Try this configuration file...
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configu
ration+files+for+SIP#Cisco7961with833SR2ConfigurationExamples
Chad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Wong
Sent: Friday,
I've always found it best to run with no CTLSEPMAC.tlv file in the
tftp server directory; it will ignore that and move on.
With the 7961's you'll be best in the 8-3-3SR2 leading edge, the DND
button is where it should be, and transfers work the way they should
again.
The XML configuration file
I contacted one of the list users and they sent me their configuration
files.
I used it as a template and it worked with my phone, so I'll be sure to
put it back up on the Wiki.
Chad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hello,
I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have
recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front
of the phone and also to hopefully resolve some issues with the phones
not registering after a long period.
Once we upgraded the phones now
The A101 does not require a PCI-X.
I have 2 A102's running in standard ports here.
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Duracom
Lists
Sent: November 27, 2006 2:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
T1 PRI's are (almost?) always copper.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: June 16, 2006 5:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] T1 Copper or T1 Fiber Line
I am
Try "rpm -qa kernel-devel" or "rpm -qa kernel-smp-devel" to verify
they're installed.
If not do a "yum install kernel-devel or kernel-smp-devel" depending on
which you have.
Chad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J.J.
FeminellaSent: June 13, 2006 9:51 AMTo:
Prepend outgoing calls by using Dial(9{exten}) instead of dialing 9 to
get out on your system...
Or, add a 9 to caller id.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
GibbsSent: May 18, 2006 11:37 AMTo: Asterisk Users Mailing
List - Non-Commercial
You can use my script, based on Chris Mason's script, to do most of what
you want, you can feed it your MAC's and Extensions and it will create the
phones.
Be warned, it's not pretty, my perl book was in storage so I did a lot of
kludging. Feel fee to update.
Are you passing the Offset through the DHCP server as well? On a linux
DHCP server this would be:
option
time-offset
-18000; # Eastern Standard Time
option
ntp-servers
192.168.x.x
The fact that the date is wrong, but the time is correct, seems a little
strange to me. Are you sure your
If you use a Sangoma T1 card, (A10x) card you can send both voice and
data down the same T1 and have the Sangoma card split it for you.
If you are talking about non-hobby usage, stay away from FXO adapters
and go with a T1.. You'll be much happier in the long run. For a
fractional T1, don't worry
SMC 6824MPE.. Does 24 ports POE, with 2x 1GB uplinks, (RJ45 or GBIC)
L3 Managed switch.
We've got four of them here, and I think they're great, the cost was
really reasonable.
Ingram no longer lists the MPE model, but it should be available still.
Chad
-Original Message-
Hi listers,
Yes, It can send Caller ID down the T1 line..
Some T1's only accept Caller ID's that match the set of DID's associated
with the T1.
Others, like mine, will take anything you send it...
Chad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jordan
NovakSent: March 30, 2006
I was just thinking, about this..
Move your Polycom Power Injecting Patch cable (Black Cable with AC
Adapter Input) into the cabling closet. You could then infuse the power
at the cabling closet and then just use a standard patch cable to patch
the phone in.
You would be looking at a line loss
Telecom Ottawa?
Large, Ultra fast pipe with direct connections to TDM providers (Which
may be at 151 Front St. in Toronto) but they should work for what you
want.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
OSSSent: February 19, 2006 12:04 AMTo:
Title: Asterisk vs. Traditional PBX
You may be using less then ideal phones. With a Polycom 501, I can't see
you having voice quality issues, With a Sangoma or Digium card and a PRI the
quality and functions of a Asterisk system are on par with most PBX's (I'd say
they're above).
It is a
Can anyone shed some light on what happened?
Asterisk 1.2.1 with Zaptel 1.2.1
Here is what I know happened:
A call came into our main number and was answered
Asterisk set the monitor CALLFILENAME and then started monitor.
The call was directed to a context called open where all calls go
during
Realtime.. As in pulling configs from a realtime database..
Or he's trying to link Asterisk to www.bestpracticals.com version of
Request Tracker (also known as RT)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Charles
Wang
Sent: February 3, 2006 8:13
BootBlock 2.5.0
Bootrom 2.6.2.0032
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron
Senykoff
Sent: January 27, 2006 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom 501 horrible echo
I've
I've been running 1.6.4.0064 for the last few weeks..
I've had no problems with it, I haven't done a whole lot of speaker
phone with it yet though.. Once my IP4000 reboots It'll be running it as
well so that will be a good test.
Chad
-Original Message-
From: [EMAIL PROTECTED]
I have had no problems running the Sip.cfg from 1.5.2 with 1.6.4 so far,
but I am looking to update in the next while.
