics.
> Can you imagine calling and asking DISCO, "What do I need for 400 calls"
> 
> an their answer is
> "Here please go read these mostly outdated anecdotal reports and call 
> back with your order"
> Sorry. I love *, but this  area of it is not where it needs to be.
> 
> Dean Collins wrote:
> >
> > Hi welcome to the asterisk community.
> >
> >  
> >
> > The answer you want are here; 
> > http://www.voip-info.org/wiki/view/Asterisk+dimensioning
> >
> >  
> >
> > The short answer is; Pretty much yes, depending on hardware and 
> > horizontal scaling with multiple servers sharing the load.
> >
> >  
> >
> >
> > Cheers,
> >
> > Dean
> >
> >
> --------
> >
> > *From:* [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] *On Behalf Of *???
> > *Sent:* Friday, 1 August 2008 9:43 AM
> > *To:* asterisk-users@lists.digium.com
> > *Subject:* [asterisk-users] HI ~ good friend,
> >
> >  
> >
> > hi ~ nice to meet you, i just join here, today,
> >
> >  
> >
> > i am a student, and i am very interesting in asterisk.
> >
> >  
> >
> > and i have a IP-PBX server, made by me with my friend,
> >
> >  
> >
> > while when i studying, i have a question,
> >
> >  
> >
> > is there any limit users for asterisk?
> >
> >  
> >
> > ex) registed users number is 1000 or 1 or 10 like that, is 
> > that possible?
> >
> >  
> >
> > and how about the concurrent calls? 1000 concurrent calls is possible?
> 
> > or 2000 concurrent calls?
> >
> >  
> >
> > my PBX server's user is just less then 15, almost my friends,
> >
> >  
> >
> > so

Re: [asterisk-users] HI ~ good friend,

[Tzafrir Cohen]

On Fri, Aug 01, 2008 at 03:51:21PM -0400, Al Baker wrote:
> I must disagree.
> "Dimensioning" of Asterisk is a very sorely lacking area and is one of 
> the main area CISCO
> and such eats its lunch. There simply no a base of solid metric that 
> allow for true "provisioning" .
> Yes, there are INVALUABLE anecdotal reports from people who have been 
> kind, and sharing of their
> experiences and for which are all very very grateful.
> BUT
> That that just is not the same as as solid, vendor based Metrics.
> Can you imagine calling and asking DISCO, "What do I need for 400 calls" 
> an their answer is
> "Here please go read these mostly outdated anecdotal reports and call 
> back with your order"
> Sorry. I love *, but this  area of it is not where it needs to be.

I'll byte.

Al, What do you need for "400 calls" on Asterisk?

Please give me a short and clear answer or I won't buy Asterisk from
you.

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] HI ~ good friend,

[Dean Collins]

Yep I totally agree with you that documentation is an area digium is
dropping the ball.


Cheers,

Dean



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Al Baker
Sent: Friday, 1 August 2008 3:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] HI ~ good friend,

I must disagree.
"Dimensioning" of Asterisk is a very sorely lacking area and is one of 
the main area CISCO
and such eats its lunch. There simply no a base of solid metric that 
allow for true "provisioning" .
Yes, there are INVALUABLE anecdotal reports from people who have been 
kind, and sharing of their
experiences and for which are all very very grateful.
BUT
That that just is not the same as as solid, vendor based Metrics.
Can you imagine calling and asking DISCO, "What do I need for 400 calls"

an their answer is
"Here please go read these mostly outdated anecdotal reports and call 
back with your order"
Sorry. I love *, but this  area of it is not where it needs to be.

Dean Collins wrote:
>
> Hi welcome to the asterisk community.
>
>  
>
> The answer you want are here; 
> http://www.voip-info.org/wiki/view/Asterisk+dimensioning
>
>  
>
> The short answer is; Pretty much yes, depending on hardware and 
> horizontal scaling with multiple servers sharing the load.
>
>  
>
>
> Cheers,
>
> Dean
>
>

>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *???
> *Sent:* Friday, 1 August 2008 9:43 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] HI ~ good friend,
>
>  
>
> hi ~ nice to meet you, i just join here, today,
>
>  
>
> i am a student, and i am very interesting in asterisk.
>
>  
>
> and i have a IP-PBX server, made by me with my friend,
>
>  
>
> while when i studying, i have a question,
>
>  
>
> is there any limit users for asterisk?
>
>  
>
> ex) registed users number is 1000 or 1 or 10 like that, is 
> that possible?
>
>  
>
> and how about the concurrent calls? 1000 concurrent calls is possible?

> or 2000 concurrent calls?
>
>  
>
> my PBX server's user is just less then 15, almost my friends,
>
>  
>
> so, i can't test, over 10 users and 1000 concurrent calls,
>
>  
>
> please tell me, it is possible or not?
>
>  
>
> thanks your permission to join there,
>
>
>
>
>

>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] HI ~ good friend,

[Al Baker]

I must disagree.
"Dimensioning" of Asterisk is a very sorely lacking area and is one of 
the main area CISCO
and such eats its lunch. There simply no a base of solid metric that 
allow for true "provisioning" .
Yes, there are INVALUABLE anecdotal reports from people who have been 
kind, and sharing of their
experiences and for which are all very very grateful.
BUT
That that just is not the same as as solid, vendor based Metrics.
Can you imagine calling and asking DISCO, "What do I need for 400 calls" 
an their answer is
"Here please go read these mostly outdated anecdotal reports and call 
back with your order"
Sorry. I love *, but this  area of it is not where it needs to be.

Dean Collins wrote:
>
> Hi welcome to the asterisk community.
>
>  
>
> The answer you want are here; 
> http://www.voip-info.org/wiki/view/Asterisk+dimensioning
>
>  
>
> The short answer is; Pretty much yes, depending on hardware and 
> horizontal scaling with multiple servers sharing the load.
>
>  
>
>
> Cheers,
>
> Dean
>
> 
>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *???
> *Sent:* Friday, 1 August 2008 9:43 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] HI ~ good friend,
>
>  
>
> hi ~ nice to meet you, i just join here, today,
>
>  
>
> i am a student, and i am very interesting in asterisk.
>
>  
>
> and i have a IP-PBX server, made by me with my friend,
>
>  
>
> while when i studying, i have a question,
>
>  
>
> is there any limit users for asterisk?
>
>  
>
> ex) registed users number is 1000 or 1 or 10 like that, is 
> that possible?
>
>  
>
> and how about the concurrent calls? 1000 concurrent calls is possible? 
> or 2000 concurrent calls?
>
>  
>
> my PBX server's user is just less then 15, almost my friends,
>
>  
>
> so, i can't test, over 10 users and 1000 concurrent calls,
>
>  
>
> please tell me, it is possible or not?
>
>  
>
> thanks your permission to join there,
>
>
>
>
> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] HI ~ good friend,

[Dean Collins]

Hi welcome to the asterisk community.

 

The answer you want are here;
http://www.voip-info.org/wiki/view/Asterisk+dimensioning 

 

The short answer is; Pretty much yes, depending on hardware and
horizontal scaling with multiple servers sharing the load.

 


Cheers,

Dean



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ???
Sent: Friday, 1 August 2008 9:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] HI ~ good friend,

 

hi ~ nice to meet you, i just join here, today,

 

i am a student, and i am very interesting in asterisk. 

 

and i have a IP-PBX server, made by me with my friend,

 

while when i studying, i have a question, 

 

is there any limit users for asterisk?

 

ex) registed users number is 1000 or 1 or 10 like that, is that
possible?

 

and how about the concurrent calls? 1000 concurrent calls is possible?
or 2000 concurrent calls? 

 

my PBX server's user is just less then 15, almost my friends, 

 

so, i can't test, over 10 users and 1000 concurrent calls,

 

please tell me, it is possible or not? 

