date:20030918

it could well be my conf files?

could you possibly let us to see your conf files?

Senad



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: 18 September 2003 07:13
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Nufone 800 numbers working?


Its working here Senad, check your configs

  -- Starting simple switch on 'Zap/4-1'
-- Executing Dial(Zap/4-1, IAX2/[EMAIL PROTECTED]/18004601446) in new 
stack
-- Called [EMAIL PROTECTED]/18004601446
-- Call accepted by 65.127.126.42 (format ILBC)
-- Format for call is ILBC
-- Accepting AUTHENTICATED call from 65.127.126.42, requested format = 
4, actual format = 1024

...

also whenever someone has a problem please be sure to paste the debug and 
the relevant stuff from (in this case, extensions.conf and iax.conf)

helps someone give a better answer than (IT WORKS HERE! fix yours) etc...

- wasim


On Thu, 18 Sep 2003, Senad Jordanovic wrote:

 well, i have same problem...
 
 it sounds like nufone is not allowing calling of #800.
 anyone from nufone care to comment?
 
 Senad
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Re: [Asterisk-Users] calls terminating abnormally

2003-09-18 Thread denzel-infotechs


Forgot to mention that I commented out
;callprogress
;busydetect
to remedy call termination.



- Original Message -
From: denzel-infotechs [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 18, 2003 7:36 PM
Subject: Re: [Asterisk-Users] calls terminating abnormally


 hi!
 Infact the problem now being shifted for temporary silence in calls
 where one party could not hear the other. This lasts for even 2 to 2.5
 seconds. I got 2 * server where one is connected to PSTN and the other to
 internal PBX. When calls are from extension to the outside, it flows
 like extension-pbx---ISDN
 PRIE1server2IAX2---server1-ISDN PRIE1---PSTN.
 Both servers are in the same LAN.
 I've got tos=reliability
 Does jitter has to do anything here. I've got my jitter set to default.
 I'll send you a debug span in time.


 denzel.




 - Original Message -
 From: Martin Pycko [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Wednesday, September 17, 2003 9:20 PM
 Subject: Re: [Asterisk-Users] calls terminating abnormally


  Can you send a pri debug span span_no trace ? Or do you have an
analog
  T1/E1 ?
 
  regards
  Martin
 
  On Wed, 17 Sep 2003, denzel-infotechs wrote:
 
   hi!
   I've got a asterisk system running with around 50 per calls per
 minute.  I've connected * to internal pabx and outside telecom using E1
 (ISDN pris). Sometimes calls disconect abnormally. Is this something we
have
 to live with or is it a bug in CVS code  ?
  
   denzel.
  
 
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RE: [Asterisk-Users] Nufone 800 numbers working?

2003-09-18 Thread wasim

soytanly

iax.conf has

[nufone]
type=peer
username=cbspak 
secret=asteriskisthekickassestpbxontheplanet
context=WORLD   ;you may have NANPA in this
host=switch-1.nufone.net 

and extensions.conf has

NUX=IAX2/[EMAIL PROTECTED]
exten = _1800NXX,1,Dial(${NUX}/${EXTEN})

ofcourse, i did this to test nufone, normally we have

IAXTEL=IAX2/[EMAIL PROTECTED]
exten = _1800NXX,1,Dial(${IAXTEL}/${EXTEN})

i love jerjer, but why pay when you get it for free!

now, someone tell me if gets any simpler than this...
eat your heart out Avaya we know about that linuxpbx 
winkwink

- wasim (sorry about top posting)

On Thu, 18 Sep 2003, Senad Jordanovic wrote:

 it could well be my conf files?
 
 could you possibly let us to see your conf files?
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Re: [Asterisk-Users] Nufone 800 numbers working?

2003-09-18 Thread TC

well, i have same problem...

it sounds like nufone is not allowing calling of #800.
anyone from nufone care to comment?
I have seen nufone die, if the callerid is not 
a cid from us 48 try setting your sic to 


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Re: [Asterisk-Users] CODECS and thier practical usage stats

2003-09-18 Thread John Todd

Hi,

What are real life bandwith stats for * supported codecs?
Is it true one can run 6-32 conversations over DSL, as stated in this list?
Senad
Google will also give you the results I just found.

http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html

JT
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Re: [Asterisk-Users] documentation?

2003-09-18 Thread John Todd

-= On Wed, 17 Sep 2003 11:01:34 -0600, Rich Adamson 
[EMAIL PROTECTED] said:

 Examples,
 Where should I have learned that *8# is the call pickup dialing sequence?
A good question.  I didn't know about any of them until James Sizemore
posted this handy list on Sept.8:
*0#  sends flash
*8#  remote call pickup (pickup phone in your group)
*67# disable caller id
*70# no call waiting
*78# do not disturb on
*79# do not disturb off
*72# enable call forwarding
*73# disable call forwarding
*82# enable callerid
All news to me. :) I do a lot of google searching on the Asterisk
archives:
  http://www.google.com/custom?sitesearch=lists.digium.com
There are some fairly rational reasons to see some of these elements 
removed from chan_zap and made into non-channel-specific applications.

http://bugs.digium.com/bug_view_page.php?bug_id=071


[snip]
There are some inconsistencies that will probably work themselves out
over time, like the whole:
 App,arg1,arg2
 App(arg1|arg2)
 App(arg1,arg2)
I can't quite figure out if some things still *require* the vertical
pipe, like going to another extensions:
  400 = Goto(139343234|1)
[snip]

...and there are people who would like to see this standardized, too, 
since everyone stumbles across the same problems as you describe 
above.

http://bugs.digium.com/bug_view_page.php?bug_id=274

JT
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[Asterisk-Users] Need help with H.323

2003-09-18 Thread Roy Sigurd Karlsbakk

hi all

I'm trying to setup a dlink dph-100h phone (actually a dph-100m but with
the h.323 software) with asterisk and chan_h323. AFACS, the dph-100h
software can only be configured to point to a gatekeeper. I know I don't
need to do this, but it's a test before I setup my Symbol Netvision
phones (I don't have an access point for them now). First, this is what
I have, and what I've done:

- Platform is Debian Woody/Stable
- Asterisk is fresh from CVS, and so is chan_h323
- chan_h323 did compile, but also died from a rather nasty SIGSEGV
  (reported earlier) with Open H.323 v1.11.7 and PWLib v1.4.11. I
  checked them out from CVS instead after being adviced so by diana
  (on irc), and it now loads into memory without any fuzz.
- Several people have told me chan_capi can work as a gatekeeper, so
  there should be no use for gnugk or any others. I have yet to find
  where this is hidden. FWICS, this is all commented out (ast_h323.cpp
  line 722). Is this right or have I overseen anything?
- If this cannot be done alone with chan_h323, I guess I need to use
  gnugk or something. Can someone help me out how these two should be
  configured to work together? Also - is it possible to run gnugk on
  the same host as asterisk?

Thanks for all help

Best regards

Roy Sigurd Karlsbakk [EMAIL PROTECTED]

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Re: [Asterisk-Users] Nufone 800 numbers working?

2003-09-18 Thread marrandy

On Thursday 18 September 2003 06:50 am, [EMAIL PROTECTED] wrote:


Well, I can do outbound calls via NuFone, but nothing on inbound.  I get a 
message that saysThe person you are calling is not reachable, 
please try again later.

IAX2 debug shows nothing.

After some time, I copied my config files elsewhere and started with a clean 
slate, simplified with just one phone (zap/2-1) using NuFone only.

Still the same.  I can call out but no inbound and no iax2 debug info.

I've asked Jeremy (NuFone) to provide the absolute minimum config files to 
call in and out on a zap/2-1 phone, in the hope that it either work or show a 
problem elsewhere.  Unfortunately, he seems unable to do that.

So, if anyone has a working inbound/outbound Nufone connection with a zap/2-1, 
I'd like their configs - zapata.conf, iax.conf  extensions.conf, replace 
actual password with 'password'  of course.

-- 
Paprika Measure:
2 dashes==  1smidgen
2 smidgens  ==  1 pinch
3 pinches   ==  1 soupcon
2 soupcons  ==  2 much paprika

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[Asterisk-Users] VoicePulse offering IAX2 services

2003-09-18 Thread Peter Pauly

I don't know if this has been mentioned yet:

Voicepulse is now offering wholesale pricing and
IAX2 connectivity for Asterisk users. No fees, pay 
as you go. They also
offer incoming calls for $7.99 per month. See
wholesale.voicepulse.com.

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Re: [Asterisk-Users] Need help with H.323

2003-09-18 Thread Klaus-Peter Junghanns


Am Don, 2003-09-18 um 16.36 schrieb Roy Sigurd Karlsbakk:
 hi all

hi roy,

 
 I'm trying to setup a dlink dph-100h phone (actually a dph-100m but with
 the h.323 software) with asterisk and chan_h323. AFACS, the dph-100h
 software can only be configured to point to a gatekeeper. I know I don't
 need to do this, but it's a test before I setup my Symbol Netvision
 phones (I don't have an access point for them now). First, this is what
 I have, and what I've done:
 
 - Platform is Debian Woody/Stable
 - Asterisk is fresh from CVS, and so is chan_h323
 - chan_h323 did compile, but also died from a rather nasty SIGSEGV
   (reported earlier) with Open H.323 v1.11.7 and PWLib v1.4.11. I
   checked them out from CVS instead after being adviced so by diana
   (on irc), and it now loads into memory without any fuzz.
 - Several people have told me chan_capi can work as a gatekeeper, so
   there should be no use for gnugk or any others. I have yet to find
   where this is hidden. FWICS, this is all commented out (ast_h323.cpp
   line 722). Is this right or have I overseen anything?

i can confirm that chan_capi will not work as a gatekeeper under any
circumstances! somebody is obviously trying to confuse you.;-)

 - If this cannot be done alone with chan_h323, I guess I need to use
   gnugk or something. Can someone help me out how these two should be
   configured to work together? Also - is it possible to run gnugk on
   the same host as asterisk?
 
 Thanks for all help
 
 Best regards
 
 Roy Sigurd Karlsbakk [EMAIL PROTECTED]
 

best regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk
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Re: [Asterisk-Users] Nufone 800 numbers working?

2003-09-18 Thread Peter Pauly

On Thu, Sep 18, 2003 at 07:02:42AM -0700, TC wrote:
 well, i have same problem...
 
 it sounds like nufone is not allowing calling of #800.
 anyone from nufone care to comment?
 I have seen nufone die, if the callerid is not 
 a cid from us 48 try setting your sic to 

I added SetCallerID and SetCIDName steps before the dial and
it works now. Funny, it worked before without these steps. 
 
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RE: [Asterisk-Users] Nufone 800 numbers working?

hmm.. this is weird...

I can call all other numbers except #800...

and I can not see anything wrong in my conf files.

here they are, if you find something, please shout...

iax.conf
---
[general]
port=5036
bandwidth=high
Deny=all
allow=iLBC
register= bicomus:[EMAIL PROTECTED]


[NuFone]
type=peer
host=switch-1.nufone.net
secret=XX
context=intern
callerid=18775891760
auth=md5
deny=all
allow=iLBC



[NuFone]
type=user
secret=XX
context=nufone-receiving
callerid=18775891760
auth=md5
deny=all
allow=iLBC

extensions.conf

[nufone-out]
exten = _5.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:1}



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[Asterisk-Users] Possible FAQ: IAX2 - SIP with G729 and no licence

2003-09-18 Thread Linus Surguy

Assuming I've got a setup where calls entering Asterisk on SIP leave on IAX2
( and the reverse), i.e. a SIP user might dial '1234' where we then have

extern = 1234,1,Dial(IAX2/somewhereelse)

Now, we don't have any G.729 functionality on this server, so what happens
if the SIP user calls with G.729 only available?

Assuming the remote IAX2 server does have G.729 can it be passed through to
it?

Linus



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[Asterisk-Users] CDR of calls transferred via IAX[2]

2003-09-18 Thread Lele Forzani



Let's say i have a network of * boxes connected via IAX, one of them is a 
switch, one or more are the gateways.

