[asterisk-users] Got 200 OK on REGISTER that isn't a register

2006-11-15 Thread Stas Khromoy
chan_sip.c: Got 200 OK on REGISTER that isn't a register.

i'm getting the above warning
while trying to register a phone from outside of asterisk network.
( so no registration what so ever, no dial tone and what not)


it registered once for about 20 minutes
exepted calls and i could call out
but with no audio on either end.

any ideas ?

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Re: [asterisk-users] Got 200 OK on REGISTER that isn't a register

2006-11-15 Thread Stas Khromoy


as far as i know sip-based

i came across something that said
this could be due to too much traffic
but the mesage was not clear on what side


 Original Message  
Subject: Re:[asterisk-users] Got 200 OK on REGISTER that isn't a register
From: Ron McLeod [EMAIL PROTECTED]
To: [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial 
Discussion' asterisk-users@lists.digium.com

Date: 11/15/2006 2:03 PM

I think this message is saying that it received a 200 OK for a REGISTER
message that Asterisk does not know about (anymore).

Is you system trying to register with an ITSP or other SIP-based system?

  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stas Khromoy
Sent: Wednesday, November 15, 2006 10:38 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Got 200 OK on REGISTER that isn't a register

chan_sip.c: Got 200 OK on REGISTER that isn't a register.

i'm getting the above warning
while trying to register a phone from outside of asterisk network.
( so no registration what so ever, no dial tone and what not)


it registered once for about 20 minutes
exepted calls and i could call out
but with no audio on either end.

any ideas ?

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Re: [Asterisk-Users] Got 200 OK on REGISTER that isn't a register

2006-11-15 Thread Stas Khromoy


i figured that out
what i can't find is a solution to the problem

 Original Message  
Subject: Re:[Asterisk-Users] Got 200 OK on REGISTER that isn't a register
From: BJ Weschke [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Date: 1/1/2006 12:26 PM

On 1/1/06, Robert La Ferla [EMAIL PROTECTED] wrote:
  

What does this warning mean?

WARNING[11065]: chan_sip.c:9596 handle_response_register: Got 200 OK on
REGISTER that isn't a register




 Your SIP device is returning a 200 OK message about a registration
attempt, but Asterisk doesn't believe there is a registration attempt
in progress with this phone. This is what's generating the message.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[asterisk-users] (no subject)

2006-11-10 Thread Stas Khromoy

i am sure this came up before
but all my searches are not resulting in anything usefull

trying to setup a grandstream phone
to connect to an asterisk server

now i am outside the network (home)
on my side
settings on the phone seem to be correct
id and password, astersik server ip, port

in pf.conf
# SIP (TCP)
voip_tcp = 5060
# SIP, IAX2, IAX, RTP, MGCP (UDP)
voip_udp = {5060, 4569, 5036,   20001, 2727}

---
on the server side

same thing
plus
voip_users =  ip from where i am connecting 

--
can't seem to find anything else that should be opened on either side
to allow connection


--
i guess, help ?
--


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[asterisk-users] asterisk and PowerEdge 1950

2006-09-21 Thread Stas Khromoy

hey folks

we're planing to install asterisk for a client of ours
was just wondering if the Dell's PowerEdge 1950
will take 2 - T1 cards.

or if there any recommendations as to which 
server would be good for our project.



thanks

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[asterisk-users] polycom soundstation 501 crash

2006-08-02 Thread Stas Khromoy

hey folks

hope some one came across this problem

one of our polycom's just crashed
after reboot it comes up with this error

error loading 0004f204fcc.cfg


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Re: [asterisk-users] polycom soundstation 501 crash

2006-08-02 Thread Stas Khromoy


i am afraid the second could be the case
since the whole block where the office is lost power yesterday

thanks

PS : FPT = FTP ? :)


 Original Message  
Subject: Re:[asterisk-users] polycom soundstation 501 crash
From: Alexander Lopez [EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com

Date: 8/2/2006 10:13 AM

Make sure you can access the file on your FPT server. Also make sure
that you did not fry the Ethernet port(s) on the phone.


  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stas Khromoy
Sent: Wednesday, August 02, 2006 9:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] polycom soundstation 501 crash

hey folks

hope some one came across this problem

one of our polycom's just crashed
after reboot it comes up with this error

error loading 0004f204fcc.cfg


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Re: [asterisk-users] polycom soundstation 501 crash

2006-08-02 Thread Stas Khromoy

not to sound like an idiot
but where do i get the files ?


these guys ?

http://www.polycom.com/resource_center/0,1454,pw-6812-12612,FF.html
SoundPoint IP/SoundStation IP SIP Software 1.6.7
SoundPoint IP/SoundStation IP BootROM 3.2.1


 Original Message  
Subject: Re:[asterisk-users] polycom soundstation 501 crash
From: Jessee J Holmes [EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com

Date: 8/2/2006 10:20 AM
This means the phone is attempting to load this configuration file and 
cannot find it from your boot server. The phone at this point must 
have a boot server with these files. Put this file on a FTP server and 
point the phone to that server to pickup and download this file.