 

thanks your permission to join there, 



 <http://mail.nate.com/web/footer/img/jem.gif>  
 <http://mail.nate.com/web/footer/img/logo_nate_20060420.gif>   
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] HI ~ good friend,

[김수환]

hi ~ nice to meet you, i just join here, today,
 
i am a student, and i am very interesting in asterisk. 
 
and i have a IP-PBX server, made by me with my friend,
 
while when i studying, i have a question, 
 
is there any limit users for asterisk?
 
ex) registed users number is 1000 or 1 or 10 like that, is that 
possible?
 
and how about the concurrent calls? 1000 concurrent calls is possible? or 
2000 concurrent calls? 
 
my PBX server's user is just less then 15, almost my friends, 
 
so, i can't test, over 10 users and 1000 concurrent calls,
 
please tell me, it is possible or not? 
 
thanks your permission to join there, 







	
		
			



			
		
	

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[Mojo with Horan & Company, LLC]

Is it behind a router? either forward the necessary ports to the sip 
phone's internal network ip address using the router, or move the phone 
outside the router to get it an external network (global ip) ;)

Mojo

sandeep.s wrote:
> Hi,
> my sip phone is unreachable for external network(global ip)
>
>
> Thanks,
> sandeep.s
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[Luis Carlos Martos Ratías]

Que has comio?
Que has comio?

(Its a joke in spanish-chanante)

2007/12/11, CunningPike <[EMAIL PROTECTED]>:
>
> Sounds like good security practice to me. YMMV.
>
> CP
>
> sandeep.s wrote:
> > Hi,
> > my sip phone is unreachable for external network(global ip)
> >
> >
> > Thanks,
> > sandeep.s
> >
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[CunningPike]

Sounds like good security practice to me. YMMV.

CP

sandeep.s wrote:
> Hi,
> my sip phone is unreachable for external network(global ip)
> 
> 
> Thanks,
> sandeep.s
> 

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[Alex Balashov]

Happens.

On Tue, 11 Dec 2007, sandeep.s wrote:

>
> Hi,
> my sip phone is unreachable for external network(global ip)
>
>
> Thanks,
> sandeep.s
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] hi

[Philipp Kempgen]

Hi :)

sandeep.s wrote:

> my sip phone is unreachable for external network(global ip)

Some of my SIP phones are unreachable as well. That's because
I unplugged the power cord.

You need to provide a bit more information or else the solution
may be described like "make sure everything is set up correctly
to make it work". SCNR

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] hi

[sandeep.s]

Hi,
my sip phone is unreachable for external network(global ip)


Thanks,
sandeep.s

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi ability solution

[Patrick]

On Mon, 2007-06-25 at 12:19 +0200, voip crazy wrote:
> Hi all,
> 
> On one of our client, I must to install an asterisk over a hi ability
> cluster. I have no experience with clusters an linux neither asterisk.
> Someone has installed an asterisk in a hi-ability enbviroment?
> How do you install the cluster? 
> Witch solution did you use?
> Witch is the best cluster solution to use with asterisk?

Have a look here for an example architecture:
http://integrics.com/products/enswitch/guides/latest/en/system/architecture/

And Steve Totaro already answered:
http://www.google.com/search?q=high+availability+asterisk&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:official
http://www.rocksclusters.org/wordpress/
http://linux-ha.org

And Senad Jordanovic already gave you this link:
http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf

Regards,
Patrick


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi ability solution

[Senad Jordanovic]

Steve Totaro wrote:
> Senad Jordanovic wrote:
>>> Any High availiability solution for asterisk?
>>> VoipCrazy
>>> 
>> 
>> 
>> http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf
>> 
> Is this free?
> 
> Benefits over opensource packages?
> 
> Thanks,
> Steve Totaro


Steve...  (and anyone else)I made a mistake replying to asterisk-users
list thinking it is asterisk-biz list.
Anything else please contact me of this list.

Thanks

Senad



___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi ability solution

[Steve Totaro]

Senad Jordanovic wrote:
>> Any High availiability solution for asterisk?
>> VoipCrazy
>> 
>
>  
> http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf
>   
Is this free?

Benefits over opensource packages?

Thanks,
Steve Totaro

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi ability solution

[Senad Jordanovic]

>Any High availiability solution for asterisk?
>VoipCrazy

 
http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi ability solution

[Steve Totaro]

There is a whole wiki page on the subject.  Google is your friend.
http://www.google.com/search?q=high+availability+asterisk&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:official


This is what I am currently playing with:
http://linux-ha.org

voip crazy wrote:
> I would say High Availability,
>
> sorry for my english.
>
> Any High availiability solution for asterisk?
>
> VoipCrazy
>
>
> 2007/6/25, Steve Totaro < [EMAIL PROTECTED] 
> >:
>
> voip crazy wrote:
> > Hi all,
> >
> > On one of our client, I must to install an asterisk over a hi
> ability
> > cluster. I have no experience with clusters an linux neither
> asterisk.
> > Someone has installed an asterisk in a hi-ability enbviroment?
> > How do you install the cluster?
> > Witch solution did you use?
> > Witch is the best cluster solution to use with asterisk?
> >
> > Thanks in advance,
> >
> > Voipcrazy
> >
> Do you mean "High Ability", or "High Availability"
>
> I think Rocks is pretty good but I just started playing with it.  I
> think it is more of a "High Ability" thing.
> http://www.rocksclusters.org/wordpress/
>
> Thanks,
> Steve Totaro
>
>
>
>


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi ability solution

[voip crazy]

I would say High Availability,

sorry for my english.

Any High availiability solution for asterisk?

VoipCrazy


2007/6/25, Steve Totaro <[EMAIL PROTECTED]>:


voip crazy wrote:
> Hi all,
>
> On one of our client, I must to install an asterisk over a hi ability
> cluster. I have no experience with clusters an linux neither asterisk.
> Someone has installed an asterisk in a hi-ability enbviroment?
> How do you install the cluster?
> Witch solution did you use?
> Witch is the best cluster solution to use with asterisk?
>
> Thanks in advance,
>
> Voipcrazy
>
Do you mean "High Ability", or "High Availability"

I think Rocks is pretty good but I just started playing with it.  I
think it is more of a "High Ability" thing.
http://www.rocksclusters.org/wordpress/

Thanks,
Steve Totaro




___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi ability solution

[Steve Totaro]

voip crazy wrote:
> Hi all,
>
> On one of our client, I must to install an asterisk over a hi ability 
> cluster. I have no experience with clusters an linux neither asterisk.
> Someone has installed an asterisk in a hi-ability enbviroment?
> How do you install the cluster?
> Witch solution did you use?
> Witch is the best cluster solution to use with asterisk?
>
> Thanks in advance,
>
> Voipcrazy
>
Do you mean "High Ability", or "High Availability"

I think Rocks is pretty good but I just started playing with it.  I 
think it is more of a "High Ability" thing.  
http://www.rocksclusters.org/wordpress/

Thanks,
Steve Totaro




___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Hi ability solution

[voip crazy]

Hi all,

On one of our client, I must to install an asterisk over a hi ability
cluster. I have no experience with clusters an linux neither asterisk.
Someone has installed an asterisk in a hi-ability enbviroment?
How do you install the cluster?
Witch solution did you use?
Witch is the best cluster solution to use with asterisk?

Thanks in advance,

Voipcrazy
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi Honies! I'm home!

[Al]

i couldnt agree more with Brian,
i'm sure we'll see more improvement in code and more improvement in asterisk 
business edition.
Al

=
I was wondering when this would happen. A lot of successful and prospering
open source company like yours seems to do this.

Much like Google did.   Once a company has grown to a point  it's more
valuable to have someone focus on the business from a businessmans
perspective working with the monies, departments, board of directors and
strategies while letting the previous guru (Mark) focus on what you always
really have needed to, the code and the product line.

It looks like Danny has a solid background and strong roles of leadership
from adtran.  I love this decision.

Go team Digium.

Brian

On 1/30/07, Mark Spencer <[EMAIL PROTECTED]> wrote:
>
> Many of you may have seen the recent announcement about Danny Windham
> coming on as the new CEO of Digium.  This is one of the most exciting
> things to happen to Digium and to Asterisk at large.  When Danny comes on
> board, I will be transitioning to the role of Chief Technical Officer
> (retaining my position of chairman of the board of directors), providing
> strategic vision for the company as well as being able to focus more
> extensively on the community, the customers and the technology.
>
> My sincere hope is that this transition will not only directly benefit the
> Asterisk community and Digium customers, but will allow me to spend much
> more time with the community and with Asterisk, playing a more important
> technical role in our roadmap for both hardware and software.
>
> I'm looking forward to working more with the community and the developers
> to help grow the future of Asterisk even more!
>
> Mark
> ___
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi Honies! I'm home!

[Chris Mason]

Your dinner's in the oven.

--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi Honies! I'm home!

[Brian McManus]

I was wondering when this would happen. A lot of successful and prospering
open source company like yours seems to do this.