- An IAX[2] customer register himself on the switch (and gets an 
accountcode for te purpose of cdr)

- The customer places a call to the switch, the switch does some magic and 
decides which gateway the call should be forwarded

- The switch authenticates the call with the gateway and then performs a 
transfer effectively connecting the customer directly with the gateway 
(which is something I specifically want, this rules out a notransfer=yes 
solution)

- The gateway does something expensive (= calls the PSTN) I want to be 
billed

From a CDR standpoint, i have a cdr record from the switch containing the 
accountcode for the user, but useless billing informations since the call has 
been transferred,
The gateway has, of course, useful billing information, but doesn't have the 
original accountcode for the user, since the call was coming from the 
switch

Given that I can't trust the callerid, and I can't set it to something else (i 
must accept from customer any callerid and pass it to the PSTN), I would 
like to bill the calls based on the accountcode for the user.
Here comes the trouble: since neither the accountcode nor the uniqueid are 
preserved during the transfer,  i do not see anything to safely correlate the 
accountcode with the billing records on the gateway. I can guess at it 
based on the call specific callerid, the time of day and such. But it would 
be guessing.


A few thoughts on it:

* one could pass via IAX a uniqueid when i transfer the call, and have this 
unique id logged in the CDR records. This way any call segment pertaining to 
the same phone call can be correlated for cdr purpose.


* one could have the gateway allow trusted sources (the switch) to set via 
IAX the accountcode when transferring the call, and log it as an 
originalaccountcode or even the accountcode itself in the cdr.
This way every cdr record in the network will have a reference to the actual 
customer that made that event happen.

* one could devise some way to give back from the gateway to the transferrer 
(the switch) an indication that the call has ended, with that many 
billable_seconds. (can this be done? i do not see it that simple...)
This way the switch would have all the cdr info in one cdr row.


Anybody has suggestions on this?

thanks
lele



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RE: [Asterisk-Users] CODECS and thier practical usage stats

John, Tx

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RE: [Asterisk-Users] Possible FAQ: IAX2 - SIP with G729 and no licence

2003-09-18 Thread Dave Wilson

 Assuming I've got a setup where calls entering Asterisk on
 SIP leave on IAX2
 ( and the reverse), i.e. a SIP user might dial '1234' where
 we then have

 extern = 1234,1,Dial(IAX2/somewhereelse)

 Now, we don't have any G.729 functionality on this server, so
 what happens
 if the SIP user calls with G.729 only available?

 Assuming the remote IAX2 server does have G.729 can it be
 passed through to
 it?


Linus,

Theoretically (in network terms), there shouldn't be an issue as G.729 is a
codec, whereas the process you are referring to describes transporting the
codecs from A to B. The transporting is handled by the transport protocols
(SIP,IAX2,etc).

Whether this theory applies to Asterisk or not - I don't know. My current
understanding is that Asterisk acts like a router in a sense, transmitting
packets along channels to the client which in turn reads the audio stream
using the codec selected. So unless Asterisk performs some other tasks with
the codecs your suggestion should work fine.

Dave





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[Asterisk-Users] e100p and E-bit alarm indication

2003-09-18 Thread Lele Forzani


We connected an * box with an e100p to an E1/PRI from a telco here in Italy.

After we had it working perfectly the telco told us that, despite the circuit 
appeared to work fine, and we could place calls on it, they had an E-bit2 
alarm indication constantly on that caused the circuit to be flagged as 
faulty every time.

(The E-bit indication, is an alarm sent back from us to the telco, telling 
them we are getting CRC-errored data from them. It should be incrementing the 
Far-End SES on their side)

Since the circuit appeared to work fine, calls went through, and the crc 
counters on our side was zero, it was impossible we were getting that many 
errors and something must have been wrong with our handling of the E-bit 
signal.

I've come across the DS21554 framer documentation and i've seen that it has a 
flag for enabling the E-bit generation in the TCR2 register and that the 
wct1xxp.c wasn't setting it. 

So i tried this small patch and the telco is perfectly happy with it, now, the 
E-bit error has disappeared.


Since there had been a thread in May (started by Konrad Gorsky) about weird 
far end CRC errors i'm posting in the hope to help somebody.

Note that i do not have a clue on what this does to the *T1* framer. I do not 
have the specs for it!

bye
lele





--- zaptel/wct1xxp.c2003-09-12 10:12:01.0 +0200
+++ zaptel-i/wct1xxp.c  2003-09-11 19:24:53.0 +0200
@@ -411,13 +411,14 @@
int alreadyrunning = wc-span.flags  ZT_FLAG_RUNNING;
long flags;
char *crcing = ;
-   unsigned char ccr1, tcr1;
+   unsigned char ccr1, tcr1, tcr2;

spin_lock_irqsave(wc-lock, flags);

/* Build up config */
ccr1 = 0;
tcr1 = 8;
+   tcr2 = 0;
if (wc-span.lineconfig  ZT_CONFIG_CCS) {
coding = CCS; /* Receive CCS */
ccr1 |= 8;
@@ -433,9 +434,11 @@
}
if (wc-span.lineconfig  ZT_CONFIG_CRC4) {
ccr1 |= 0x11;
+   tcr2 |= 0x02;   // xxx Enable E-bit alarm
crcing =  with CRC4;
}
__t1_set_reg(wc, 0x12, tcr1);
+   __t1_set_reg(wc, 0x13, tcr2);
__t1_set_reg(wc, 0x14, ccr1);
__t1_set_reg(wc, 0x18, 0x20);   /* 120 Ohm */

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Re: [Asterisk-Users] calls terminating abnormally

On Thu, 2003-09-18 at 08:36, denzel-infotechs wrote:
 hi!
 Infact the problem now being shifted for temporary silence in calls
 where one party could not hear the other. This lasts for even 2 to 2.5
 seconds. I got 2 * server where one is connected to PSTN and the other to
 internal PBX. When calls are from extension to the outside, it flows
 like extension-pbx---ISDN
 PRIE1server2IAX2---server1-ISDN PRIE1---PSTN.
 Both servers are in the same LAN.
 I've got tos=reliability
 Does jitter has to do anything here. I've got my jitter set to default.
 I'll send you a debug span in time.


Jitter buffer is probably the culprit now. Turn jitter off and you
should have no problems. This is what occurred to us on our 2 * servers
with a T1 data link in between.

 - Original Message -
 From: Martin Pycko [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Wednesday, September 17, 2003 9:20 PM
 Subject: Re: [Asterisk-Users] calls terminating abnormally
 
 
  Can you send a pri debug span span_no trace ? Or do you have an analog
  T1/E1 ?
 
  regards
  Martin
 
  On Wed, 17 Sep 2003, denzel-infotechs wrote:
 
   hi!
   I've got a asterisk system running with around 50 per calls per
 minute.  I've connected * to internal pabx and outside telecom using E1
 (ISDN pris). Sometimes calls disconect abnormally. Is this something we have
 to live with or is it a bug in CVS code  ?
  
   denzel.
  
 
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 ___
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-- 
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Re: AW: AW: AW: [Asterisk-Users] h323 gatekeeper registration failed

2003-09-18 Thread Michael Manousos

Thomas Haeger wrote:
Hi Michael,

registration is working now, it dials out the phone is ringing but then
comes a hang up
I'am i lttle newbe on h323  :-)
Can you take a look on the log file ?
Your connection attempt terminates with a EndedByRefusal
reason. My guess is that you are not allowed to use the codec
you are trying to use (e.g. if you are using a g.711 try to switch
to a lower bit-rate one). Or some other reason?
Thanks,

Thomas.



Michael.

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Michael
Manousos
Gesendet: Dienstag, 16. September 2003 15:54
An: [EMAIL PROTECTED]
Betreff: Re: AW: AW: [Asterisk-Users] h323 gatekeeper registration
failed
Thomas Haeger wrote:

No. I have installed the versions wich your special friend has
recommended.

Shall i try to update to the newest versions ? (But then wouldn't work the
chan_h323.so further...)


I don't know what are the problems with that driver, but, yes,
you should install the latest versions.
Before this, check the configuration of the remote gatekeeper
(if this is possible) and see if there are special requirements
for the registration.
Michael.



-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Michael
Manousos
Gesendet: Dienstag, 16. September 2003 13:53
An: [EMAIL PROTECTED]
Betreff: Re: AW: [Asterisk-Users] h323 gatekeeper registration failed
Thomas Haeger wrote:


Hi Michael,

this gatekeeper works without a password but with a H323-ID, but this will
be send with the dial command, i think.


No, this id is provided during registration.



Here is the trace with trace level 10 (?) 


Unfortunately, the GK rejects the registration attempt
with an undefined reason (!).
Did you try it with the latest OpenH323/pwlib ?


Regards,

Thomas.



Michael.




-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Michael
Manousos
Gesendet: Dienstag, 16. September 2003 12:22
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] h323 gatekeeper registration failed


If the gatekeeper requires a password and you don't provide one
during the registration, then it will fail.
In oh323.conf use the gatekeeperPassword to provide the passwd.
If this is not the case enable tracing info in oh323.conf, rerun
and send me
the trace file to take a look.
Michael.

Thomas Haeger wrote:



Hi all,

i have tried to connect to a clarent gatekeeper.
I have used both of h323 drivers chan_h323.so and chan_oh323.so.
But no one can register to this gatekeeper.
Our ip is activated on this gatekeeper.
Maybe, i do wrong anything

I have only set the gatekeeper option in the h323.conf or oh323.conf to
the ip address from the gatekeeper.
gatekeeper=x.x.x.x
But no one of the both driver can register to this gateway.

Is there another thing that i have to keep ?

I need yours help urgently. We want to go online with our *-gateway as
soon



as possible.

Thanks,
Thomas.
***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email:  [EMAIL PROTECTED]
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[Asterisk-Users] SIP registration

2003-09-18 Thread Tais M. Hansen

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

I'm having problems letting a SIP endpoint register at Asterisk. Here's the 
debug output from Asterisk:

Sip read:
REGISTER sip:s.s.s.s;transport=UDP SIP/2.0
User-Agent: ATI-RG613/1-1-0_8
From: atrg613test sip:[EMAIL PROTECTED];tag=AABcMQAMRhB0AAxx
To: atrg613test sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 94 REGISTER
Contact: sip:[EMAIL PROTECTED]
Max-Forwards: 70
Via: SIP/2.0/UDP c.c.c.c;branch=z9hG4bKAQA4Mwxx
Content-Length: 0


10 headers, 0 lines
Using latest request as basis request
Sending to c.c.c.c : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP c.c.c.c;branch=z9hG4bKAQA4Mwxx
From: atrg613test sip:[EMAIL PROTECTED];tag=AABcMQAMRhB0AAxx
To: atrg613test sip:[EMAIL PROTECTED];tag=as1966d2fc
Call-ID: [EMAIL PROTECTED]
CSeq: 94 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


 to c.c.c.c:5060

...

sip.conf:

[general]
port=5060
bindaddr=s.s.s.s
context=cxnet-in
tos=lowdelay

[siptestphone]
type=friend
user=atrg613test
host=dynamic
defaultip=c.c.c.c


- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/Linux)

iD8DBQE/adS12TEAILET3McRAjnlAJ9HE+zxry1+qp2/Y7fqJFh8ea4MFACbB2/E
YOLGiZTXMKqBtGCtZqBryD4=
=3qTK
-END PGP SIGNATURE-

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Re: [Asterisk-Users] VoicePulse offering IAX2 services

2003-09-18 Thread Tjardick van der Kraan

- Original Message - 
From: Peter Pauly [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 18, 2003 4:44 PM
Subject: [Asterisk-Users] VoicePulse offering IAX2 services

 I don't know if this has been mentioned yet:

 Voicepulse is now offering wholesale pricing and
 IAX2 connectivity for Asterisk users. No fees, pay
 as you go. They also
 offer incoming calls for $7.99 per month. See
 wholesale.voicepulse.com.

Can't believe there are still companies who ask for credit-card information
on their form on a page which is not https...

Tj

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Re: [Asterisk-Users] Nufone 800 numbers working?

2003-09-18 Thread marrandy



On Thursday 18 September 2003 10:48 am, marrandy wrote:
 On Thursday 18 September 2003 06:50 am, [EMAIL PROTECTED] wrote:
 
 
 Well, I can do outbound calls via NuFone, but nothing on inbound.  I get a 
 message that saysThe person you are calling is not reachable, 
 please try again later.
 
 IAX2 debug shows nothing.

Hello.

Well, finally, the problem is solved.

register = user:[EMAIL PROTECTED]

MUST be in the [general] section of the iax.conf to work.

If it's elsewhere, it fails.  The instructions I received didn't say, or even 
hint, where it should be placed, so I placed it at the bottom followed by the 
[NuFone] contexts.

Regards...Martin
-- 
Getting there is only half as far as getting there and back.

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[Asterisk-Users] SIP error messages

2003-09-18 Thread marrandy

Hello.

I'm seeing this at the console.

NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration from 
'sip:[EMAIL PROTECTED]' failed for '192.168.1.70'

What's this all about ?

Regards...Martin
-- 
Osborn's Law:
Variables won't; constants aren't.

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Re: [Asterisk-Users] Skinny + XMLDefault

2003-09-18 Thread Eric Wieling

If your Asterisk server is on the same machine as your DHCP server then
you should not need the .cnf file.  My 7910 (running SCCP/Skinny) finds
my Asterisk server just fine.  If your DHCP server is not running on the
same machine as your Asterisk server you may fine this URL to be
helpful.  If talks about how the SCCP phones find their Call Manager.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00800c4bba.html

On Thu, 2003-09-18 at 04:03, Alexander Noack wrote:
 Please forgive me my ignorance ...
 
 I've spent two days trying to find out something about the format of the
 default configuration file, which CCM produces. The only example I have so
 far is the one from the chan_sccp source.
 
 There were tons of references on entering the callmanager commands on a
 cisco command line - which I don't have (don't need thanks to
 chan_skinny + chan_sccp).
 
 I guess cisco doesn't want you to know, since you're supposed to do
 everything via CCM. If someone found out something, I'd appreciate any
 input on this!
 
 Thanks,
 Alex
-- 
Sample configs and more: http://www.fnords.org/~eric/asterisk/

BTEL Consulting
+1-850-484-4535 x2111 (Pensacola)
+1-504-595-3916 x2111 (New Orleans)
+1-877-677-9643 x2111 (Toll Free)

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AW: AW: AW: AW: [Asterisk-Users] h323 gatekeeper registration failed

2003-09-18 Thread Thomas Haeger

Ahh... you mean it's a codec problem? This can be...
I ask my provider :-).

If this was not the prob, i would get in touch with you.

Regards,
Thomas.

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Michael
Manousos
Gesendet: Donnerstag, 18. September 2003 17:48
An: [EMAIL PROTECTED]
Betreff: Re: AW: AW: AW: [Asterisk-Users] h323 gatekeeper registration
failed


Thomas Haeger wrote:
 Hi Michael,

 registration is working now, it dials out the phone is ringing but then
 comes a hang up
 I'am i lttle newbe on h323  :-)

 Can you take a look on the log file ?

Your connection attempt terminates with a EndedByRefusal
reason. My guess is that you are not allowed to use the codec
you are trying to use (e.g. if you are using a g.711 try to switch
to a lower bit-rate one). Or some other reason?


 Thanks,

 Thomas.



Michael.


 -Ursprüngliche Nachricht-
 Von: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Auftrag von Michael
 Manousos
 Gesendet: Dienstag, 16. September 2003 15:54
 An: [EMAIL PROTECTED]
 Betreff: Re: AW: AW: [Asterisk-Users] h323 gatekeeper registration
 failed


 Thomas Haeger wrote:

No. I have installed the versions wich your special friend has

 recommended.

Shall i try to update to the newest versions ? (But then wouldn't work the
chan_h323.so further...)


 I don't know what are the problems with that driver, but, yes,
 you should install the latest versions.
 Before this, check the configuration of the remote gatekeeper
 (if this is possible) and see if there are special requirements
 for the registration.

 Michael.



-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Michael
Manousos
Gesendet: Dienstag, 16. September 2003 13:53
An: [EMAIL PROTECTED]
Betreff: Re: AW: [Asterisk-Users] h323 gatekeeper registration failed


Thomas Haeger wrote:


Hi Michael,

this gatekeeper works without a password but with a H323-ID, but this
will
be send with the dial command, i think.


No, this id is provided during registration.



Here is the trace with trace level 10 (?) 


Unfortunately, the GK rejects the registration attempt
with an undefined reason (!).
Did you try it with the latest OpenH323/pwlib ?



Regards,

Thomas.



Michael.




-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Michael
Manousos
Gesendet: Dienstag, 16. September 2003 12:22
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] h323 gatekeeper registration failed



If the gatekeeper requires a password and you don't provide one
during the registration, then it will fail.
In oh323.conf use the gatekeeperPassword to provide the passwd.

If this is not the case enable tracing info in oh323.conf, rerun
and send me
the trace file to take a look.

Michael.


Thomas Haeger wrote:



Hi all,

i have tried to connect to a clarent gatekeeper.
I have used both of h323 drivers chan_h323.so and chan_oh323.so.
But no one can register to this gatekeeper.
Our ip is activated on this gatekeeper.

Maybe, i do wrong anything

I have only set the gatekeeper option in the h323.conf or oh323.conf
to
the ip address from the gatekeeper.
gatekeeper=x.x.x.x

But no one of the both driver can register to this gateway.

Is there another thing that i have to keep ?


I need yours help urgently. We want to go online with our *-gateway as

soon



as possible.

Thanks,
Thomas.

***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow

FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email:  [EMAIL PROTECTED]
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Re: [Asterisk-Users] End Hide

So is that a new kind of klez virus unleashed only to mailing lists? 
For those not used to looking at headers, the true sender of that
message used a machine in Sri Lanka to send the virus. I say used
because I'm not convinced it wasn't an intentional message. 
-- 
Steven Critchfield  [EMAIL PROTECTED]

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RE: [Asterisk-Users] frames/packet

2003-09-18 Thread Abdul Hakeem

Hi,
A bit late replying to this.
My comments are below:

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Lambert
Sent: 03 September 2003 17:16
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] frames/packet

Not yet. implies that it is coming. I know it would help on Internet
connections such as fixed wireless and cable modem where packet rate is
an issue. 20ms translates to 50 packets/sec. I believe cable modem
upstream packet rates cap at 150-200 packets/sec. G729 gets the bit rate
down to 8kbits.
You can actually set the bytes to about 200 or more, that should
reduce the packet rates down to about 10/sec

 So based on a bit rate of 256K the theory is that the link could handle
32 calls. But, that would produce packets coming out at a rate of 1600
packets/sec beyond the limitation of most Internet connections including
a T1.
we have managed to run 120 simultaneous calls on 1xE1 link which is
about 2.048kbps of bandwidth(slightly bigger that a T1).

The theory, many a times, do not actually hold.
Cheers,
Abdul

Martin Pycko wrote:

 Not yet.

 Asterisk always sends 20 ms of voice data per packet.

 regards
 Martin

 On Wed, 3 Sep 2003, Paul Lambert wrote:

  Noticed that I can adjust the number if frames/packet on the 
  GrandStream phone. Can * do the same? 
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[Asterisk-Users] h.323 - success

2003-09-18 Thread Roy Sigurd Karlsbakk

hi

seems like things are closing in to something that might look like
success. I have one problem left: I don't get ring indicator when I dial
out from the h.323 phone... Sound is good, so it doesn't look like a
codec problem. I'm using chan_capi with early B3. I also use gnugp to
route the calls from the phones to asterisk, as the dlink dph-100h
requires this. Debug output follows:

Any ideas?

roy
 DEBUG -
*CLI exten b4: 98013356
-- Executing Dial(H323/ip$10.47.0.1:39307/29476,
CAPI/22545070:b98013356|300|T) in new stack
-- Called 22545070:b98013356
  us: 0.0.0.0:6124
them: 0.0.0.0:0
info: 0.0.0.0:6124
  us: 0.0.0.0:6124
them: 0.0.0.0:0
info: 0.0.0.0:6124
-- CAPI[contr1/22545070]/8 is ringing




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Re: [Asterisk-Users] Radio for Music on Hold?

2003-09-18 Thread Chris Albertson


You can't legally do this.  At least no here in the
US.
The music being bradcast by the station may not
legally be re-broadcast.  Yes many people do this and
yess thay also download copywritten MP3 files.

Call any radio station and ASK THEM if it is OK don't
trust some e-mail list.  Who knows maybe they won't
care what you do but they do have to pay  for the
rights to broadcast music and teir payment does
notcover re-broadcast.

Tell your client that some callers put on hold may
know about the above and radio on hold would make
the company look at best ignorent.

Yesterday I called a company, was put on hold and
heard the local govenment weather radio broadcast. 
This was smart and legal too.



 I have a client who is interested in using a
 radio for the music on 
 hold, since that is what they did with their old
 phone system.


=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

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Re: [Asterisk-Users] Grandstream Source?

You guys are a tough crowd. I do have to admit I did get this one, however. 

I don't know about Senad, but this is not an easy list to pick up on. In order to 
search the list, you have to know the terms/acronyms. In order to know the terms, you 
have to learn/ask. Many of you know this stuff back and forth. You know the 
relaionships of what-does-what. You have connected the dots and put these pieces 
together. I am still trying to get a handle on MOST all of this stuff. I can barely 
get the demo to work ;)

Let's face it, there will always be dumb questions (like most of mine). Please be nice 
and think of the many factors that can contribute. Think of knowledge and language and 
barriers. 

This */IVR/VOIP/Telephony stuff is only easy when you get to *REALY* know it. I am not 
there! I know my GNU/Linux systems... I don't know this... please be nice to me 
atleast ;)

On Wed, Sep 17, 2003 at 10:38:58AM +0100, Alastair Maw wrote:
 Senad Jordanovic wrote:
  have you more info on this free phone offer? please send it to me off the
  lest?
 
 Just as a totally wild guess, and call me crazy and amazingly 
 intelligent for thinking of it, but how about looking at www.nikotel.com?
 
 I remain astonished by how many people need constant spoon feeding...
 
 -- 
 Alastair Maw [EMAIL PROTECTED]
 MX Telecom - Systems Analyst
 http://www.mxtelecom.com
 
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RE: [Asterisk-Users] Radio for Music on Hold?

2003-09-18 Thread Bryan Nolen

In Australia this is legal provided that rights are paid to APRA (Australian
Performing Rights Assoc.) like the RIAA in the USA.

Last I checked the rates were about AU$50/line/year for music on hold (of
ANY kind)

-Bryan

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Chris Albertson
 Sent: Friday, 19 September 2003 2:25 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Radio for Music on Hold?
 
 
 
 You can't legally do this.  At least no here in the
 US.
 The music being bradcast by the station may not
 legally be re-broadcast.  Yes many people do this and
 yess thay also download copywritten MP3 files.
 
 Call any radio station and ASK THEM if it is OK don't
 trust some e-mail list.  Who knows maybe they won't
 care what you do but they do have to pay  for the
 rights to broadcast music and teir payment does
 notcover re-broadcast.
 
 Tell your client that some callers put on hold may
 know about the above and radio on hold would make
 the company look at best ignorent.
 
 Yesterday I called a company, was put on hold and
 heard the local govenment weather radio broadcast. 
 This was smart and legal too.
 
 
 
  I have a client who is interested in using a
  radio for the music on 
  hold, since that is what they did with their old
  phone system.
 
 
 =
 Chris Albertson
   Home:   310-376-1029  [EMAIL PROTECTED]
   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
   KG6OMK
 
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Re: [Asterisk-Users] Radio for Music on Hold?

2003-09-18 Thread CW_ASN

Yes, right, that is US problem. In other countries you can re-broadcast
stations. If you ask to FM stations, they answer Yes, sure! Why not?.

Following the main line of subject, in Cisco systems moh-radio uses fxo
configured as EM. Moh application dials to EM port. I don't know if * can
do this.

Regards,

Gus


- Original Message -
From: Chris Albertson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 18, 2003 1:24 PM
Subject: Re: [Asterisk-Users] Radio for Music on Hold?



 You can't legally do this.  At least no here in the
 US.
 The music being bradcast by the station may not
 legally be re-broadcast.  Yes many people do this and
 yess thay also download copywritten MP3 files.

 Call any radio station and ASK THEM if it is OK don't
 trust some e-mail list.  Who knows maybe they won't
 care what you do but they do have to pay  for the
 rights to broadcast music and teir payment does
 notcover re-broadcast.

 Tell your client that some callers put on hold may
 know about the above and radio on hold would make
 the company look at best ignorent.

 Yesterday I called a company, was put on hold and
 heard the local govenment weather radio broadcast.
 This was smart and legal too.



  I have a client who is interested in using a
  radio for the music on
  hold, since that is what they did with their old
  phone system.


 =
 Chris Albertson
   Home:   310-376-1029  [EMAIL PROTECTED]
   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
   KG6OMK

 __
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 Yahoo! SiteBuilder - Free, easy-to-use web site design software
 http://sitebuilder.yahoo.com
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List ettiquette (was Re: [Asterisk-Users] Grandstream Source?)