Jessee Holmes

Atacomm / Ataractic Corporation

www.atacomm.com

V: 1-877-700-VOIP

[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]


Looking for voice over IP products?  Visit our VoIP store at 
http://voipstore.atacomm.com/




On Aug 2, 2006, at 8:50 AM, Stas Khromoy wrote:


hey folks

hope some one came across this problem

one of our polycom's just crashed
after reboot it comes up with this error

error loading 0004f204fcc.cfg


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[Asterisk-Users] budget tone 100

2006-05-11 Thread Stas Khromoy



does any one know
if Grandstream BT 100 has a 'do not disturb' function ?

one of the BT's is not picking up calls all of a sudden
out going is fine
but when you dial in
you get forwarded directly to voice mail.

thanks
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Re: [Asterisk-Users] budget tone 100

2006-05-11 Thread Stas Khromoy

i've checked for that
it is 'subscribe to MWI'
but it is set to 'no' already

i looked in the full log
for error and disabled came up with these

May 11 16:17:39 VERBOSE[26307] logger.c:
--  dialparties.agi: Extension 107 cf is disabled
May 11 16:17:39 VERBOSE[26307] logger.c:
--  dialparties.agi: Extension 107 do not disturb is disabled
May 11 16:17:39 VERBOSE[26307]
logger.c:   dialparties.agi: Extension 107 has call waiting disabled




May 11 16:15:31 WARNING[26274] ast_expr2.fl: ast_yyerror(): syntax 
error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL 
or TOK_LP or TOKEN; Input:
May 11 16:15:38 WARNING[26274] ast_expr2.fl: ast_yyerror(): syntax 
error: syntax error, unexpected TOK_NE, expecting TOK_MINUS or TOK_COMPL 
or TOK_LP or TOKEN; Input:



thanks for help

 Original Message  
Subject: Re:[Asterisk-Users] budget tone 100
From: Steve Jones [EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com

Date: 5/11/2006 3:52 PM


On mine, I had that happen, until I turned off subscribe to MWI or
something like that in the config (sorry - I can't remember the exact
verbiage.) I also upgraded the firmware around the same time, but I
think from advice I got on this list, the MWI setting was the reason...

-Steve

-Original Message-
From: Stas Khromoy [mailto:[EMAIL PROTECTED] 
Sent: Thursday, May 11, 2006 11:44 AM

To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] budget tone 100



does any one know
if Grandstream BT 100 has a 'do not disturb' function ?

one of the BT's is not picking up calls all of a sudden
out going is fine
but when you dial in
you get forwarded directly to voice mail.

thanks



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[Asterisk-Users] a few questions

2005-12-09 Thread Stas Khromoy

we are beginning to test asterisk for our office
one of the features of the current phone system that is very heavily 
used is overhead paging


now i came accross this post
http://forums.digium.com/viewtopic.php?t=2844highlight=features

that basically says it is not possible with asterisk.

let's hope that i am not understanding it right, since i am new to the 
telephony.


can any one help me out and explain it to the unfortunate ? :))

second question is as follows:

when you access voice mail
the default msg is 'welcome to comedian mail'
is there any way to get rid of this par of the greeting ?


thanks


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Re: [Asterisk-Users] a few questions

2005-12-09 Thread Stas Khromoy

my apologies if anything
but as i said i am not that knowledgeable
and most probably misunderstood the post.

as it looks from your reply i have

if you don't mind letting me know what i got wrong
i would greatly appreciate it.




On 12/9/05, Stas Khromoy [EMAIL PROTECTED] wrote:

we are beginning to test asterisk for our office
one of the features of the current phone system that is very heavily
used is overhead paging


Overhead paging can be done with asteirsk in anyway you want, you can
even do mutilple zones, all zones, or whatever you want.


now i came accross this post
http://forums.digium.com/viewtopic.php?t=2844highlight=features



I cound't find *anything* on that page that has to do with paging.


that basically says it is not possible with asterisk.



Exactly where on that page???


let's hope that i am not understanding it right, since i am new to the
telephony.

can any one help me out and explain it to the unfortunate ? :))

second question is as follows:

when you access voice mail
the default msg is 'welcome to comedian mail'
is there any way to get rid of this par of the greeting ?



Yeah, just rerecord that massage.


thanks


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[Asterisk-Users] phone intergration

2005-11-18 Thread Stas Khromoy

hey folks

is there a way to integrate toshiba  dkt2010-sd into asterik network ?
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