Much like Google did.   Once a company has grown to a point  it's more
valuable to have someone focus on the business from a businessmans
perspective working with the monies, departments, board of directors and
strategies while letting the previous guru (Mark) focus on what you always
really have needed to, the code and the product line.

It looks like Danny has a solid background and strong roles of leadership
from adtran.  I love this decision.

Go team Digium.

Brian

On 1/30/07, Mark Spencer <[EMAIL PROTECTED]> wrote:


Many of you may have seen the recent announcement about Danny Windham
coming on as the new CEO of Digium.  This is one of the most exciting
things to happen to Digium and to Asterisk at large.  When Danny comes on
board, I will be transitioning to the role of Chief Technical Officer
(retaining my position of chairman of the board of directors), providing
strategic vision for the company as well as being able to focus more
extensively on the community, the customers and the technology.

My sincere hope is that this transition will not only directly benefit the
Asterisk community and Digium customers, but will allow me to spend much
more time with the community and with Asterisk, playing a more important
technical role in our roadmap for both hardware and software.

I'm looking forward to working more with the community and the developers
to help grow the future of Asterisk even more!

Mark
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users





--
--Brian McManus
--(208) 329-0818
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Hi Honies! I'm home!

[Mark Spencer]

Many of you may have seen the recent announcement about Danny Windham 
coming on as the new CEO of Digium.  This is one of the most exciting 
things to happen to Digium and to Asterisk at large.  When Danny comes on 
board, I will be transitioning to the role of Chief Technical Officer 
(retaining my position of chairman of the board of directors), providing 
strategic vision for the company as well as being able to focus more 
extensively on the community, the customers and the technology.


My sincere hope is that this transition will not only directly benefit the 
Asterisk community and Digium customers, but will allow me to spend much 
more time with the community and with Asterisk, playing a more important 
technical role in our roadmap for both hardware and software.


I'm looking forward to working more with the community and the developers 
to help grow the future of Asterisk even more!


Mark
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi reg. asterisk Compilation

[Tzafrir Cohen]

On Thu, Jan 04, 2007 at 11:55:44AM -0500, Noah Miller wrote:
> Hi Thiru -
> 
> > Hi i need to done some  modify/changes in one of the asterisk c source 
> > code
> >,,eg: app_meetme.c
> > How can i compile and debug it without compile the whole module of
> >asterisk...
> 
> I would not recommend trying to compile only small pieces of asterisk.
> It is broken up into many sections, and many of these sections depend
> on other sections, and won't compile correctly when compiled alone.
> 
> Nonetheless, if you do want to try to compile just app_meetme.c, it is
> in the "apps" directory of the source code.  For example, that may be
> here:
> 
> /usr/local/src/asterisk-1.2.14/apps
> 
> You'll probably have to compile asterisk as a whole first, 

./configure
make

mind you, you need to have a recent (1.4) version of Zaptel availble.
Otherwise the configure script will consider zaptel not availble and not
build meetme.

> and then
> figure out a gcc statement that links with all the other asterisk
> components to compile just app_meetme.c.

make

Asterisk's build system does not support running make from a
subdirectory or for specific targets of subdirectories.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi reg. asterisk Compilation

[Noah Miller]

Hi Thiru -


 Hi i need to done some  modify/changes in one of the asterisk c source code
,,eg: app_meetme.c
 How can i compile and debug it without compile the whole module of
asterisk...


I would not recommend trying to compile only small pieces of asterisk.
It is broken up into many sections, and many of these sections depend
on other sections, and won't compile correctly when compiled alone.

Nonetheless, if you do want to try to compile just app_meetme.c, it is
in the "apps" directory of the source code.  For example, that may be
here:

/usr/local/src/asterisk-1.2.14/apps

You'll probably have to compile asterisk as a whole first, and then
figure out a gcc statement that links with all the other asterisk
components to compile just app_meetme.c.

Unless you're on a really slow machine, I think it would probably be
easier to just recompile all of asterisk.



 and also let me know which editor suitable it ,I'm using suse linux..
 Plz help me reg. this ..


That's a religious question.  Everybody has their favorite editor.  I
like gnu's nano, but there's lots of other popular editors: emacs, vi,
gedit, etc.


- Noah
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Hi reg. asterisk Compilation

[Thirumal Saminathan]

Hi moises,
Hi i need to done some  modify/changes in one of the asterisk c source code
,,eg: app_meetme.c
How can i compile and debug it without compile the whole module of
asterisk...
and also let me know which editor suitable it ,I'm using suse linux..
Plz help me reg. this ..


-nsthi,
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] hi guys, a new newbie here needing help :D

[pedro noticioso]

I just installed rpm binaries in a new mandriva and I
see a frew error messages with asterisk -vvvcfg,
btw I would also like a little guidance to just set up
a couple sip phones to start playing with soft phone
communication with 3 pcs on the network

thanks :)


ng '/etc/asterisk/agents.conf': Found
 [skipping chan_alsa.so]
 [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
  == Registered custom function IAXPEER
May 12 15:50:12 WARNING[6173]: chan_iax2.c:9212
load_module: Unable to open IAX timing interface: No
such file or directory
  == Registered application 'IAX2Provision'
  == Manager registered action IAXpeers
  == Manager registered action IAXnetstats
  == Parsing '/etc/asterisk/iax.conf': Found
-- doing lookup for '216.207.245.47'
  == Registered channel type 'IAX2' (Inter Asterisk
eXchange Driver (Ver 2))
  == Using TOS bits 16
  == Binding IAX2 to default address 0.0.0.0:4569
  == IAX Ready and Listening
  == Loaded firmware 'iaxy.bin'
  == Parsing '/etc/asterisk/iaxprov.conf': Found
-- Loaded provisioning template 'default'
 [chan_local.so] => (Local Proxy Channel)
  == Registered channel type 'Local' (Local Proxy
Channel Driver)
 [chan_mgcp.so] => (Media Gateway Control Protocol
(MGCP))
  == Parsing '/etc/asterisk/mgcp.conf






ng '/etc/asterisk/agents.conf': Found
 [skipping chan_alsa.so]
 [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
  == Registered custom function IAXPEER
May 12 15:50:12 WARNING[6173]: chan_iax2.c:9212
load_module: Unable to open IAX timing interface: No
such file or directory
  == Registered application 'IAX2Provision'
  == Manager registered action IAXpeers
  == Manager registered action IAXnetstats
  == Parsing '/etc/asterisk/iax.conf': Found
-- doing lookup for '216.207.245.47'
  == Registered channel type 'IAX2' (Inter Asterisk
eXchange Driver (Ver 2))
  == Using TOS bits 16
  == Binding IAX2 to default address 0.0.0.0:4569
  == IAX Ready and Listening
  == Loaded firmware 'iaxy.bin'
  == Parsing '/etc/asterisk/iaxprov.conf': Found
-- Loaded provisioning template 'default'
 [chan_local.so] => (Local Proxy Channel)
  == Registered channel type 'Local' (Local Proxy
Channel Driver)
 [chan_mgcp.so] => (Media Gateway Control Protocol
(MGCP))
  == Parsing '/etc/asterisk/mgcp.conf


 [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec
Translator)
  == Parsing '/etc/asterisk/codecs.conf': Found
-- codec_gsm: using generic PLC
  == Registered translator 'gsmtolin' from format gsm
to slin, cost 1
May 12 15:50:34 WARNING[6173]: config_old.c:28
ast_load: ast_load is deprecated, use ast_config_load
instead!
  == Parsing '/etc/asterisk/rpt.conf': Found
  == Registered translator 'lintogsm' from format slin
to gsm, cost 3

__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Hi...Please help me

[Brian C. Fertig]

Chandra, 

 

In all honesty if they are proprietary and
you want to use them you will need a FXO card.  Alternatively there are

a few good termination providers out there
that are inexpensive. 

 

The top 3 most inexpensive that come to
mind are: 

 

Plainvoip  -   http://www.plainvoip.com Domestic
starting at 1.1c

VoipJet    -   http://www.voipjet.com    Domestic
starting at 1.3c

NuFone   -   http://www.nufone.net  Domestic
starting at 2c (I believe)

 

 

Anyone of these providers can supply you
with USA
and also international dialing.