2003-09-18 Thread Alastair Maw

PJ Welsh wrote:

 This */IVR/VOIP/Telephony stuff is only easy when you get to *REALY*
 know it. I am not there! I know my GNU/Linux systems... I don't know
 this... please be nice to me atleast ;)
I am nice. :)

The point of that tongue-in-cheek e-mail was that hopefully Senad will 
type the single obvious word into Google next time before he wastes 
hundreds of people's time (albeit only 5 seconds each) with questions he 
can answer for himself very very easily.

VoIP is complex. PSTN systems are complex. But using Google isn't. If 
someone points out that Company Xyzzy sells a product/service, I can't 
imagine why anybody would even bother asking a mailing list about it, 
rather than just going straight to Google and searching for Xyzzy.

If you have a genuine problem, the list is friendly and nice. If someone 
can't be bothered to type a single and specific word into Google, and 
it's very obvious they haven't made an attempt to think/look for 
themselves, then it's hardly surprising that most people have little 
patience for them.

So, as a reference for all you people who get burnt when posting to the 
list, here is a guide:

 - Ask a new question by clicking the new/compose button in your
   mail client. Only hit reply if you are actually replying. In
   particular, don't hit reply, delete the whole of the subject line,
   and attempt to start a new thread this way. Stephen will flame you,
   and the rest of us with threaded mail readers will silently sit and
   seethe quietly in a corner (or miss it altogether, having marked that
   thread as uninteresting/irrelevant/don't know anything about it).
 - Don't post in HTML/RTF. Basically, it holds no advantage over plain
   text, and has many disadvantages (size, accessibility, etc, etc.)
 - Use Google if you think the question might be obvious. In particular,
   search like so to look in the list archives (e.g.):
 site:lists.digium.com SIP H323 gateway
 - If you can't find it after five minutes of looking, but still worry
   that it's quite an easy obvious question, everyone will like you lots
   if you say things like It's probably quite easy, but I can't find
   anything on Google about it unless I'm being blind...
And that's about it, really. Simple, see?

--
Alastair Maw
MX Telecom - Systems Analyst
http://www.mxtelecom.com
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Re: [Asterisk-Users] SIP registration

2003-09-18 Thread Hielke Christian Braun

Hello,


try to change  [siptestphone] to [atrg613test] in sip.conf. Maybe
that helps.

Regards,
 Christian.

On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote:
 Hi,
 
 I'm having problems letting a SIP endpoint register at Asterisk. Here's the 
 debug output from Asterisk:
 
 
 ...
 
 sip.conf:
 
 [general]
 port=5060
 bindaddr=s.s.s.s
 context=cxnet-in
 tos=lowdelay
 
 [siptestphone]
 type=friend
 user=atrg613test
 host=dynamic
 defaultip=c.c.c.c
 
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[Asterisk-Users] SIP, X-Lite

2003-09-18 Thread Lars Fredriksson

Hi folks!

I bought a X100P a while ago and know I've tried to get it working here at 
home again ... but I can't manage to get my X-Lite client working with 
Asterisk (CVS from a day ago) ...

I've downloaded the latest version of X-Lite and I believe that I've set it 
up correctly ;-) But I cant get it to register with my Asterisk - I only 
get Login timed out, contact your network admin  But, I can call 
voicemail and other SIP clients anyway - I can call voicemail and other SIP 
clients even if I enter a username that is not existing in my sip.conf???

The only error message I get in my Asterisk console is;

NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration 
from 'sip:[EMAIL PROTECTED]' failed for '192.168.5.10'


192.168.5.1 is the Asterisk server and 192.168.5.10 is my client.

Below is my sip.conf - is there anyone that can ponit out what I've done 
wrong I would be very, very, very happy ;-)
Maybe an short description in what I would enter where in the X-Lite 
configuration wouldn't hurt ...

Thanks for any help!

Best regards Lars Fredriksson, Sweden

[general]
port = 5060 ; Port to bind to
bindaddr = 192.168.5.1  ; Address to bind to
context = default   ; Default for incoming calls
;srvlookup = yes; Enable SRV lookups on outbound calls
;pedantic = yes ; Enable slow, pedantic checking for 
Pingtel
;tos=lowdelay
;tos=184
;maxexpirey=3600; Max length of incoming registration we 
allow
;defaultexpirey=120 ; Default length of incoming/outoing 
registrati
;notifymimetype=text/plain  ; Allow overriding of mime type in NOTIFY
;videosupport=yes   ; Turn on support for SIP video
;disallow=all   ; Disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc
;
;register = [EMAIL PROTECTED] ; Register with a SIP provider
;register = [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider as 
123
;

[sip7101]
context=sip
type=friend
secret=blah
auth=md5
; defaultip=192.168.5.10
host=dynamic
dtmfmode=inband
mailbox=7101
--
Using M2, Opera's revolutionary e-mail client: http://www.opera.com/m2/
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RE: [Asterisk-Users] SIP, X-Lite

2003-09-18 Thread Zac Sprackett

Add a username field to your sip.conf.

[sip7101]
context=sip
type=friend
username=7101
secret=blah
auth=md5
host=dynamic
dtmfmode=inband
mailbox=7101

-z

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Lars
 Fredriksson
 Sent: Thursday, September 18, 2003 1:21 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] SIP, X-Lite
 
 
 Hi folks!
 
 I bought a X100P a while ago and know I've tried to get it 
 working here at 
 home again ... but I can't manage to get my X-Lite client working with 
 Asterisk (CVS from a day ago) ...
 
 I've downloaded the latest version of X-Lite and I believe that 
 I've set it 
 up correctly ;-) But I cant get it to register with my Asterisk - I only 
 get Login timed out, contact your network admin  But, I can call 
 voicemail and other SIP clients anyway - I can call voicemail and 
 other SIP 
 clients even if I enter a username that is not existing in my sip.conf???
 
 The only error message I get in my Asterisk console is;
 
 NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration 
 from 'sip:[EMAIL PROTECTED]' failed for '192.168.5.10'
 
 
 192.168.5.1 is the Asterisk server and 192.168.5.10 is my client.
 
 Below is my sip.conf - is there anyone that can ponit out what I've done 
 wrong I would be very, very, very happy ;-)
 Maybe an short description in what I would enter where in the X-Lite 
 configuration wouldn't hurt ...
 
 Thanks for any help!
 
 
 Best regards Lars Fredriksson, Sweden
 
 [general]
 port = 5060 ; Port to bind to
 bindaddr = 192.168.5.1  ; Address to bind to
 context = default   ; Default for incoming calls
 ;srvlookup = yes; Enable SRV lookups on outbound calls
 ;pedantic = yes ; Enable slow, pedantic checking for 
 Pingtel
 ;tos=lowdelay
 ;tos=184
 ;maxexpirey=3600; Max length of incoming registration we 
 allow
 ;defaultexpirey=120 ; Default length of incoming/outoing 
 registrati
 ;notifymimetype=text/plain  ; Allow overriding of mime type in NOTIFY
 ;videosupport=yes   ; Turn on support for SIP video
 ;disallow=all   ; Disallow all codecs
 ;allow=ulaw ; Allow codecs in order of preference
 ;allow=ilbc
 ;
 ;register = [EMAIL PROTECTED] ; Register with a SIP provider
 ;register = [EMAIL PROTECTED]/1234 ; Register 2345 at sip 
 provider as 
 123
 ;
 
 [sip7101]
 context=sip
 type=friend
 secret=blah
 auth=md5
 ; defaultip=192.168.5.10
 host=dynamic
 dtmfmode=inband
 mailbox=7101
 
 
 -- 
 Using M2, Opera's revolutionary e-mail client: http://www.opera.com/m2/
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Re: [Asterisk-Users] SIP, X-Lite

2003-09-18 Thread Eric Wieling

See my changes below

On Thu, 2003-09-18 at 12:20, Lars Fredriksson wrote:

[7101]
context=sip
type=friend
host=dynamic
dtmfmode=inband
mailbox=7101
-- 
Sample configs and more: http://www.fnords.org/~eric/asterisk/

BTEL Consulting
+1-850-484-4535 x2111 (Pensacola)
+1-504-595-3916 x2111 (New Orleans)
+1-877-677-9643 x2111 (Toll Free)

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Re: [Asterisk-Users] Grandstream Source?

On Thu, 2003-09-18 at 11:33, PJ Welsh wrote:
 You guys are a tough crowd. I do have to admit I did get this one,
 however. 
 
 I don't know about Senad, but this is not an easy list to pick up on.
 In order to search the list, you have to know the terms/acronyms. In
 order to know the terms, you have to learn/ask. Many of you know this
 stuff back and forth. You know the relaionships of what-does-what. You
 have connected the dots and put these pieces together. I am still
 trying to get a handle on MOST all of this stuff. I can barely get the
 demo to work ;)
 
 Let's face it, there will always be dumb questions (like most of
 mine). Please be nice and think of the many factors that can
 contribute. Think of knowledge and language and barriers. 

To help bridge the gap from the other side of the knowledge gap, I'd
love it if people would read at least the introduction to this page..
http://www.catb.org/~esr/faqs/smart-questions.html

I hate to throw that link out to often because people tend to start
considering it rude also. 

After reading it again myself and reading this quote...

Indeed, one of my major complaints about the computer field is that
whereas Newton could say, If I have seen a little farther than others,
it is because I have stood on the shoulders of giants, I am forced to
say, Today we stand on each other's feet. Perhaps the central problem
we face in all of computer science is how we are to get to the
situation
where we build on top of the work of others rather than redoing so much
of it in a trivially different way. Science is supposed to be
cumulative,
not almost endless duplication of the same kind of things.

-- Richard W. Hamming, One Man's View of Computer Science, 1968
Turing
Award Lecture, quoting from Sir Issac Newton's letter to Robert Hooke,
February 5, 1675/76. See ACM Turing Award Lectures: the First Twenty
Years: 1966-1985. (ACM Press. 1987). See also this 1986 talk.


I see where the hacker culture does not always lend itself to the
advancement of the cause as much as advancement of the individuals in
the cause.

I regularly have to point out to people who ask me questions that when I
ask them to think about their problem and ask them questions that point
them in the right direction of figuring out the answer for themselves
that I have helped them advance themselves. My family included do not
always like the fact that I don't always answer questions with facts,
but pointed questions to make them solve their own problems. Its funny
how good teachers do the same thing, and all of them are considered hard
and not always liked. These kinds of teachers though are the ones who
get you farther in life.  

So in conclusion, ask your question when you need help, think about what
it is exactly you need to know, and do not take it as a personal attack
if there is a comment made about how to solve the problem yourself. 


-- 
Steven Critchfield  [EMAIL PROTECTED]

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Re: [Asterisk-Users] VoicePulse offering IAX2 services

2003-09-18 Thread tom


Interesting that they use IAX2, but 2.9 cents/minute seems kind of high
for a wholesale rate, especially in the lower 48. I'm shopping for a
good wholesaler right now. 

Regards,

Tom

On Thu, 2003-09-18 at 08:44, Peter Pauly wrote:
 I don't know if this has been mentioned yet:
 
 Voicepulse is now offering wholesale pricing and
 IAX2 connectivity for Asterisk users. No fees, pay 
 as you go. They also
 offer incoming calls for $7.99 per month. See
 wholesale.voicepulse.com.
 
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Re: List ettiquette (was Re: [Asterisk-Users] Grandstream Source?)

On Thu, 2003-09-18 at 12:03, Alastair Maw wrote:
   - Ask a new question by clicking the new/compose button in your
 mail client. Only hit reply if you are actually replying. In
 particular, don't hit reply, delete the whole of the subject line,
 and attempt to start a new thread this way. Stephen will flame you,
 and the rest of us with threaded mail readers will silently sit and
 seethe quietly in a corner (or miss it altogether, having marked that
 thread as uninteresting/irrelevant/don't know anything about it).
 
   - Don't post in HTML/RTF. Basically, it holds no advantage over plain
 text, and has many disadvantages (size, accessibility, etc, etc.)

I'm getting better about this. I am only including an introductory flame
if I answer the question, else I ignore the message. 

But to reiterate the second point. I may be alone, or I may be part of a
larger group, but I rarely will read a message that is in HTML unless it
had an interesting subject line. And then if it is difficult to read
because of the HTML, it will quickly get ignored.
-- 
Steven Critchfield  [EMAIL PROTECTED]

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RE: [Asterisk-Users] Grandstream Source?

I did not read the comment made by Alistair hence why I am replying to it
now.
And thanks, to PJ Welsh for bringing it up. Your points are true, valid and
I am sure most people will agree with you. (Even the old timers where
newbies at some stage)!

