[Asterisk-Users] Cisco 7960 Skinny Firware

2005-10-30 Thread Bobby Lacey 








Hello,



I have just acquired my first 7960 from a business sale. It
has already been preloaded with the 7.3 SIP image which works flawlessly with
my Asterisk box. I want to experiment with chan_sccp and therefore I would need
the skinny firmware = 7, I
guess. Could someone tell me an outlet where I could purchase a Smartnet contract to download this firmware? I have been
unable to find a retailer online that can help. Thanks in advance for any help.



BL






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[Asterisk-Users] Intercom with Aastra 480e?

2005-04-11 Thread Bobby Lacey 








Hello list,



I have been successful in setting up my first * box with a
pair of x100ps, Cisco 7960, and a Digium iAXy.



I would like to incorporate an Aastra
480e using my iAXy and ADSI. I want to be able to
answer phone calls with my 7960 in the back of the house and park the call, then in turn call the intercom on the 480e in the front
(using two way audio) to announce that there is a call that needs to be picked
up on 701.



Also, by using the Aastra 480e,
can I see my Zap line status to see what lines are available and also if
extensions are in use?



Thanks in advance.



B. Lacey






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[Asterisk-Users] 2 x100p + Static + echo

2005-02-04 Thread Bobby Lacey 
Title: Message



Hello

I recently have got 
my setup up and running. I am using 2 x100p cards in a Dell 
400SC.

I get static 
(inbound and outbound) within the first 20 seconds on one card, but on the other 
card it is crystal clear all the time. They both have their own irq and I have 
tried to remedy the problem by doing a hdparm found in the wiki, but still 
nothing. I've tried rxgain and txgain, but that just seems to increase and 
descreas the static volume.

The other problem I 
have is that I get a bad echo on both cards when using them. I can hear myself 
constantly. Here is my zapata.conf:

signalling=fxs_ksechocancel=yesechocancelwhenbridged=yesechotraining=400callerid=asreceivedgroup=1context=pots1-inboundchannel 
= 1

signalling=fxs_ksechocancel=yesechocancelwhenbridged=yesechotraining=400callerid=asreceivedgroup=1context=pots2-inboundchannel 
= 2

Any help is greatly 
appreciated! TIA.

-Bobby
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[Asterisk-Users] Slackware + Asterisk + asterisk-addons

2005-01-30 Thread Bobby Lacey 
Title: Message



Hello

I am trying to get 
asterisk-addons installed so that I can use the mysql cdr feature. OK, I have 
the MySQL server (mysqld) installed, but I noticed that mysql-devel is also 
required. I tried to compile asterisk-addons and got a:

--CUT---
res_config_mysql.c:422: error: unknown field `realtime_multi_func' 
specified in initializerres_config_mysql.c:422: warning: excess elements in 
struct initializerres_config_mysql.c:422: warning: (near initialization for 
`mysql_engine')res_config_mysql.c:423: error: unknown field `update_func' 
specified in initializerres_config_mysql.c:424: warning: excess elements in 
struct initializerres_config_mysql.c:424: warning: (near initialization for 
`mysql_engine')res_config_mysql.c: In function 
`parse_config':res_config_mysql.c:491: warning: assignment makes pointer 
from integer without a cast/usr/include/asterisk/utils.h: At top 
level:res_config_mysql.c:418: error: storage size of `mysql_engine' isn't 
knownmake: *** [res_config_mysql.o] Error 1[EMAIL PROTECTED]:/usr/src/asterisk-addons#

So I'm assuming that 
I need mysql-devel? I went to download the package from mysql.com, but the only 
format I saw was .rpm. I am using Slackware 10.0 so rpm doesnt want to work 
well. I get the following:

rpm -Uvh 
MySQL-devel-4.1.9-0.i386.rpmwarning: MySQL-devel-4.1.9-0.i386.rpm: V3 DSA 
signature: NOKEY, key ID 5072e1f5error: Failed 
dependencies: /bin/sh is needed by 
MySQL-devel-4.1.9-0


Does anyone 
have any suggestions or help they could offer?

Thanks in 
advance..
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RE: [Asterisk-Users] IP Phone for IP PBX

2005-01-28 Thread Bobby Lacey 
Title: Message



I just 
purchased a Cisco 7960 IP phone and I couldn't be more happier. It supports 6 
lines. You can usually find a good one off eBay.

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Bilal 
  GhayadSent: Friday, January 28, 2005 11:02 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] IP Phone 
  for IP PBX
  Hi 
  All;Asterisk is IP PBX, so there should be special IP Phones to be 
  used withthis IP PBX and this IP Phone should has special buttons working 
  withAsterisk IP PBX, like: Line1, Line2, Line3, ... Line8 (these to be 
  able tocapture direct PSTN lines), also a button for Hold and transferring 
  foranother extension ... So can some one advise me which is the 
  recommended IPPhone to be used with Asterisk and able to do these 
  functionalities?RegardsBilal
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[Asterisk-Users] Cisco Maintence Contract for my 7960

2005-01-24 Thread Bobby Lacey 
Title: Message



Hello 
Everyone,

I am new to Asterisk 
and this list. So far the archives have been very useful.

I just purchased a 
new Cisco 7960and I am having trouble getting a service contract out of 
CDW and Insight. Both of them have giving me dead end. I just need a contract so 
that I can get the firmware upgraded to make the phone compatible with SIP. Does 
anyone else know of a outlet that I can take in order to get the 
contract/firmware?

Thanks in 
advance,

B. 
Lacey
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[Asterisk-Users] Nufone and Dialing Out

2005-01-24 Thread Bobby Lacey 
Title: Message



Good 
evening,

I just signed up 
with Nufone and I am able to receive calls with no problem via my 800 number. 
Outgoing callsare not going through 
though. 


My extensions.conf 
is as follows:

[nufone-out]
exten = 
_91NXXNXX,1,SetCallerID(mynumber)exten = 
_91NXXNXX,2,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:1})exten 
= _91NXXNXX,3,Congestion

Whenever I try to 
dial out through X-Lite, I get a:

Called Failed: 503 
Service Unavailable


Asterisk shows 
the following:

 
-- Executing SetCallerID("SIP/1234-7c6e", "xx") in new 
stack -- Executing Dial("SIP/1234-7c6e", 
"IAX2/user:[EMAIL PROTECTED]/1xx") in new 
stack -- Called 
user:[EMAIL PROTECTED]/1xx
 Jan 24 22:05:23 WARNING[9154]: 
chan_iax2.c:5996 socket_read: Call rejected by 198.22.67.70: No authority 
found -- Hungup 'IAX2/198.22.67.70:4569/1' == No 
one is available to answer at this time (1:0/0/0) -- 
Executing Congestion("SIP/1234-7c6e", "") in new stack == Spawn 
extension (default, 91xx, 3) exited non-zero on 
'SIP/1234-7c6e'

Note: 
xx is the given phone nubmer I am dialing. Also user:pass is my defined 
username and password.

TIA!

B. 
Lacey

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