 



_.._
Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE
Data/Telecom Engineer
IT Administrator
Planet Telecom, Inc
Tampa, FL
Office
o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 
SIP URI: [EMAIL PROTECTED]











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Monday, May 08, 2006 8:43 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Hi...Please help me



 

Hi Friends,

Thank you for your quick response. I have successfully implemented Intercom
(Dialling within my office LAN) using Asterisk. To implement this, I am using
X-Lite Softphone. 

Now, I want to make calls to US using VoIP Asterisk. 

I have registered with Vebtel (VoIP IP Telephony Service provider). They had
given me one VoIP Modem called "Voice Finder AP 200" and the below
values:

Inbound Number: 123456789
Public IP Number:
55.23.789.145
Password: xyz

(These values are dummy values)

Currently we are making US calls using VoIP provided by "Vebtel".
Now, I want to make US calls using this Vebtel service from Asterisk. How can I
do this?

I am unable to understand where to give above mentioned values? What
configuration files I should use to implement this using the Vebtel SIP
provider? Do I need to provide any more values to implement this using Asterisk
from Vebtel?

Waiting for your quick response. Thank you.  

Regards,
Chandra.







Yahoo! Messenger with Voice. Make
PC-to-Phone Calls to the US
(and 30+ countries) for 2¢/min or less.





This email was scanned by:  Mcafee GroupShield
 CONFIDENTIAL DISCLAMER 
All information provided in this email is considered confidential
and proprietary of Planet Telecom, Inc. and Telecenter Inc.
Use of this information by anyone other than the recipient or 
sender will be considered in breach of agreement.




___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Hi...Please help me

[Crazy Boy]

Hi Friends,  Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone.   Now, I want to make calls to US using VoIP Asterisk.   I have registered with Vebtel (VoIP IP Telephony Service provider). They had given me one VoIP Modem called "Voice Finder AP 200" and the below values:  Inbound Number: 123456789 Public IP Number: 55.23.789.145 Password: xyz  (These values are dummy values)  Currently we are making US calls using VoIP provided by "Vebtel". Now, I want to make US calls using this Vebtel service from Asterisk. How can I do this?  I am unable to understand where to give above mentioned values? What configuration files I should use to implement this using the Vebtel SIP provider? Do I need to provide any more values to implement this using Asterisk from Vebtel?  Waiting for your quick response. Thank you.    Regards, Chandra. 
		Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Hi...Please help me

[William Piper]

Wouldn't it be easier to replace the callername to the exten.

example:

exten => _x.,1,SetCallerIDname(${EXTEN})
exten => _x.,2,SetCallerIDnum(${CALLERIDNUM})
exten => _x.,3,dial,SIP/number

That way, the Caller Name would show the extension it is ringing and the
callerid will still show the calling party.

Now you don't need the softphone to do it... just a phone with callerid.

bp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, May 02, 2006 5:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Hi...Please help me

On Tuesday 02 May 2006 16:42, hugolivude wrote:
> We share SIP phones at the office in a 1:4 ratio.  You're probably
> asking - how do you know when a ringing phone is for you?  Well,
> everyone in our office gets an XLite softphone, and I direct calls to
> make BOTH the SIP phone AND the XLite ring.  If your XLite pops up,
> you know that ring phone is for you.

That seems to be humongous overkill... why not just use any of the caller ID

popup apps instead of running that behemoth X-Lite?  If the popup comes up, 
the phone's for you.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


__ NOD32 1.1443 (20060314) Information __

This message was checked by NOD32 antivirus system.
http://www.eset.com


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi...Please help me

[Andrew Kohlsmith]

On Tuesday 02 May 2006 16:42, hugolivude wrote:
> We share SIP phones at the office in a 1:4 ratio.  You're probably
> asking – how do you know when a ringing phone is for you?  Well,
> everyone in our office gets an XLite softphone, and I direct calls to
> make BOTH the SIP phone AND the XLite ring.  If your XLite pops up,
> you know that ring phone is for you…

That seems to be humongous overkill... why not just use any of the caller ID 
popup apps instead of running that behemoth X-Lite?  If the popup comes up, 
the phone's for you.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi...Please help me

[hugolivude]

First off, I agree w/ Gonzalo – softphones didn't work out for me
either.  One thing that did work great tho was a combo.

We share SIP phones at the office in a 1:4 ratio.  You're probably
asking – how do you know when a ringing phone is for you?  Well,
everyone in our office gets an XLite softphone, and I direct calls to
make BOTH the SIP phone AND the XLite ring.  If your XLite pops up,
you know that ring phone is for you…

Here's some answers to your other questions

•   What I have to install in client PC's?

Just the softphone client (e.g. XLite (SIP) Cubix (IAX)
http://www.virbiage.com/cubix.php

•   What hardware I need?

Nothing too fancy.  Your PCs seem OK.  For Asterisk, I'm using an old
Pentium 4 beater with 1Gig memory and it handles the whole office (19)
just fine.

•   How can I take decission to buy extra hardware (like Zaptel
products) OR no need of buying extra hardware? ( I will be using
Asterisk for 70 PC's and a server)

This depends on what you want in the way of handsets, and what kind of
connectivity you want to the PSTN (Public Switched Telephone Network).
You could get away with no extra hardware in a pure VoIP solution. 
Connect Asterisk to the Internet w/ an Ethernet cable and use SIP

based phones that also communicate over a network.  Note that if you
don't use any Digium hardware, I believe that you need to use ztdummy
to control timing (never used it myself)
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy

•   Is it sufficient to buy hardware for server only OR for client PC's 
also?

Again, your PCs seem OK.  How you kit out your server depends upon
what you want.

•   How can I connect my VoIP phone to server?

Once you have Asterisk installed, you have to configure your VoIP
phone to register with it.  For example, look here for how to
configure Polycom Soundpoint 501s -
http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501.
You'll also have to have the appropriate entries in SIP.conf for the
phone AND to connect to your VoIP service provider
http://www.voip-info.org/wiki-Asterisk+config+sip.conf

•   How can I connect hardware to server?

Don't understand this one.  If you use telephony boards, you'll need
drivers.  Depending upon the board you may also have to physically
connect your phone to it with a telephone wire (as is the case with
TDM boards for example)

•   How can I connect PSTN line to server PC?

Assuming analogue phones you'll need a TDM card with an FXO port
(outgoing) for each line you have
(http://www.digium.com/en/products/hardware/analogcards.php).  You'll
also need an FXS port for each phone you have on your TDM card as
well.

Yours,
H

On 5/2/06, William Piper <[EMAIL PROTECTED]> wrote:




You are missing the dtmf mode, and most importantly… the codec to be used.

I would also add the nat=yes, that is probably why your phone isn't
registering.



See below for example config:




[chandra]

type=friend

username=chandra

secret=chandra


nat=yes

host=dynamic

dtmfmode=rfc2833

disallow=all

allow=ulaw

allow=g729

context=tutorial

canreinvite=no



 


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Crazy Boy
 Sent: Tuesday, May 02, 2006 8:58 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
 Subject: RE: [Asterisk-Users] Hi...Please help me




Hi friends,

 Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6
version. I have installed Asterisk in my PC and "X-Lite" as softphone in my
PC and client PC. Here my user name is "chandra" and client user name is
"aarti". I have added these lines to configuration files at the end of file.

 added contents in sip.conf:

 [aarti]
 type=friend
 username=aarti
 secret=aarti
 host=dynamic
 context=tutorial

 [chandra]
 type=friend
 username=chandra
 secret=chandra
 host=dynamic
 context=tutorial

 added contents in extensions.conf:

 [tutorial]
 exten => 101,1,Dial(SIP/aarti)
 exten => 102,1,Dial(SIP/chandra)

 Here, "aarti" is client, "chandra" is mine and Asterisk is also installed
in my PC (chandra) and it is successfully connected to Asterisk server using
"X-Lite" softphone.

 But, when i try to connect from "aarti" system using softphone, it displays
an error message "login timedout, contact system admin".

 Is there any problem with the content of sip.conf file or extensions.conf
file? I have not connected any external hardware to my pc. I just want to
connect Asterisk server to my collegues PC's like Intercom within my office
LAN using headphones. How can I do this? Please tell me. Looking forward for
your response.

 Thank you.

 Regards,
 Chandra.



 Evalyn Wafula <[EMAIL PROTECTED]> wrote:

Hi Chandra, I am also new to Asterisk and I have only just started
installing a test syste

RE: [Asterisk-Users] Hi...Please help me

[William Piper]

You are missing the dtmf mode, and most importantly…
the codec to be used.  