Full VOIP understanding takes time. There are many little pieces of
information to know in order
for all of it to make sense. A month ago, I did not know what E1/T1 is, let
alone all complexity associated with VOIP.
However, personally I am determined to get there as long it takes.

Mark, all people at Digium and all members of this community should only
benefit by having people getting interest in *. We all know how hard is to
sell something to somebody, and I for one will support Digium, by buying its
hardware as my part of my two cents and appreciation of the *. Thanks
guys.

In regards, to my question to Michael Koehler, the question was directed
directly at Michael, presuming Michael has more info or is connected with
Nikotel in same way and not asking anyone else in particular to make their
comments.
I REPLIED to that thread, I certainly did not CREATE new thread!!!

If that was not understood by Alistair, than it is true that some people
even when spoon fed still do not understand what they just read!!!

Also, guys thank you all for your support you have offered and given so far.

Senad


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Re: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread Dave Cotton

On Thu, 2003-09-18 at 19:44, Steven Critchfield wrote:

 I regularly have to point out to people who ask me questions that when I
 ask them to think about their problem and ask them questions that point
 them in the right direction of figuring out the answer for themselves
 that I have helped them advance themselves. My family included do not
 always like the fact that I don't always answer questions with facts,
 but pointed questions to make them solve their own problems. Its funny
 how good teachers do the same thing, and all of them are considered hard
 and not always liked. These kinds of teachers though are the ones who
 get you farther in life.  

Absolutely agree with you Steve.  I left teachers training college in
1970. I shock some teachers when I said that in all the years since I
haven't taught anyone anything. I've just enabled them to learn.
The problem is that in most national education systems the teacher is
expected to provide the answers to pass some test at the end of the
course. Thinking is not part of the curriculum.
-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] Cisco 7960 + 5.x Firmware + *

2003-09-18 Thread Brian West

Ya I just got my phone and upgraded it to 5.3 without a problem.  It works
perfect with *

bkw

On Wed, 17 Sep 2003, Travis Johnson wrote:

 Yes. 30 phones in production environment. No problems so far. :)

 Travis


 At 08:21 PM 9/17/2003 -0500, you wrote:
 Anyone running the 5.x firmware on their 7960's with asterisk?
 
 bkw
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Re: [Asterisk-Users] phpconfig is out in CVS

2003-09-18 Thread Rainer Jochem


Hi,

we've written a small set of php scripts and web pages
to give users the possibility of changing their 
SIP-Passwords in addition with the usage of MD5-Hashes
as secret. The md5secrets are stored in a MySQL-database
to avoid trouble if several users update their passwords
the same time. 
We've alse written a small perl script which checks for 
changes between database and sip.conf and - if neccessary - 
it updates the sip.conf and reloads asterisk.


The idea with the md5 hashes was just to prevent the
admin of the asterisk-server seeing the user's 
secrets in plain-text. 
And as the MD5-hashes are needed for SIP anyway
we just did this patch.

I've posted it already this evening on -dev and put
it in the bugtracker:

http://bugs.digium.com/bug_view_page.php?bug_id=288




Cheers,
 
 Rainer

-- 
http://graphics.cs.uni-sb.de/VoIP/

pgp0.pgp
Description: PGP signature

Re: [Asterisk-Users] Grandstream Source?

I have to defend us newbies on this.

This environment does not facilitate sequential knowledge building! Based on my entry 
to Asterisk, I should have already known 
T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you get the idea 
(still trying to figure out skinny...cisco something, I know). Heck, I'm struggling 
to get a grip on what and how to use/confiure SIP for linux and keep my hair. 

You don't start off with a prerequisite of knowledge to join like a class/school. You 
don't have the you-must-have-asterisk-101-before going to asterisk-102 before you can 
join this list. You have a forum that is GENERAL.

I would like to a better effort to provide a more sensible way to start helping us 
newbies. I have to say that the Digium handbook helped a little, but not much. I have 
googled till I couldn't see straight. I just don't yet have the big picture that 
most of you do. I couldn't even tell you if I need a channel bank or a channel changer 
;) at this point.

A group of you seem to expect people to have a knowledge base that allows for entering 
keywords to google. I don't know those keywords. You know the context to search for 
when someone says I'm having a problem with insert-thing-here.

Instead of the usual, Search the archives. It would be more helpfull to give a hint 
on what to search for. I could search for SIP and get back several hundred answers. 
Then I have to figure out where that answer lies in the series of possible answers. 
Then I have to somehow figure out if it works.

As most of you teachers (past and present) should know, not all of us learn the same. 
Some people just get written material. Some NEED the spoon to make it to the next 
level. Some need the hands-on experience and other's just can't learn any more than 
they have already know(those people are not likely on this list, however).

You do realize that the http://www.asterisk.org/index.php?menu=support lists the 
mailing list first for support, don't you. In fact, you have to go to the second page 
before you even see the google reference. More a few people tend to look for the FIRST 
way to get help not ALL ways to get help...

flame suit on


On Thu, Sep 18, 2003 at 08:31:59PM +0200, Dave Cotton wrote:
...
 Absolutely agree with you Steve.  I left teachers training college in
 1970. I shock some teachers when I said that in all the years since I
 haven't taught anyone anything. I've just enabled them to learn.
 The problem is that in most national education systems the teacher is
 expected to provide the answers to pass some test at the end of the
 course. Thinking is not part of the curriculum.
 -- 
 Dave Cotton [EMAIL PROTECTED]
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RE: [Asterisk-Users] [Release] Skinny Support in cvs

2003-09-18 Thread Dan Austin

I grabbed the latest CVS (this morning).  Recompiled and installed
everything.

Registration works fine.
Calls to Zap and SIP phones almost work.  I get one-way audio from those
channels.
Calls from Zap also result in one-way audio.
To be clear, audio from the skinny phone can not be heard.  Audio from
the other
channels comes through to the skinny phone just fine.
Calls from SIP continue to ring after the Skinny phone answers

I have not added another skinny phone yet, but I will do so this
afternoon.

I haven't added this to the bug list, since it is likely an issue with
my config,
but I have not strayed to far from the samples.


Thanks for the cool feature.
Dan
-Original Message-
From: Florian Overkamp [mailto:[EMAIL PROTECTED] 
Sent: Thursday, September 18, 2003 1:11 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] [Release] Skinny Support in cvs


At 16:25 17-9-2003 -0700, you wrote:
So I've been trying to pay attention, but I hadn't seen any updates on 
SourceForge.

I inferred from the thread I could get a copy using CVS, but it looks 
like our firewall is keeping me out of CVS.  Is there another way to 
come by the source?

Dan,

you can now find it in the channels directory of your mainstream
asterisk 
cvs (brought to you by digium)

Florian

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Re: [Asterisk-Users] Skinny + XMLDefault

2003-09-18 Thread Alexander Noack

Thanks Eric, but I do have the 7960 configuered to find its
tftp-server. Thanks to the example XMLDefault.cnf.xml it finds its
callmanager along with the phone images too.

What I am trying to do is to get all the buttons working, and hopefully
direct the phone to some useful Service/Directory/Idle URL. My hopes
are high to somehow place that information in the Config-XML-file. (I
know it can be done using the SIP-Images)

for reference, I put the relevant part of the XMLDefault.cnf.xml here,
as I found it in the chan_sccp source:

Default
 callManagerGroup
  members
   member  priority=0
callManager
 ports
  ethernetPhonePort2001/ethernetPhonePort
 /ports
 processNodeName139.30.208.29/processNodeName
/callManager
   /member
   member  priority=1
callManager
 ports
  ethernetPhonePort2000/ethernetPhonePort
 /ports
 processNodeName139.30.208.29/processNodeName
/callManager
   /member
  /members
 /callManagerGroup
 loadInformation7 model=IP Phone
7960P00303020214/loadInformation7
 loadInformation124 model=Addon
7914S00103020002.bin/loadInformation124
/Default

I am actually using chan_skinny and chan_sccp at the same time (ports
2001 + 2000)...

Again, thanks for any suggestions!

Alex



This is included for reference:
 If your Asterisk server is on the same machine as your DHCP
 server then you should not need the .cnf file.  My 7910
 (running SCCP/Skinny) finds my Asterisk server just fine.
 If your DHCP server is not running on the same machine as
 your Asterisk server you may fine this URL to be helpful.
 If talks about how the SCCP phones find their Call Manager.

 I've spent two days trying to find out something about the
 format of the default configuration file, which CCM
 produces. The only example I have so far is the one from
 the chan_sccp source.

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[Asterisk-Users] New message 0 in mailbox 7606

2003-09-18 Thread noc

Hello,

I recently started playing with voicemail2.  I'm having two minor problems that I 
can't seem to find discussed in the archives.

1) New message 0 in mailbox 7606.  New voice mail message count seems to start with 0 
for the first new message instead of 1.  Any tricks to fix this?

2) When listening to messages with VoicemailMain2, the time stamp is in GMT and not 
corrected for the local time zone offset.  But the email that voicemail2 sends has the 
correct time.  I added |tz=eastern to the end of the mailbox definitions in 
voicemail.conf, but that did not seem to fix the problem.

Any thoughts on these two problems?  I'm running a recent CVS from 9/14/03.

Thanks,

-Jon


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[Asterisk-Users] Hanging up one call when you have call waiting

2003-09-18 Thread jerk face

I would like to do the following:

A calls B
C calls A
A hears call waiting beep and flashes the line to
talk to C
::Here's where I run into a problem::
A hangs up on C and immediately returns to a
conversation with B

The only way I have got this to work is if C hangs
up.  Then A is connected to B.  If I hit flash for
a second time, then it becomes a three way call.

Is there a key sequence to hang up one of the calls?

Thank you for your time.

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RE: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread Wade J. Weppler

I agree.  I was in exactly the same spot as you just over a year ago.  I
jumped into Asterisk without any idea of what any of the terms you
mention mean.

I vowed to setup a FAQ for users in my position, but now that I'm knee
deep in it, it's hard to put myself back into that mindset and decide
what's necessary and what isn't.

As you're currently in that position, I'd be more than happy to answer a
set of questions, and post them as a newbie-FAQ.

-wade


 I have to defend us newbies on this.
 
 This environment does not facilitate sequential knowledge building!
Based
 on my entry to Asterisk, I should have already known
 T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you
get
 the idea (still trying to figure out skinny...cisco something, I
know).
 Heck, I'm struggling to get a grip on what and how to use/confiure SIP
for
 linux and keep my hair.
 
 You don't start off with a prerequisite of knowledge to join like a
 class/school. You don't have the you-must-have-asterisk-101-before
going
 to asterisk-102 before you can join this list. You have a forum that
is
 GENERAL.
 
 I would like to a better effort to provide a more sensible way to
start
 helping us newbies. I have to say that the Digium handbook helped a
 little, but not much. I have googled till I couldn't see straight. I
just
 don't yet have the big picture that most of you do. I couldn't even
tell
 you if I need a channel bank or a channel changer ;) at this point.
 
 A group of you seem to expect people to have a knowledge base that
allows
 for entering keywords to google. I don't know those keywords. You know
the
 context to search for when someone says I'm having a problem with
insert-
 thing-here.
 
 Instead of the usual, Search the archives. It would be more helpfull
to
 give a hint on what to search for. I could search for SIP and get back
 several hundred answers. Then I have to figure out where that answer
 lies in the series of possible answers. Then I have to somehow figure
out
 if it works.
 
 As most of you teachers (past and present) should know, not all of us
 learn the same. Some people just get written material. Some NEED the
 spoon to make it to the next level. Some need the hands-on
experience
 and other's just can't learn any more than they have already
know(those
 people are not likely on this list, however).
 
 You do realize that the http://www.asterisk.org/index.php?menu=support
 lists the mailing list first for support, don't you. In fact, you have
to
 go to the second page before you even see the google reference. More a
few
 people tend to look for the FIRST way to get help not ALL ways to get
 help...
 
 flame suit on
 
 
 On Thu, Sep 18, 2003 at 08:31:59PM +0200, Dave Cotton wrote:
 ...
  Absolutely agree with you Steve.  I left teachers training college
in
  1970. I shock some teachers when I said that in all the years since
I
  haven't taught anyone anything. I've just enabled them to learn.
  The problem is that in most national education systems the teacher
is
  expected to provide the answers to pass some test at the end of the
  course. Thinking is not part of the curriculum.
  --
  Dave Cotton [EMAIL PROTECTED]
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Re: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread Olle E. Johansson

PJ Welsh wrote:

I have to defend us newbies on this.