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Miller
Sent: January 27, 2006 1:07 PM
To: Asterisk Users Mailing List -
Supporting authentication directly against voicemail.conf or using
an LDAP directory,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: January 23, 2006 3:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
RJ11Plantronics
13
24
31
42
RJ11 Pin 1 is on the left when looking at the contact points.
Plantronics Pin 1 is on the left when looking at the contacts (through
the plastic sheild)
My multimeter battery is low, so YMMV, but:
Pin's 1,4 are connected with ~160
Have you considered the Sangoma cards? I have an a102 running in 2x
X306's and they're running fantastic.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Harry
McGregor
Sent: December 19, 2005 4:20 PM
To: asterisk-users@lists.digium.com
Subject:
I
have had no issues where asterisk is affected by a Sangoma card being
down.
I
ran my test server like that for a few weeks doing lots of testing before I
brought it up with a dummy card. Even now, if it's up or down it doesn't matter
to asterisk.
Chad
From: [EMAIL PROTECTED]
The value of 14800 is correct.
I had issues with my TDM400p with 2x FXO's installed and using the Xlite
client. I could not get the echo stable at the initial call.
Changing to a hard phone made everything work correctly. I still had problems
with the off location I called, but mostly worked
From what I understand (From Sangoma's tech support) and having a IBM
x306 SCSI system with an A102u I believe that the system will scale up to 4xT1's
easily.
With a full T1 of traffic coming in and playing music on hold,
theCPU was at 7% with no transcoding.
Sangomacards are supposed to
To use the old phones and existing wiring you'll need some E1/T1 FXS
Channel banks and a T1/E1 Card. Each bank will handle 30/24 phones and
pipe them into a single E1/T1 connection.
You can connect up to 4 (or 8 soon from Sangoma) T1's per card. I really
like the Sangoma cards, there are also
Title: Message
Has anyone else seen
this problem? MWI works, but when you press the messages button the display
shows -1 urgent, 1 new, and 0 old.
Anyone know how to
fix this?
Chad
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I do not know about Thailand DID's, but I would rather not see you post
six times about this.
There should be some information in the Wiki about providers all across
the world and Google may have some additional information.
Try the -biz list for biz' related questions.
Or, the Wiki has a lot
Remove Callerid and set immediate=yes
Callerid is sent between the first and second rings, so asterisk has to
wait for it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh
Divecha
Sent: Monday, July 18, 2005 5:05 PM
To: Asterisk Users Mailing List
or the older
phones that run 1.5.2.
On 7/14/05, Chad Osmond [EMAIL PROTECTED] wrote:
CVS Head from 07/07/2005
I'm trying to make an IP-501 auto answer a call.
exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans)
exten = 301,2,SetVar(ALERT_INFO=Ring_Ans) # Tried both
combinations
exten = 301,3,Dial(SIP
But once you have one setup you can just buy a barcode scanner, scan the
MAC from the label, print your secret and other data entries as barcodes
and use the script to set them all up.
I'm loving the polycom setup at this point, the central configuration
setup is fantastic.
Chad
-Original
CVS Head from 07/07/2005
I'm trying to make an IP-501 auto answer a call.
exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans)
exten = 301,2,SetVar(ALERT_INFO=Ring_Ans) # Tried both combinations
exten = 301,3,Dial(SIP/5001,15)
exten = 301,4,Hangup
Sip.cfg for Polycom phone
alertInfo
Can you be a bit more specific as to what the problems is?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mohammad
Sent: July 6, 2005 2:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] converting windows .wav to .gsm
HI ALL;
Did you try tailing the /var/log/dmesg to see what happened when you
loaded zaptel and wctdm with modprobe?
Check that /etc/modprobe.conf still contains the correct module entries.
Does /lib/modules/2.6.11-1.35_FC3smp/misc still contain and correct
wctdm.ko files?
-Original Message-
For someone that places outbound calls only, in a fairly low volume, is
there a recommendation for which one would be best for me?
I have had continual audio trouble with LiveVOIP, though other services
(FWD) work fine, so I'd want something that has good audio quality.
I will toss in my $0.02
Yes,
I'm running it right now, CVS as of a few days ago, and * 1.0.7 on 2.6.x
kernel and FC2.
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: June 20, 2005 2:44 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
The Dual 2.8GHz will be much faster for running everything. If it is the
same price it should be a no brainier, take the two CPU system.
Depending on the manufacture of the system it may even take a failure of
one CPU.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Welcome to DSL, the telco didn't do any more tests then required to get
sync for 30 seconds.
Cancel the DSL and get another line. That's about the extent of it, or
at least in Ontario it is, I've had this problem with 5 or 6
connections.
Chad
-Original Message-
From: [EMAIL PROTECTED]
-Original Message-
snipped
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Sampson
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
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