I would also add the nat=yes, that is probably why your
phone isn’t registering.

 

See below for example config:

 

[chandra]

type=friend

username=chandra

secret=chandra

nat=yes

host=dynamic

dtmfmode=rfc2833

disallow=all

allow=ulaw

allow=g729

context=tutorial

canreinvite=no

 









From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Crazy Boy
Sent: Tuesday, May 02, 2006 8:58
AM
To: [EMAIL PROTECTED]; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Hi...Please help me



 

Hi friends,

Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version.
I have installed Asterisk in my PC and "X-Lite" as softphone in my PC
and client PC. Here my user name is "chandra" and client user name is
"aarti". I have added these lines to configuration files at the end
of file.

added contents in sip.conf:

[aarti]
type=friend
username=aarti
secret=aarti
host=dynamic
context=tutorial

[chandra]
type=friend
username=chandra
secret=chandra
host=dynamic
context=tutorial

added contents in extensions.conf:

[tutorial]
exten => 101,1,Dial(SIP/aarti)
exten => 102,1,Dial(SIP/chandra)

Here, "aarti" is client, "chandra" is mine and Asterisk is
also installed in my PC (chandra) and it is successfully connected to Asterisk
server using "X-Lite" softphone. 

But, when i try to connect from "aarti" system using softphone, it
displays an error message "login timedout, contact system admin". 

Is there any problem with the content of sip.conf file or extensions.conf file?
I have not connected any external hardware to my pc. I just want to connect
Asterisk server to my collegues PC's like Intercom within my office LAN using
headphones. How can I do this? Please tell me. Looking forward for your
response. 

Thank you.

Regards,
Chandra.



Evalyn Wafula <[EMAIL PROTECTED]>
wrote: 

Hi Chandra, I am also new to Asterisk and
I have only just started installing a test system but I probably can help
clarify one or two things.

 


 I think asterisk
 "clients" are phones not PCs unless you use "soft
 phones" which is software on the PC (somewhat like Skype)
 that you use to make and answer phone calls. So you might not need to
 install anything on your PCs if you will use IP phones or ATAs as
 mentioned by Gonzalo. 
 The hardware you
 need depends on what you require your asterisk to do. If you will be
 making only IP calls using IP phones, then you only need asterisk running
 on your server with no extra hardware. But if you need to connect with
 analog/digital phone equipment, then you need extra hardware on the
 server.
 You do not
 physically connect your VOIP phone to the asterisk server. You connect it
 to the network that has the server through a normal network point and
 configure it to find the server.
 You probably ought
 to take Gonzalo's advice and head over to:    http://www.voip-info.org/wiki-Asterisk and
 do some reading before you even start as it will help you fit many pieces
 of the asterisk "puzzle" together. It helped me get started.
 Then you probably will have fewer questions that list members will answer
 more readily :)


Regards



 





Wafula



 







Blab-away for as little as 1¢/min. Make PC-to-Phone
Calls using Yahoo! Messenger with Voice.






___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Hi...Please help me

[John Joseph]

can u check what this command gives 
"iptables -L"

or do "iptables -F " [ Not advisable , but for testing
OK ]

 then try again 


--- Crazy Boy <[EMAIL PROTECTED]> wrote:

> Hi friends,
>   
>   Thank you for your response. I am using SuSe Linux
> 9.3 with kernel 2.6 version. I have installed
> Asterisk in my PC and "X-Lite" as softphone in my PC
> and client PC. Here my user name is "chandra" and
> client user name is "aarti". I have added these
> lines to configuration files at the end of file.
>   
>   added contents in sip.conf:
>   
>   [aarti]
>   type=friend
>   username=aarti
>   secret=aarti
>   host=dynamic
>   context=tutorial
>   
>   [chandra]
>   type=friend
>   username=chandra
>   secret=chandra
>   host=dynamic
>   context=tutorial
>   
>   added contents in extensions.conf:
>   
>   [tutorial]
>   exten => 101,1,Dial(SIP/aarti)
>   exten => 102,1,Dial(SIP/chandra)
>   
>   Here, "aarti" is client, "chandra" is mine and
> Asterisk is also installed in my PC (chandra) and it
> is successfully connected to Asterisk server using
> "X-Lite" softphone. 
>   
>   But, when i try to connect from "aarti" system
> using softphone, it displays an error message "login
> timedout, contact system admin". 
>   
>   Is there any problem with the content of sip.conf
> file or extensions.conf file? I have not connected
> any external hardware to my pc. I just want to
> connect Asterisk server to my collegues PC's like
> Intercom within my office LAN using headphones. How
> can I do this? Please tell me. Looking forward for
> your response. 
>   
>   Thank you.
>   
>   Regards,
>   Chandra.
>   
>   
>  
>  Evalyn Wafula <[EMAIL PROTECTED]> wrote:  Hi
> Chandra, I am also new to Asterisk and I have only
> just started installing a test system but I probably
> can help clarify one or two things.
>   
>  
>   I think asterisk "clients" are phones not PCs
> unless you use "soft phones" which is software on
> the PC (somewhat like Skype) that you use to make
> and answer phone calls. So you might not need to
> install anything on your PCs if you will use IP
> phones or ATAs as mentioned by Gonzalo. 
> 
>   The hardware you need depends on what you
> require your asterisk to do. If you will be making
> only IP calls using IP phones, then you only need
> asterisk running on your server with no extra
> hardware. But if you need to connect with
> analog/digital phone equipment, then you need extra
> hardware on the server.
> 
>   You do not physically connect your VOIP phone
> to the asterisk server. You connect it to the
> network that has the server through a normal network
> point and configure it to find the server.
> 
>   You probably oughtto take Gonzalo's advice
> and head over to:   
> http://www.voip-info.org/wiki-Asterisk and do some
> reading before you even start as it will help you
> fit many pieces of the asterisk "puzzle" together.
> It helped me get started. Then you probably will
> have fewer questions that list members will answer
> more readily :)
> 
>  Regards
>   
>  Wafula
>  
>  
>   
> -
> Blab-away for as little as 1¢/min. Make  PC-to-Phone
> Calls using Yahoo! Messenger with Voice.>
___
> --Bandwidth and Colocation provided by Easynews.com
> --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 


Send instant messages to your online friends
http://uk.messenger.yahoo.com 





___ 
Win tickets to the 2006 FIFA World Cup Germany with Yahoo! Messenger. 
http://advision.webevents.yahoo.com/fifaworldcup_uk/
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Hi...Please help me

[Crazy Boy]

Hi friends,Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version. I have installed Asterisk in my PC and "X-Lite" as softphone in my PC and client PC. Here my user name is "chandra" and client user name is "aarti". I have added these lines to configuration files at the end of file.added contents in sip.conf:[aarti]  type=friend  username=aarti  secret=aarti  host=dynamic  context=tutorial[chandra]  type=friend  username=chandra  secret=chandra  host=dynamic  context=tutorialadded contents in extensions.conf:[tutorial]  exten => 101,1,Dial(SIP/aarti)  exten => 102,1,Dial(SIP/chandra)Here, "aarti" is client, "chandra" is mine and Asterisk is also installed in my PC (chandra) and it is successfully connected to Asterisk server using "X-Lite" softphone. But, when i try to connect from "aarti" system using
 softphone, it displays an error message "login timedout, contact system admin". Is there any problem with the content of sip.conf file or extensions.conf file? I have not connected any external hardware to my pc. I just want to connect Asterisk server to my collegues PC's like Intercom within my office LAN using headphones. How can I do this? Please tell me. Looking forward for your response. Thank you.Regards,  Chandra.  Evalyn Wafula <[EMAIL PROTECTED]> wrote:  Hi Chandra, I am also new to Asterisk and I have only just started installing a test
 system but I probably can help clarify one or two things.  I think asterisk "clients" are phones not PCs unless you use "soft phones" which is software on the PC (somewhat like Skype) that you use to make and answer phone calls. So you might not need to install anything on your PCs if you will use IP phones or ATAs as mentioned by Gonzalo.The hardware you need depends on what you require your asterisk to do. If you will be making only IP calls using IP phones, then you only need asterisk running on your server with no extra hardware. But if you need to connect with analog/digital
 phone equipment, then you need extra hardware on the server.   You do not physically connect your VOIP phone to the asterisk server. You connect it to the network that has the server through a normal network point and configure it to find the server.   You probably oughtto take Gonzalo's advice and head over to:    http://www.voip-info.org/wiki-Asterisk and do some reading before you even start as it will help you fit many pieces of the asterisk "puzzle" together. It helped me get started. Then you probably will have fewer questions that list members will answer more readily
 :) Regards   Wafula  
		Blab-away for as little as 1¢/min. Make  PC-to-Phone Calls using Yahoo! Messenger with Voice.___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Hi...Please help me

[Evalyn Wafula]

Hi Chandra, I am also new to Asterisk and I have only just 
started installing a test system but I probably can help clarify one or two 
things.
 