This environment does not facilitate sequential knowledge building! 
You do realize that the http://www.asterisk.org/index.php?menu=support lists the mailing list first for support, don't you. In fact, you have to go to the second page before you even see the google reference. More a few people tend to look for the FIRST way to get help not ALL ways to get help...
PJ,
I realized the same and started a process to collect a lot of that information and 
build a
knowledge base on http://www.voip-forum.org/
Click on Asterisk on the home page and you'll find a lot of information. On that 
web, you'll also
find information I gathered about the rest of the telecom stuff I didn't know anything 
about.
So have others. There's plenty of pages with facts, explanations and pointers to find 
there.
It's a start, please help us helping other newcomers by adding stuff, questions and 
keywords
you don't know. If you haven't got an explanation, create a page named by the term you
don't now and simply add What's a pyroflax? on it. Someone will notice and explain
what a pyroflax is...
The environment surrounding the Asterisk Open Source project is built by all of us.
Now, you're part of this environment. Welcome!
/Olle
...still learning and trying to understand FXO, ISUPs, RDNIS and other terms...
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Re: [Asterisk-Users] New message 0 in mailbox 7606

On Thu, 2003-09-18 at 14:37, noc wrote:

I don't use VM2 yet, but lets see if I can answer a couple of questions.

 I recently started playing with voicemail2.  I'm having two minor
 problems that I can't seem to find discussed in the archives.
 
 1) New message 0 in mailbox 7606.  New voice mail message count seems
 to start with 0 for the first new message instead of 1.  Any tricks to
 fix this?

Is this what is read to you, or is this a storage question? C
programmers usually start counting from 0. Maybe a slip up to not add 1
before presentation. 

 2) When listening to messages with VoicemailMain2, the time stamp is
 in GMT and not corrected for the local time zone offset.  But the
 email that voicemail2 sends has the correct time.  I added
 |tz=eastern to the end of the mailbox definitions in voicemail.conf,
 but that did not seem to fix the problem.

I think Mark added a patch this morning at 8:30am -5 to default to
grabbing system timezone.

 Any thoughts on these two problems?  I'm running a recent CVS from
 9/14/03.

Gasp, you are running four day old software, what will the neighbors
think ;)

As usual, if you seem to have a problem, please upgrade to current and
see if the problem persists. If it does persist, then ask for help or
confirmation of the problem. 
-- 
Steven Critchfield  [EMAIL PROTECTED]

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Re: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread Ariel Batista

I just want to thank you very much PJ Welsh for saying something I have wanted to say. 
 And your right this is suppose to be the place to get help.  I am new to Asterisk and 
I am learning the hard way.  There have been some people here thinking that we are all 
programers or 100% Linux types.  The list said user's.  I am a user of the system I 
got this system installed and it's hard to configure it all!  I am learning but there 
is no real help file!  Some of us are using this system in the real world and would 
like help with it!  It's not a toy.  The only way that this system will grow is with 
good support!  And at present it's very hard to get support or there is no support! I 
can see the future is going to be with something like Asterisk why not let it be 
Asterisk.   

Again thank you for your comments.

-- Original Message --
From: PJ Welsh [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date:  Thu, 18 Sep 2003 14:17:17 -0500

I have to defend us newbies on this.

This environment does not facilitate sequential knowledge building! Based on my entry 
to Asterisk, I should have already known 
T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you get the idea 
(still trying to figure out skinny...cisco something, I know). Heck, I'm struggling 
to get a grip on what and how to use/confiure SIP for linux and keep my hair. 

You don't start off with a prerequisite of knowledge to join like a class/school. You 
don't have the you-must-have-asterisk-101-before going to asterisk-102 before you can 
join this list. You have a forum that is GENERAL.

I would like to a better effort to provide a more sensible way to start helping us 
newbies. I have to say that the Digium handbook helped a little, but not much. I have 
googled till I couldn't see straight. I just don't yet have the big picture that 
most of you do. I couldn't even tell you if I need a channel bank or a channel 
changer ;) at this point.

A group of you seem to expect people to have a knowledge base that allows for 
entering keywords to google. I don't know those keywords. You know the context to 
search for when someone says I'm having a problem with insert-thing-here.

Instead of the usual, Search the archives. It would be more helpfull to give a hint 
on what to search for. I could search for SIP and get back several hundred answers. 
Then I have to figure out where that answer lies in the series of possible answers. 
Then I have to somehow figure out if it works.

As most of you teachers (past and present) should know, not all of us learn the same. 
Some people just get written material. Some NEED the spoon to make it to the next 
level. Some need the hands-on experience and other's just can't learn any more than 
they have already know(those people are not likely on this list, however).

You do realize that the http://www.asterisk.org/index.php?menu=support lists the 
mailing list first for support, don't you. In fact, you have to go to the second page 
before you even see the google reference. More a few people tend to look for the 
FIRST way to get help not ALL ways to get help...

flame suit on


On Thu, Sep 18, 2003 at 08:31:59PM +0200, Dave Cotton wrote:
...
 Absolutely agree with you Steve.  I left teachers training college in
 1970. I shock some teachers when I said that in all the years since I
 haven't taught anyone anything. I've just enabled them to learn.
 The problem is that in most national education systems the teacher is
 expected to provide the answers to pass some test at the end of the
 course. Thinking is not part of the curriculum.
 -- 
 Dave Cotton [EMAIL PROTECTED]
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[Asterisk-Users] Cisco 7910 w/SCCP

2003-09-18 Thread Christopher J. Wolff

Has anyone managed to get a 7910 working with * through SCCP?

Regards,
Christopher J. Wolff, VP, CIO
Broadband Laboratories
http://www.bblabs.com


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Re: [Asterisk-Users] ADSI Vista/Aastra 350

2003-09-18 Thread Armand A. Verstappen

Hi,

On Wed, 2003-09-10 at 15:50, Matthew M. Gamble wrote:
 I have ADSI working on my Aastra (Vista/Nortel) 350 phone and everything is
 working fine.
 
 However, I want the asterisk.adsi to load into the 'self-load' slot but
 can't figure out what the correct FDN for doing this is.  Does anyone know
 the right FDN for the SL slot on these phones?

I have hammered Aastra support for three weeks, and finally got the
answer that it is impossible to load something in the self-load slot,
unless I would buy a custom model.
This is utter nonsens, as dialing the webconfig number will happily load
a script into the self-load slot. I have yet to receive reply to that
remark, and I'm confident by now that I never will.

So, the only solution I see is trying different fdn's untill you hit the
jackpot. I haven't found the motivation to do that. If you do, and do
find the right FDN, please let us know.

As I'm disgusted by Aastra's approach to this issue, I'm looking for
other ADSI phones that will allow me to load the self-load slot.
Suggestions, anyone?

wkr,

-- 
Envida http://www.envida.net/
Armand A. Verstappen   Graadt van Roggenweg 328
[EMAIL PROTECTED]   3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht


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Re: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread Rich Adamson


 I realized the same and started a process to collect a lot of that information and 
 build a
 knowledge base on http://www.voip-forum.org/

The url does not seem to respond. Are you sure its up and working?



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Re: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread Steve Totaro

It seems we have a mailing list:

THE NATURAL LIFE CYCLE OF MAILING LISTS

Every list seems to go through the same cycle:

1. Initial enthusiasm (people introduce themselves, and gush a lot about how
wonderful it is to find kindred souls).

2. Evangelism (people moan about how few folks are posting to the list, and
brainstorm recruitment strategies).

3. Growth (more and more people join, more and more lengthy threads develop,
occasional off-topic threads pop up)

4. Community (lots of threads, some more relevant than others; lots of
information and advice is exchanged; experts help other
experts as well as less experienced colleagues; friendships develop; people
tease each other; newcomers are welcomed with
generosity and patience; everyone---newbie and expert alike---feels
comfortable asking questions, suggesting answers, and
sharing opinions)

5. Discomfort with diversity (the number of messages increases dramatically;
not every thread is fascinating to every
reader; people start complaining about the signal-to-noise ratio; person 1
threatens to quit if *other* people don't
limit discussion to person 1's pet topic; person 2 agrees with person 1;
person 3 tells 1  2 to lighten up; more
bandwidth is wasted complaining about off-topic threads than is used for the
threads themselves; everyone gets
annoyed)

6a. Smug complacency and stagnation (the purists flame everyone who asks an
'old' question or responds with humor to a serious post; newbies are
rebuffed; traffic drops to a doze-producing level of a few minor issues; all
interesting discussions happen by private email and are limited to a few
participants; the purists spend lots of time self-righteously congratulating
each other on keeping off-topic threads off the list)

OR

6b. Maturity (a few people quit in a huff; the rest of the participants stay
near stage 4, with stage 5 popping up briefly
every few weeks; many people wear out their second or third 'delete' key,
but the list lives contentedly ever after)

- Original Message -
From: Wade J. Weppler [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 18, 2003 12:40 PM
Subject: RE: [Asterisk-Users] Grandstream Source?


I agree.  I was in exactly the same spot as you just over a year ago.  I
jumped into Asterisk without any idea of what any of the terms you
mention mean.

I vowed to setup a FAQ for users in my position, but now that I'm knee
deep in it, it's hard to put myself back into that mindset and decide
what's necessary and what isn't.

As you're currently in that position, I'd be more than happy to answer a
set of questions, and post them as a newbie-FAQ.

-wade


 I have to defend us newbies on this.

 This environment does not facilitate sequential knowledge building!
Based
 on my entry to Asterisk, I should have already known
 T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you
get
 the idea (still trying to figure out skinny...cisco something, I
know).
 Heck, I'm struggling to get a grip on what and how to use/confiure SIP
for
 linux and keep my hair.

 You don't start off with a prerequisite of knowledge to join like a
 class/school. You don't have the you-must-have-asterisk-101-before
going
 to asterisk-102 before you can join this list. You have a forum that
is
 GENERAL.

 I would like to a better effort to provide a more sensible way to
start
 helping us newbies. I have to say that the Digium handbook helped a
 little, but not much. I have googled till I couldn't see straight. I
just
 don't yet have the big picture that most of you do. I couldn't even
tell
 you if I need a channel bank or a channel changer ;) at this point.

 A group of you seem to expect people to have a knowledge base that
allows
 for entering keywords to google. I don't know those keywords. You know
the
 context to search for when someone says I'm having a problem with
insert-
 thing-here.

 Instead of the usual, Search the archives. It would be more helpfull
to
 give a hint on what to search for. I could search for SIP and get back
 several hundred answers. Then I have to figure out where that answer
 lies in the series of possible answers. Then I have to somehow figure
out
 if it works.

 As most of you teachers (past and present) should know, not all of us
 learn the same. Some people just get written material. Some NEED the
 spoon to make it to the next level. Some need the hands-on
experience
 and other's just can't learn any more than they have already
know(those
 people are not likely on this list, however).

 You do realize that the http://www.asterisk.org/index.php?menu=support
 lists the mailing list first for support, don't you. In fact, you have
to
 go to the second page before you even see the google reference. More a
few
 people tend to look for the FIRST way to get help not ALL ways to get
 help...

 flame suit on


 On Thu, Sep 18, 2003 at 08:31:59PM +0200, Dave Cotton wrote:
 ...
  Absolutely agree with you Steve.  I left teachers training college
in

Re: [Asterisk-Users] Grandstream Source?

Sorry about changing the original incorrect subject of Re: [Asterisk-Users]
Grandstream Source? . Many have already written that thread off and this may be a
good place to start on a positive note.

Yes, I forgot to mention some of the sites that I have found usefull. I do have to say
that http://www.voip-forum.org/ has been a very good resource!

Keywords: newbie help support search google documentation links spoon feed

So, I would say that these are some sites of interest in no real order:

http://www.voip-forum.org/
http://www.asterisk.org/index.php?menu=support
http://www.fnords.org/~eric/asterisk/
http://asterisk.gnuinter.net/
http://megaglobal.net/docs/asterisk/html/
http://home.cogeco.ca/~camstuff/
http://www.wwworks-inc.com/asterisk/
http://www.google.com/custom?q=sa=Google+Searchcof=LW%3A40%3BL%3Ahttp%3A%2F%2Fwww.asterisk.org%2Fimages%2Ftopics%2Fasterisk.png%3BLH%3A40%3B%0D%0AAH%3Acenter%3BGL%3A0%3BS%3Ahttp%3A%2F%2Fwww.AsteriskPBX.org%3BAWFID%3Ad7bc203313616854%3Bdomains=www.marko.netsitesearch=www.marko.net

Please feel free to add to this list

On Thu, Sep 18, 2003 at 09:53:44PM +0200, Olle E. Johansson wrote:
I realized the same and started a process to collect a lot of that information and
build a
knowledge base on http://www.voip-forum.org/

Click on Asterisk on the home page and you'll find a lot of information. On that
web, you'll also
find information I gathered about the rest of the telecom stuff I didn't know
anything about.
So have others. There's plenty of pages with facts, explanations and pointers to
find there.