  
  I think asterisk "clients" are phones not PCs unless you use "soft 
  phones" which is software on the PC (somewhat like Skype) that you 
  use to make and answer phone calls. So you might not need to install anything 
  on your PCs if you will use IP phones or ATAs as mentioned by Gonzalo. 
  
  
  The hardware 
  you need depends on what you require your asterisk to do. If you will be 
  making only IP calls using IP phones, then you only need asterisk running on 
  your server with no extra hardware. But if you need to connect with 
  analog/digital phone equipment, then you need extra hardware on the 
  server.
  
  You do not physically connect your VOIP phone to the asterisk server. 
  You connect it to the network that has the server through a normal network 
  point and configure it to find the server.
  
  You probably ought 
  to take Gonzalo's advice and head over to:    http://www.voip-info.org/wiki-Asterisk and 
  do some reading before you even start as it will help you fit many pieces of 
  the asterisk "puzzle" together. It helped me get started. Then you probably 
  will have fewer questions that list members will answer more readily 
  :)
Regards
 
Wafula




From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy 
BoySent: 26 April 2006 14:56To: Asterisk Users Mailing 
List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
Hi...Please help me
Hi,Thank you for your response. Basically, I follow "O Reilly 
AsteriskTFOT.pdf" book and some other eBooks. They have mentioned how to install 
Asterisk in server. But, they have not mentioned 

  What I have to install in client PC's?
  What hardware I need?
  How can I take decission to buy extra hardware (like Zaptel products) OR 
  no need of buying extra hardware? ( I will be using Asterisk for 70 PC's and a 
  server)
  Is it sufficient to buy hardware for server only OR for client PC's also?
  How can I connect my VoIP phone to server?
  How  can I connect hardware to server?
  How can I connect PSTN line to server PC?Please guide me to 
complete this task. Waiting for your response. Thank 
you.Regards,Chandra.Gonzalo Servat 
<[EMAIL PROTECTED]> wrote:
On 
  4/24/06, Crazy Boy <[EMAIL PROTECTED]>wrote:> Hi 
  Friends,>[..snip..]> ---> Employee 1 PC (Softphone 
  i.e., Headphones with Mic)> ---> Employee 2 PC (Softphone i.e., 
  Headphones with Mic)> ---> Employee 3 PC (Softphone i.e., 
  Headphones with Mic)> ---> --> ---> 
  --> ---> Employee 10 PC (Softphone i.e., Headphones 
  with> Mic)>> and vice versa.>> How can I 
  implement this? Is it possible to implement this using "Asterisk"> 
  software? If It can be implemented using "Asterisk" software, What 
  softwares> I should install in Server and Employee PC's? Is there any 
  need of buying> extra hardware?[..snip..]It can be done 
  with Asterisk. For the server side, you would need toinstall Asterisk on 
  your Fedora 5 box, Zaptel and lots of Wikireading.I don't 
  recommend using softphones for your employee PCs. It lookslike an 
  attractive solution at first (from a cost perspective) but inreality it's 
  not very practical (at least that was my experience).Buying 5 x 2 port 
  ATAs will cost you around $300-$350 which is notreally expensive 
  considering the kind of powerful PBX you will have atyour disposal. I 
  would have suggested some Digium hardware for the FXS(extensions) but I 
  think it will be a lot more expensive (for 10extensions) than the ATAs 
  solution. You could also look into a channelbank, but again it will be 
  more expensive than the 5 ATAs. As for theFXO (incoming/outgoing PSTN) I 
  recommend buying Digium hardware(TDM400P).Hope this helps, and 
  good 
  luck!Regards,Gonzalo.___--Bandwidth 
  and Colocation provided by Easynews.com --Asterisk-Users mailing 
  listTo UNSUBSCRIBE or update options 
  visit:http://lists.digium.com/mailman/listinfo/asterisk-users


Blab-away for as little as 1¢/min. Make PC-to-Phone 
Calls using Yahoo! Messenger with Voice.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi...Please help me

[Crazy Boy]

Hi,Thank you for your response. Basically, I follow "O Reilly AsteriskTFOT.pdf" book and some other eBooks. They have mentioned how to install Asterisk in server. But, they have not mentioned What I have to install in client PC's?What hardware I need?How can I take decission to buy extra hardware (like Zaptel products) OR no need of buying extra hardware? ( I will be using Asterisk for 70 PC's and a server)Is it sufficient to buy hardware for server only OR for client PC's also?How can I connect my VoIP phone to server?How  can I connect hardware to server?How can I connect PSTN line to server PC?Please guide me to complete this task. Waiting for your response. Thank you.Regards,Chandra.Gonzalo Servat <[EMAIL PROTECTED]> wrote: On 4/24/06, Crazy Boy <[EMAIL PROTECTED]> wrote:> Hi Friends,>[..snip..]> ---> Employee 1 PC (Softphone i.e., Headphones with Mic)> ---> Employee 2 PC (Softphone i.e., Headphones with Mic)> ---> Employee 3 PC (Softphone i.e., Headphones with Mic)> --->--> --->--> ---> Employee 10 PC (Softphone i.e., Headphones with> Mic)>> and vice versa.>> How can I implement this? Is it possible to implement this using "Asterisk"> software? If It can be implemented using "Asterisk" software, What softwares> I should install in Server and Employee PC's? Is there any need of buying> extra hardware?[..snip..]It can be done with Asterisk. For the server side, you would need toinstall Asterisk on
 your Fedora 5 box, Zaptel and lots of Wikireading.I don't recommend using softphones for your employee PCs. It lookslike an attractive solution at first (from a cost perspective) but inreality it's not very practical (at least that was my experience).Buying 5 x 2 port ATAs will cost you around $300-$350 which is notreally expensive considering the kind of powerful PBX you will have atyour disposal. I would have suggested some Digium hardware for the FXS(extensions) but I think it will be a lot more expensive (for 10extensions) than the ATAs solution. You could also look into a channelbank, but again it will be more expensive than the 5 ATAs. As for theFXO (incoming/outgoing PSTN) I recommend buying Digium hardware(TDM400P).Hope this helps, and good luck!Regards,Gonzalo.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users
 mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
		Blab-away for as little as 1¢/min. Make  PC-to-Phone Calls using Yahoo! Messenger with Voice.___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi...Please help me

[Gonzalo Servat]

On 4/24/06, Crazy Boy <[EMAIL PROTECTED]> wrote:
> Hi Friends,
>
[..snip..]
> ---> Employee 1 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 2 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 3 PC (Softphone i.e., Headphones with Mic)
> --->--
> --->--
> ---> Employee 10 PC (Softphone i.e., Headphones with
> Mic)
>
> and vice versa.
>
> How can I implement this? Is it possible to implement this using "Asterisk"
> software? If It can be implemented using "Asterisk" software, What softwares
> I should install in Server and Employee PC's? Is there any need of buying
> extra hardware?
[..snip..]

It can be done with Asterisk. For the server side, you would need to
install Asterisk on your Fedora 5 box, Zaptel and lots of Wiki
reading.

I don't recommend using softphones for your employee PCs. It looks
like an attractive solution at first (from a cost perspective) but in
reality it's not very practical (at least that was my experience).
Buying 5 x 2 port ATAs will cost you around $300-$350 which is not
really expensive considering the kind of powerful PBX you will have at
your disposal. I would have suggested some Digium hardware for the FXS
(extensions) but I think it will be a lot more expensive (for 10
extensions) than the ATAs solution. You could also look into a channel
bank, but again it will be more expensive than the 5 ATAs. As for the
FXO (incoming/outgoing PSTN) I recommend buying Digium hardware
(TDM400P).

Hope this helps, and good luck!

Regards,
Gonzalo.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi...Please help me

[Paul Hales]

It's all possible.

Paul Hales

--
Paul Hales
Technical Manager
Asterisk IT
bus: 03 8320 8100
mob: 0434 225 491

Crazy Boy wrote:


Hi Friends,

I want to implement VOIP PBX service in my office. I have 10 computers 
and a server. All computers are Pentium IV processors with 512 MB RAM. 
All employee computers have Windows 2000 Professional OS and Server 
computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have 
a VOIP phone and have registered with VoIP service provider. Now, I 
want to implement VOIP PBX facility to all of my systems.


The structure for the same is:

PSTN (Phone call) ---> VOIP phone ---> Server system --->

---> Employee 1 PC (Softphone i.e., Headphones 
with Mic)
---> Employee 2 PC (Softphone i.e., Headphones 
with Mic)
---> Employee 3 PC (Softphone i.e., Headphones 
with Mic)

--->--
--->--
---> Employee 10 PC (Softphone i.e., Headphones 
with Mic)


and vice versa.

How can I implement this? Is it possible to implement this using 
"Asterisk" software? If It can be implemented using "Asterisk" 
software, What softwares I should install in Server and Employee PC's? 
Is there any need of buying extra hardware?


Please reply me. Thank you

Thanks & Regards,

Chandra.


Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great 
rates starting at 1¢/min. 
 







___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi...Please help me

[Marcel Hecko]

For hardware check out this page:
http://www.digium.com/en/products/hardware/

Marcel

Crazy Boy wrote:
> Hi Friends,
> 
> I want to implement VOIP PBX service in my office. I have 10 computers
> and a server. All computers are Pentium IV processors with 512 MB RAM.
> All employee computers have Windows 2000 Professional OS and Server
> computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a
> VOIP phone and have registered with VoIP service provider. Now, I want
> to implement VOIP PBX facility to all of my systems.
> 
> The structure for the same is:
> 
> PSTN (Phone call) ---> VOIP phone ---> Server system --->
> 
> ---> Employee 1 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 2 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 3 PC (Softphone i.e., Headphones with Mic)
> --->--
> --->--
> ---> Employee 10 PC (Softphone i.e., Headphones with
> Mic)
> 
> and vice versa.
> 
> How can I implement this? Is it possible to implement this using
> "Asterisk" software? If It can be implemented using "Asterisk" software,
> What softwares I should install in Server and Employee PC's? Is there
> any need of buying extra hardware?
> 
> Please reply me. Thank you
> 
> Thanks & Regards,
> 
> Chandra.
> 
> Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great
> rates starting at 1¢/min.
> 
> 
> 
> 
> 
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi...Please help me

[Marcel Hecko]

Yes it is possible - check out the Asterisk manual or nice book from
O'Reilly - Asterisk PBX (The Furute of telephony)

Marcel

Crazy Boy wrote:
> Hi Friends,
> 
> I want to implement VOIP PBX service in my office. I have 10 computers
> and a server. All computers are Pentium IV processors with 512 MB RAM.
> All employee computers have Windows 2000 Professional OS and Server
> computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a
> VOIP phone and have registered with VoIP service provider. Now, I want
> to implement VOIP PBX facility to all of my systems.
> 
> The structure for the same is:
> 
> PSTN (Phone call) ---> VOIP phone ---> Server system --->
> 
> ---> Employee 1 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 2 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 3 PC (Softphone i.e., Headphones with Mic)
> --->--
> --->--
> ---> Employee 10 PC (Softphone i.e., Headphones with
> Mic)
> 
> and vice versa.
> 
> How can I implement this? Is it possible to implement this using
> "Asterisk" software? If It can be implemented using "Asterisk" software,
> What softwares I should install in Server and Employee PC's? Is there
> any need of buying extra hardware?
> 
> Please reply me. Thank you
> 
> Thanks & Regards,
> 
> Chandra.
> 
> Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great
> rates starting at 1¢/min.
> 
> 
> 
> 
> 
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Hi...Please help me

[Crazy Boy]

Hi Friends,I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is:PSTN (Phone call) ---> VOIP phone ---> Server system --->             ---> Employee 1 PC (Softphone i.e., Headphones with Mic)            ---> Employee 2 PC (Softphone i.e., Headphones with Mic)            ---> Employee 3 PC (Softphone i.e., Headphones with
 Mic)                    --->     --                    --->     --            ---> Employee 10 PC (Softphone i.e., Headphones with Mic)and vice versa.How can I implement this? Is it possible to implement this using "Asterisk" software? If It can be implemented using "Asterisk" software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank youThanks & Regards,Chandra.
		Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls.  Great rates starting at 1¢/min.___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Hi there..

[Ariel Batista]

All the samples are on your system
/usr/src/asterisk/configs/  the files have a .sample on them.

 

Also there is allow of information on the
Wiki http://www.voip-info.org/wiki-Asterisk

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bharat M. Sarvan
Sent: Thursday, March 17, 2005
6:21 AM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Hi
there..



 

Hello Everybody,


This is Bharat here. I am on the way of learning Asterisks, and I just wished
to know how I go about if got to write dailplans for outbound calls and inbound
calls. If you could provide me with a simple example, I could get thru.

   
Waiting for your response

 

 

 

Regards

 

Bharat M. Sarvan

 

 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Hi there..

[Shaoul Jacobson - TELLINK]

Hi,

 

Welcome.

 

Read the samples *.conf files

(in /etc/asterisk)

extension.conf, sip.conf are
some good places to start.

 

Read & search the wiki.
Many info there (also not always very clear)

 

 

success

 

 



 



 

Shaoul Jacobson

Senior VoIP Consultant

Tellink

Tel : +32 3 201 96 36

Fax :    +32 3 227 09 81

e-mail [EMAIL PROTECTED]



 

-Original Message-
From: Bharat M. Sarvan
[mailto:[EMAIL PROTECTED] 
Sent: jeudi 17 mars 2005 12:21
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Hi
there..

 

Hello Everybody,


This is Bharat here. I am on the way of learning Asterisks, and I just wished
to know how I go about if got to write dailplans for outbound calls and inbound
calls. If you could provide me with a simple example, I could get thru.

   
Waiting for your response

 

 

 

Regards

 

Bharat M. Sarvan

 

 






<>___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi there..

[David Uzzell]

Bharat M. Sarvan wrote:

Hello Everybody,
 This is Bharat here. I am on the way of 
learning Asterisks, and I just wished to know how I go about if got to 
write dailplans for outbound calls and inbound calls. If you could 
provide me with a simple example, I could get thru.

Waiting for your response
 

If you go to http://www.voip-info.org/tiki-index.php and search for 
extensions you will find exactly what you are in need off.

David

 

 

Regards
 

Bharat M. Sarvan
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi there..

[Michiel van Baak]

On 16:51, Thu 17 Mar 05, Bharat M. Sarvan wrote:
> Hello Everybody,
> 
>  This is Bharat here. I am on the way of learning
> Asterisks, and I just wished to know how I go about if got to write
> dailplans for outbound calls and inbound calls. If you could provide me with
> a simple example, I could get thru.
> 
> Waiting for your response
> 

Grab a comfy chair, lots of your fav. caffeiniated drink,
fire up your browser and start reading
http://www.voip-info.org

-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence."

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Hi there..

[Bharat M. Sarvan]

Hello Everybody,


This is Bharat here. I am on the way of learning Asterisks, and I just wished
to know how I go about if got to write dailplans for outbound calls and inbound
calls. If you could provide me with a simple example, I could get thru.

   
Waiting for your response

 

 

 

Regards

 

Bharat M. Sarvan

 

 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi Newbie question

[Rakesh Tiwari]

Thank you all for the quick and straightforward response I was looking out for.

Ed, I am an Indian and have parents and friends in India. So that is
the country that I would like to call to.

But let me tell you that although I am al linux geek and can get some
or more work done with it without swetting, this is my first venture
in the telecom field. Hence I am sorry to say that I didnt quite
follow what is mean by "VOIP termination" or DID's.

To clarify another question that the group may have, I need to call on
landlines or Cell phones.

My parents are not computer  savy people :)

If I could get a howto for the hardawre & software & steps required to
get this working, it would be great.

If there is no such documentation, then I can with the help of the
group work out and then document the procedure for the group.

Thanks for all your help.