It's a start, please help us helping other newcomers by adding stuff, questions and
keywords
you don't know. If you haven't got an explanation, create a page named by the term
you
don't now and simply add What's a pyroflax? on it. Someone will notice and explain
what a pyroflax is...

The environment surrounding the Asterisk Open Source project is built by all of us.
Now, you're part of this environment. Welcome!

/Olle
...still learning and trying to understand FXO, ISUPs, RDNIS and other terms...

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Re: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread John Vozza

BS! :)

Take the time to read and learn as much as you can from what's available
and believe it or not you may just learn something. Even if that something
is what to ask/search for.

All those that get paid to answer questions on this list please raise your
hand. I know my hand is still on the keyboard.

I always amazes me how so many EXPECT so much for nothing...

Regards

John
-
NetRom Internet Services973-208-1339 voice
[EMAIL PROTECTED]   973-208-0942 fax
http://www.netrom.com
-


On Thu, 18 Sep 2003, PJ Welsh wrote:

 I have to defend us newbies on this.

 This environment does not facilitate sequential knowledge building! Based on my 
 entry to Asterisk, I should have already known 
 T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you get the idea 
 (still trying to figure out skinny...cisco something, I know). Heck, I'm 
 struggling to get a grip on what and how to use/confiure SIP for linux and keep my 
 hair.

 You don't start off with a prerequisite of knowledge to join like a class/school. 
 You don't have the you-must-have-asterisk-101-before going to asterisk-102 before 
 you can join this list. You have a forum that is GENERAL.

 I would like to a better effort to provide a more sensible way to start helping us 
 newbies. I have to say that the Digium handbook helped a little, but not much. I 
 have googled till I couldn't see straight. I just don't yet have the big picture 
 that most of you do. I couldn't even tell you if I need a channel bank or a channel 
 changer ;) at this point.

 A group of you seem to expect people to have a knowledge base that allows for 
 entering keywords to google. I don't know those keywords. You know the context to 
 search for when someone says I'm having a problem with insert-thing-here.

 Instead of the usual, Search the archives. It would be more helpfull to give a 
 hint on what to search for. I could search for SIP and get back several hundred 
 answers. Then I have to figure out where that answer lies in the series of 
 possible answers. Then I have to somehow figure out if it works.

 As most of you teachers (past and present) should know, not all of us learn the 
 same. Some people just get written material. Some NEED the spoon to make it to 
 the next level. Some need the hands-on experience and other's just can't learn any 
 more than they have already know(those people are not likely on this list, however).

 You do realize that the http://www.asterisk.org/index.php?menu=support lists the 
 mailing list first for support, don't you. In fact, you have to go to the second 
 page before you even see the google reference. More a few people tend to look for 
 the FIRST way to get help not ALL ways to get help...

 flame suit on


 On Thu, Sep 18, 2003 at 08:31:59PM +0200, Dave Cotton wrote:
 ...
  Absolutely agree with you Steve.  I left teachers training college in
  1970. I shock some teachers when I said that in all the years since I
  haven't taught anyone anything. I've just enabled them to learn.
  The problem is that in most national education systems the teacher is
  expected to provide the answers to pass some test at the end of the
  course. Thinking is not part of the curriculum.
  --
  Dave Cotton [EMAIL PROTECTED]
 ___
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Re: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread Sean P. Robertson

Maybe you are not talking about the same place, but I thought that
http://www.voip-info.org was the Wiki.

Sean
___

Sean Robertson

NETXUSA
p. 800-289-6389
f.  864-233-4344  Ask me about Voice over IP.
http://www.netxusa.com/

- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 18, 2003 5:15 PM
Subject: Re: [Asterisk-Users] Grandstream Source?



  I realized the same and started a process to collect a lot of that
information and build a
  knowledge base on http://www.voip-forum.org/

 The url does not seem to respond. Are you sure its up and working?



 ___
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Re: [Asterisk-Users] New message 0 in mailbox 7606

2003-09-18 Thread Tilghman Lesher

On Thursday 18 September 2003 14:37, noc wrote:
 2) When listening to messages with VoicemailMain2, the time stamp is
 in GMT and not corrected for the local time zone offset.  But the
 email that voicemail2 sends has the correct time.  I added
 |tz=eastern to the end of the mailbox definitions in
 voicemail.conf, but that did not seem to fix the problem.

Please send more information about your configuration.  Include details
like distribution version and the contents of voicemail.conf.

This isn't a problem on either Mandrake or Slackware, and since Critch
isn't complaining, probably not on Debian, either.  However, I'm working
with somebody who has this problem on RedHat 8.

-Tilghman

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RE: [Asterisk-Users] Radio for Music on Hold?

2003-09-18 Thread Paul Crick

 Tell your client that some callers put on hold may
 know about the above and radio on hold would make
 the company look at best ignorent.
I read something somewhere.. can't remember where.. some PBX buyer's guide
maybe? ANYWAY.. point is.. it sounds bad to the callers.. and you never know
what they're hearing.. dodgy music, a DJ going off on one, throwing a fit,
an advert for a competitor or something else inappropriate..

Come on people! Fork out $50 for a discman and another few bucks for some
royalty free library music and have that on hold instead.. You're in
control, you know what your callers are listening to, and you're also legal
:-)

Oh yeah.. we're talking Asterisk.. the physical connection to an external
source is what sparked this whole thread off.. sorry, my bad - I forgot..
ok, forget the discman, fork out for the music, rip it to MP3 and use the
built in MOH solution?

Or.. are we still talking about the MOH being the output of the radio
station that's actually being called, that's using Asterisk as its PBX?

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Re: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread Tilghman Lesher

On Thursday 18 September 2003 16:15, Rich Adamson wrote:
  I realized the same and started a process to collect a lot of that
  information and build a knowledge base on
  http://www.voip-forum.org/

 The url does not seem to respond. Are you sure its up and working?

That should be .com, not .org.

-Tilghman

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Re: [Asterisk-Users] Grandstream Source?

On Thu, 2003-09-18 at 16:15, Rich Adamson wrote:
  I realized the same and started a process to collect a lot of that information and 
  build a
  knowledge base on http://www.voip-forum.org/
 
 The url does not seem to respond. Are you sure its up and working?

hmm, looks like verisign is broken.

try 
http://www.voip-info.org/ 
-- 
Steven Critchfield  [EMAIL PROTECTED]

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Re: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread Steve Totaro

then ignore the thread.

to use your words...I always amazes me how so many EXPECT so much for
nothing...

- Original Message -
From: John Vozza [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 18, 2003 1:19 PM
Subject: Re: [Asterisk-Users] Grandstream Source?

 BS! :)

 Take the time to read and learn as much as you can from what's available
 and believe it or not you may just learn something. Even if that something
 is what to ask/search for.

 All those that get paid to answer questions on this list please raise your
 hand. I know my hand is still on the keyboard.

 I always amazes me how so many EXPECT so much for nothing...

 Regards

 John
 -
 NetRom Internet Services 973-208-1339 voice
 [EMAIL PROTECTED] 973-208-0942 fax
 http://www.netrom.com
 -

 On Thu, 18 Sep 2003, PJ Welsh wrote:

  I have to defend us newbies on this.

  This environment does not facilitate sequential knowledge building!
Based on my entry to Asterisk, I should have already known
T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you get
the idea (still trying to figure out skinny...cisco something, I know).
Heck, I'm struggling to get a grip on what and how to use/confiure SIP for
linux and keep my hair.

  You don't start off with a prerequisite of knowledge to join like a
class/school. You don't have the you-must-have-asterisk-101-before going to
asterisk-102 before you can join this list. You have a forum that is
GENERAL.

  I would like to a better effort to provide a more sensible way to start
helping us newbies. I have to say that the Digium handbook helped a little,
but not much. I have googled till I couldn't see straight. I just don't yet
have the big picture that most of you do. I couldn't even tell you if I
need a channel bank or a channel changer ;) at this point.

  A group of you seem to expect people to have a knowledge base that
allows for entering keywords to google. I don't know those keywords. You
know the context to search for when someone says I'm having a problem with
insert-thing-here.

  Instead of the usual, Search the archives. It would be more helpfull
to give a hint on what to search for. I could search for SIP and get back
several hundred answers. Then I have to figure out where that answer lies
in the series of possible answers. Then I have to somehow figure out if it
works.

  As most of you teachers (past and present) should know, not all of us
learn the same. Some people just get written material. Some NEED the
spoon to make it to the next level. Some need the hands-on experience and
other's just can't learn any more than they have already know(those people
are not likely on this list, however).

  You do realize that the http://www.asterisk.org/index.php?menu=support
lists the mailing list first for support, don't you. In fact, you have to go
to the second page before you even see the google reference. More a few
people tend to look for the FIRST way to get help not ALL ways to get
help...

  flame suit on

  On Thu, Sep 18, 2003 at 08:31:59PM +0200, Dave Cotton wrote:
  ...
   Absolutely agree with you Steve.  I left teachers training college in
   1970. I shock some teachers when I said that in all the years since I
   haven't taught anyone anything. I've just enabled them to learn.
   The problem is that in most national education systems the teacher is
   expected to provide the answers to pass some test at the end of the
   course. Thinking is not part of the curriculum.
   --
   Dave Cotton [EMAIL PROTECTED]
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RE: [Asterisk-Users] Grandstream Source?

I found that site very useful as well, but is very slow.
The webmaster of that site...!!!
I can provide FREE hosting for that site and it should be much faster. (
Another two cents from me)
Please do get in touch if interested.

(Web hosting is something I do not need spoon feeding for CERTAIN. (Not sure
about some other people)

Senad


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Re: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread Rmi Letot

Olle E. Johansson [EMAIL PROTECTED] writes:

couic

 I realized the same and started a process to collect a lot of that
 information and build a knowledge base on http://www.voip-forum.org/

The link doesn't work :-(

couic

 don't now and simply add What's a pyroflax? on it. Someone will
 notice and explain what a pyroflax is...

A what ? :-)

couic

-- 
Rémi 

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Re: [Asterisk-Users] Grandstream Source?

On Thu, 2003-09-18 at 15:19, John Vozza wrote:
 BS! :)
 
 Take the time to read and learn as much as you can from what's available
 and believe it or not you may just learn something. Even if that something
 is what to ask/search for.
 
 All those that get paid to answer questions on this list please raise your
 hand. I know my hand is still on the keyboard.
 
 I always amazes me how so many EXPECT so much for nothing...

We all expect something. The difference is whether we express that
expectation in our questions. Most people here do not express the
expectation and gladly accept the help they get. I'm including the
people who have been at the receiving end of flames sent by myself there
too. Only a few times have we had to deal with people coming in
demanding support. We have dealt with it when it comes up. 

Maybe I was a bit lucky when I came into the asterisk fold that I was
not under any time constraints to get a system up and working. This gave
me the luxury of lurking a bit more to understand the terms before
jumping too deep into it. Learning is a long term project. Learning
telephony is a really long term project. If you don't have time to learn
it yourself, you should seek a consultant or a commercial product.
Remember time is money, and you will either spend time or money on a
project. This is true no matter what the project is. 

-- 
Steven Critchfield  [EMAIL PROTECTED]

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Newbie delimas was Re: [Asterisk-Users] Grandstream Source?

I expect a user list to be for users' questions. I expect a user list to support that 
what it's a list for. In return *I* should help someone when/if I can! There is no 
for Nothing. You help me, then I help some newbie 10 years from now when I 
understand this stuff. So, in the meantime, my only contribution is the list of sites 
I have found to be usefull. I forgot to change the subject line, however.

I am finding that it's hard to find out what's available when I don't know what's 
available...

Don't get me wrong, I would like for this to be a *constructive* thread! I don't not 
want anyone to get offendend. I would just like a general realization of the newbie 
situation. I still think this list is good! I still think * is great. I still have 
faith that I can figure all of this out. I know it will take the help of many good 
people with ALOT of patience and understanding and experience to help me. 

I am very greatful for all of the information that you list goers have provided! So 
many of would do anything to help and do. The more I search through the archives, the 
more I know that I'm still in the right place to help me.