Regards,
Rakesh

On Sat, 19 Feb 2005 13:23:02 -0800, Ed Greenberg <[EMAIL PROTECTED]> wrote:
> How does he get his offshore relatives into FWD? Nobody said that they have
> broadband. Just telephones.
> 
> 
> 
> --On Saturday, February 19, 2005 3:15 PM -0600 Rich Adamson
> <[EMAIL PROTECTED]> wrote:
> 
> > Or, he could just sign both ends up with FWD and not have to mess
> > with this *.
> >
> > 
> >
> >> Yes, but he can buy overseas VOIP temination a heck of a lot cheaper
> >> than  just calling overseas from Nebraska.
> >>
> >> He may also be able to start overseas DIDs that route to his box here in
> >> the states.
> >>
> >> Rakesh, if this is what you have in mind, let us know and we'll point
> >> you  in the right direction.
> >>
> >> What countries are you interested in?
> >>
> >> 
> >>
> >>
> >> --On Saturday, February 19, 2005 11:40 AM -0500 [EMAIL PROTECTED]
> >> wrote:
> >>
> >> >> I am in Nebraska, US.
> >> >> I have broadband cable connection at my home. And I have friend and
> >> >> family in other country.
> >> >>
> >> >> Using asterisk and some hardware is it possible for me to call to
> >> >> landlines to other countries. whiout the need to go through or take
> >> >> any service from say "Vonage" or any other service provider.
> >> > No. if you want to call them on the PSTN (landlines), your call need
> >> > at some point to be interfaced with the PSTN.
> >> >
> >> > If you want to talk to them without paying longdistance fee, you can
> >> > always get them VoIP phones (hard or soft) and connect them to your
> >> > asterisk.
> >> >
> >> > hth
> >> > ___
> >> > Asterisk-Users mailing list
> >> > Asterisk-Users@lists.digium.com
> >> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >> > To UNSUBSCRIBE or update options visit:
> >> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >>
> >>
> >> ___
> >> Asterisk-Users mailing list
> >> Asterisk-Users@lists.digium.com
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ---End of Original Message-
> >
> >
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi Newbie question

[Ed Greenberg]

How does he get his offshore relatives into FWD? Nobody said that they have 
broadband. Just telephones.


--On Saturday, February 19, 2005 3:15 PM -0600 Rich Adamson 
<[EMAIL PROTECTED]> wrote:

Or, he could just sign both ends up with FWD and not have to mess
with this *.

Yes, but he can buy overseas VOIP temination a heck of a lot cheaper
than  just calling overseas from Nebraska.
He may also be able to start overseas DIDs that route to his box here in
the states.
Rakesh, if this is what you have in mind, let us know and we'll point
you  in the right direction.
What countries are you interested in?

--On Saturday, February 19, 2005 11:40 AM -0500 [EMAIL PROTECTED]
wrote:
>> I am in Nebraska, US.
>> I have broadband cable connection at my home. And I have friend and
>> family in other country.
>>
>> Using asterisk and some hardware is it possible for me to call to
>> landlines to other countries. whiout the need to go through or take
>> any service from say "Vonage" or any other service provider.
> No. if you want to call them on the PSTN (landlines), your call need
> at some point to be interfaced with the PSTN.
>
> If you want to talk to them without paying longdistance fee, you can
> always get them VoIP phones (hard or soft) and connect them to your
> asterisk.
>
> hth
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
---End of Original Message-
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi Newbie question

[Rich Adamson]

Or, he could just sign both ends up with FWD and not have to mess
with this *.



> Yes, but he can buy overseas VOIP temination a heck of a lot cheaper than 
> just calling overseas from Nebraska.
> 
> He may also be able to start overseas DIDs that route to his box here in 
> the states.
> 
> Rakesh, if this is what you have in mind, let us know and we'll point you 
> in the right direction.
> 
> What countries are you interested in?
> 
> 
> 
> 
> --On Saturday, February 19, 2005 11:40 AM -0500 [EMAIL PROTECTED] 
> wrote:
> 
> >> I am in Nebraska, US.
> >> I have broadband cable connection at my home. And I have friend and
> >> family in other country.
> >>
> >> Using asterisk and some hardware is it possible for me to call to
> >> landlines to other countries. whiout the need to go through or take
> >> any service from say "Vonage" or any other service provider.
> > No. if you want to call them on the PSTN (landlines), your call need
> > at some point to be interfaced with the PSTN.
> >
> > If you want to talk to them without paying longdistance fee, you can
> > always get them VoIP phones (hard or soft) and connect them to your
> > asterisk.
> >
> > hth
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

---End of Original Message-


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi Newbie question

[Ed Greenberg]

Yes, but he can buy overseas VOIP temination a heck of a lot cheaper than 
just calling overseas from Nebraska.

He may also be able to start overseas DIDs that route to his box here in 
the states.

Rakesh, if this is what you have in mind, let us know and we'll point you 
in the right direction.

What countries are you interested in?

--On Saturday, February 19, 2005 11:40 AM -0500 [EMAIL PROTECTED] 
wrote:

I am in Nebraska, US.
I have broadband cable connection at my home. And I have friend and
family in other country.
Using asterisk and some hardware is it possible for me to call to
landlines to other countries. whiout the need to go through or take
any service from say "Vonage" or any other service provider.
No. if you want to call them on the PSTN (landlines), your call need
at some point to be interfaced with the PSTN.
If you want to talk to them without paying longdistance fee, you can
always get them VoIP phones (hard or soft) and connect them to your
asterisk.
hth
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hi Newbie question

[timebandit001]

> I am in Nebraska, US.
> I have broadband cable connection at my home. And I have friend and
> family in other country.
> 
> Using asterisk and some hardware is it possible for me to call to
> landlines to other countries. whiout the need to go through or take
> any service from say "Vonage" or any other service provider.
No. if you want to call them on the PSTN (landlines), your call need
at some point to be interfaced with the PSTN.

If you want to talk to them without paying longdistance fee, you can
always get them VoIP phones (hard or soft) and connect them to your
asterisk.

hth
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Hi Newbie question

[Rakesh Tiwari]

Hi List,

I am a newbie, just came to know about asterisk a few days back while
installing suse 9.2.

I have a question for which I am sorry to say,but I havntread through
all the archives, but AFAIK i didnt get the answer in the archives.

My question is like this.

I am in Nebraska, US.
I have broadband cable connection at my home. And I have friend and
family in other country.


Using asterisk and some hardware is it possible for me to call to
landlines to other countries. whiout the need to go through or take
any service from say "Vonage" or any other service provider.

Regards,
Rakesh
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Hi, Can any one send me a sample uac.xml for SIPP test?

[VoIP]

Hi, Just want to use SIPP to test my asterisk box. But confused with the
uac.xml file configuration. Can any one send me one copy?

Thanks in advance. 
T.Y.


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Hi

[surajee]

 Test =)
tuexgjbxgqndvdnmc
--
Test, yep.
<>

Re: [Asterisk-Users] (HI,new to asterisk)connecting asterisk to telephonyhardware

[Jeremy McNamara]

reddy wrote:

Hi,

i have configured asterisk to SIP soft phones.

Iam confused about the telephony interfaces like T1,E1,PRI interfaces  and how to cnfigure them to asterisk.

If iam using a dialogic card in the asterisk server .Can i use  this card  with a telephone line to dialout to different numebr in the telecom network.

SIP-softphone--->---asterisk-->Dialogic card--->telephone line -->called party
 

Don't use Dialogic.  Support Asterisk and pick up an appropriate Zaptel 
based card from Digium.

http://www.digium.com/

Jeremy McNamara



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] (HI,new to asterisk)connecting asterisk to telephonyhardware

[reddy]

Hi,

i have configured asterisk to SIP soft phones.

Iam confused about the telephony interfaces like T1,E1,PRI interfaces  and how to 
cnfigure them to asterisk.

If iam using a dialogic card in the asterisk server .Can i use  this card  with a 
telephone line to dialout to different numebr in the telecom network.

SIP-softphone--->---asterisk-->Dialogic card--->telephone line 
-->called party

or vice a versa.

Do  i have to make any software modifications to achieve this

Thanks & Regards

Jagadheeswar Reddy

Senior Wireless Engineer
Java Wireless Competency Centre 
National University Of Singapore
DID   : +65 68709535
Fax   : +65 68723546
Email : [EMAIL PROTECTED]
URL   : http://www.jwcc.net

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] HI.. .Any comments about VoicePulse

[Alvaro Parres]

Hi... some one can tell me his comments about VoicePulse Services...

for Pre-Paid long distance..

Alvaro parres





___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users