Again Thank you for your understanding, help, time and effort!

On Thu, Sep 18, 2003 at 04:19:04PM -0400, John Vozza wrote:
 BS! :)
 
 Take the time to read and learn as much as you can from what's available
 and believe it or not you may just learn something. Even if that something
 is what to ask/search for.
 
 All those that get paid to answer questions on this list please raise your
 hand. I know my hand is still on the keyboard.
 
 I always amazes me how so many EXPECT so much for nothing...
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[Asterisk-Users] Adpcm quality

2003-09-18 Thread Alex Zarubin

Title: Adpcm quality 





Please, try


exten = 99,1,Wait,1
exten = 99,2,Record,/tmp/pcmfile:pcm
exten = 99,3,Wait,1
exten = 99,4,Playback,/tmp/pcmfile
exten = 99,5,Wait,1
exten = 99,6,Record,/tmp/voxfile:vox
exten = 99,7,Wait,1
exten = 99,8,Playback,/tmp/voxfile


(put your own extension).


Pcm recording is OK, playback is OK.
Adpcm recording is noticeably worse. Adpcm playback is very bad/unusable.


Is it just us (all our servers with T400P and TE410P), or it's a common adpcm codec problem?


Thank you.


Alex Zarubin
Webley Systems, Inc.

Re: [Asterisk-Users] New message 0 in mailbox 7606

2003-09-18 Thread noc

-- Original Message --
From: Steven Critchfield [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date:  Thu, 18 Sep 2003 14:55:04 -0500

On Thu, 2003-09-18 at 14:37, noc wrote:

I don't use VM2 yet, but lets see if I can answer a couple of questions.

 I recently started playing with voicemail2.  I'm having two minor
 problems that I can't seem to find discussed in the archives.

 1) New message 0 in mailbox 7606.  New voice mail message count seems
 to start with 0 for the first new message instead of 1.  Any tricks to
 fix this?

Is this what is read to you, or is this a storage question? C
programmers usually start counting from 0. Maybe a slip up to not add 1
before presentation. 

This is a problem with both the text of the email notification (New message 0) and the 
message storage (msg.WAV).  If I call VoicemailMain2, it is announced as You have 
1 new message, so that works.  It's just the email notifications and the message 
storage that starts with 0.

 2) When listening to messages with VoicemailMain2, the time stamp is
 in GMT and not corrected for the local time zone offset.  But the
 email that voicemail2 sends has the correct time.  I added
 |tz=eastern to the end of the mailbox definitions in voicemail.conf,
 but that did not seem to fix the problem.

I think Mark added a patch this morning at 8:30am -5 to default to
grabbing system timezone.

I just updated to the latest CVS.  This is still a problem.  The email notification 
has the correct time, but the VoicemailMain2 announcement of the time stamp is wrong.

 Any thoughts on these two problems?  I'm running a recent CVS from
 9/14/03.

Gasp, you are running four day old software, what will the neighbors
think ;)

As usual, if you seem to have a problem, please upgrade to current and
see if the problem persists. If it does persist, then ask for help or
confirmation of the problem. 
-- 
Steven Critchfield  [EMAIL PROTECTED]

Thanks for all the help,

-Jon

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Re: [Asterisk-Users] New message 0 in mailbox 7606

2003-09-18 Thread noc

-- Original Message --
From: Tilghman Lesher [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date:  Thu, 18 Sep 2003 15:20:54 -0500

On Thursday 18 September 2003 14:37, noc wrote:
 2) When listening to messages with VoicemailMain2, the time stamp is
 in GMT and not corrected for the local time zone offset.  But the
 email that voicemail2 sends has the correct time.  I added
 |tz=eastern to the end of the mailbox definitions in
 voicemail.conf, but that did not seem to fix the problem.

Please send more information about your configuration.  Include details
like distribution version and the contents of voicemail.conf.

This isn't a problem on either Mandrake or Slackware, and since Critch
isn't complaining, probably not on Debian, either.  However, I'm working
with somebody who has this problem on RedHat 8.

-Tilghman

I'm running the latest asterisk from CVS on Redhat 7.1.  My voicemail.conf is fairly 
vanilla.  Here's what the last few lines look like:

[zonemessages] 
eastern=America/NewYork|'vm-received' Q 'digits/at' IMp 
central=America/Chicago|'vm-received' Q 'digits/at' IMp 
central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' M 
'digits/hours'

[default]

7606 = 7606,My Name,[EMAIL PROTECTED]

I also tried this:

7606 = 7606,My Name,[EMAIL PROTECTED]|tz=eastern

Adding |tz=eastern to the end did not help.

Thanks,

-Jon

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[Asterisk-Users] loading dialogic drivers

2003-09-18 Thread pedro bulach gapski

I am one of those trying to use old dialogic hardware with *. I have the 
following error when loading the driver:
[chan_dialogic.so] = (Dialogic Global Call API Support)
dlopen of libicapi.so failed: dlerror=/usr/dialogic/lib/libicapi.so: 
undefined symbol: gcdb_InsertLinedev
WARNING[1024]: File chan_dialogic.c, Line 832 (load_module): Failed to 
start Global Call (GC)
WARNING[1024]: File loader.c, Line 299 (ast_load_resource): 
chan_dialogic.so: load_module failed, returning -1
WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module 
chan_dialogic.so failed!

Indeed, gcdb_InsertLinedev is not defined in libicapi. It is defined in 
libgc, which is linked to chan_dialogic.

Anyone has seen this before?

[],

pedro

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Re: [Asterisk-Users] I need your help

2003-09-18 Thread Armand A. Verstappen

Hi,

On Thu, 2003-09-11 at 09:49, Steve Meyers wrote:
 P.S. Anyone want to take bets on how long it will take for Steven
 Critchfield to berate this guy for improper email usage? :)

Please don't make it look as if Steven is being foolish. I fully agree
with him on the improper mail usage. I just costs me less time to break
something expensive than to reply and try to educate the culprit.

I think top-posting and html mail are a clear sign that the sender
thinks his own time and comfort more important than those of the people
they're soliciting help from.

wkr,

-- 
Envida http://www.envida.net/
Armand A. Verstappen   Graadt van Roggenweg 328
[EMAIL PROTECTED]   3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht


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Description: This is a digitally signed message part

Re: [Asterisk-Users] CDR of calls transferred via IAX[2]

2003-09-18 Thread Brancaleoni Matteo

Ciao lele.

that's a good question. Of course I don't know an answer (yet).
anyone does?
anyway, here's some comments:

 * one could pass via IAX a uniqueid when i transfer the call, and have this 
 unique id logged in the CDR records. This way any call segment pertaining to 
 the same phone call can be correlated for cdr purpose.
nice, but that way we have to join 2 records for a real name... what
happens when the switch  gateway are 'very remote'

 * one could have the gateway allow trusted sources (the switch) to set via 
 IAX the accountcode when transferring the call, and log it as an 
 originalaccountcode or even the accountcode itself in the cdr.
 This way every cdr record in the network will have a reference to the actual 
 customer that made that event happen.
I would stay on that, but without adding another cdr field
(originalaccountcode), but simply, since we don't mind 'bout the
user as we authenticate from the switch statement, just get the
accountcode from the originating switch,so I agree with you.
Seems also the simplest way to do that.
I would add a sort of @switch in the accountcode, so my user 'caller'
will be added to the cdr (of the gateway) as 'caller@switch_name'
in order to be able to know that the user originated from a remote
machine. if the accountcode hasn't the @swicth_name part, means
that the user is local.

 * one could devise some way to give back from the gateway to the transferrer 
 (the switch) an indication that the call has ended, with that many 
 billable_seconds. (can this be done? i do not see it that simple...)
 This way the switch would have all the cdr info in one cdr row.
that means a connection back. I would discard that

 thanks
 lele

of course, only my 2 cents ;)

-- 
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl

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Re: [Asterisk-Users] Distinctive ringing

2003-09-18 Thread Robert Boardman

Does  asterisk know when each ring comes in or just the first ring, ie 
so the cadence can be worked out? say over two rings?

Robb
Martin Pycko wrote:
The X100P together with asterisk does not support the distinctive ringing
detection on the line. Asterisk however can generate the distinctive ring
over FXS ports.
regards
Martin
 

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Re: [Asterisk-Users] Grandstream Source?

2003-09-18 Thread Steve Creel

I am NOT a VoIP guru.  I am NOT an Asterisk guru.  I am NOT a telephony
guru.  Take that as a disclaimer for the information below, as well as to
say that the best learning comes from reading anything you can get your
hands on.  The idea of post any question to the mailing list works well
with 10 people.  It scales horribly.  Reading through the archives, you
will see the same questions asked (and answered) over and over.  At _some_
point, it's okay to say I've answered it 15 times, YOU can go look it
up on YOUR time.  Besides, I'd rather spend 3 hours looking for the
answer than just ask my question, because I hate looking like an idiot.

This isn't a flame, nor a sarcastic, snide response.  I don't want to
complain about people asking what is a  if I've never made an
attempt to answer that question for someone.

On Thu, 18 Sep 2003, PJ Welsh wrote:

I have to defend us newbies on this.

This environment does not facilitate sequential knowledge building! Based
on my entry to Asterisk, I should have already known
T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you get
the idea (still trying to figure out skinny...cisco something, I know).
Heck, I'm struggling to get a grip on what and how to use/confiure SIP
for linux and keep my hair.

A T1 is technology used to deliver digital data from one device to
another.  Most of us are familiar with data T1s - 1.544mbps.  When used
for voice, they can be PRI (primary rate interface) or Channelized T1.  A
PRI has 23 voice channels and a bearer channel.  The Channelized T1 has 24
voice channels.  Depending on the specific application, one may be better
suited than another (or depending on the price).  There are many other
technical characteristics about a T1, but know we've established what it
is.

An E1 is used for the same purposes as a T1.  Which one is it depends on
your geographic location - T1 in US, Canada, and Japan (according to a
telecom dictionary on the shelf here, sorry if misinformed).  Other parts
of the world use E1.

VoIP refers to the high-level use of an IP network (or IP equipment) to
deliver telephone service.  Sometimes this means telephone calls from a
software app on one machine to another software app.  It could mean a call
from one physical analog phone to another that was connected by way of an
IP network.  It could refer to an off-premise extension of your desk phone
to home.

SIP is session initiated protocol.  There are two parts to VoIP
protocols - the call setup and the audio stream.  All of the audio is
handled similarly with most protocols.  The difference is usually in call
setup.  You can use SIP to call from one phone to another directly,
without a callmanager, gatekeeper, or any other VoIP equipment.  SIP
allows IP addresses to be entered and called directly.  SIP seems to be
best for single-line extensions, I want to call my brother in _ ,
and for most consumer-grade VoIP for home use.  The biggest user
experience thing I can think to mention about SIP is that dialing
_usually_ (excluding early dial) works like a cellphone - dial number 
press send.

Skinny (or SCCP used interchangably) is Cisco's Skinny Client Control
Protocol.  It is a proprietary protocol that Cisco uses in their Call
Manager system.  The Cisco phones use SCCP to talk to the server (yes,
like how a SIP phone would use SIP to talk to another phone, or to a SIP
server).  Because Cisco is Cisco, there is a certain demand to use their
devices.  To accomodate this, they have offered SIP firmware to load on
some of their phones.  However, the SIP firmware does not offer all of the
features of the firmware for SCCP.  Some of this is protocol limitations,
some is because they didn't include it.  Asterisk's support for SCCP is
beginning to be functional (no disrespect to those who have put tons of
time in on it already - beginning in that it's beginning to be offered,
not beginning to be worked on).

FreeWorld is Free World Dialup, or FWD.  Their website,
www.freeworlddialup.com, says the following:
Free World Dialup (FWD)  allows you to make free phone calls over
the Internet using a 'regular' telephone or a computer program.

Free World Dialup does not directly provide access to the
traditional telephone networks or cellular networks. FWD members
can only call other FWD members and customers of IP-based service
providers who have a business relationship with FWD. If you are
interested in learning about VoIP and would like to setup your own
personal PBX, give Asterisk a try.

H.232 is a typo, the protocol is H.323.  My understanding is that it is
essentially the first-generation VoIP protocol.  Generally this is
associated with older equipment, or a last-resort for interfacing
otherwise incompatible equipment.  Netmeeting used to use it, and still
may.  That's all I know about H.323, and I may be wrong about all of it.


An X100P is a Digium FXO card (see FXO quickly explained

Re: [Asterisk-Users] Grandstream Source?