Maybe I should ask this question that I know has been
discussed to death.
"stable" = 1.0 release
"CVS HEAD' = 1.1 release
Is this a correct statment
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
SampsonSent: Thursday, September 15, 2005 12:17 PMTo:
Best scenario does not route faxes over the IP network as a VoIP call.
You can either use spandsp as a fax on the Asterisk box, (has problems,
but the delveloper is behind solving them)
You can route the calls to a fax server located in the same colo via
tdm. (you can use HylaFax on Linix of any
I am game.
What do you need from me???
Locked, loaded and ready to GO!!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory
Andrews
Sent: Wednesday, September 14, 2005 2:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL
I se what you are talking about I an able to reproduce!!!
However your PRI may be in a Round-Robin picking order, that would cycle
through all of the channels until it reaches an end and then it repeats.
I set our PRI to first available hunting instead of RR and it will use
the same channel
] On Behalf Of Alexander Lopez
Sent: Tuesday, September 13, 2005 9:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] PRI zap channels not cleared when no
matchincontext for dialed number on inbound call
I se what you are talking about I an able
Agents logging out is the prefered method of saying I can't be bothered
right now
If you want you can use the power of Asterisk and write something that
does what you want.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mark Phillips
Sent:
-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Wrapup time for agents.
Alexander Lopez wrote:
Agents logging out is the prefered method of saying I can't be
bothered right now
CVS HEAD also supports pause/unpause for agents, which allows
them to be unavailable without the queue
If what you are asking is that the phone you are calling from displays
'Voice Mail' when ext 1000 is dialed then that is a function of the
phone NOT of asterisk. Setting callerID would ONLY be displayed on a
phone that is ringing!!!
On the Cisco IP phones I set the messages_url to 'voicemail'
Can someone tell me how, if it has been done.
What I an looking for is the ability to have a assistant or other
authorized person 'whisper' in my ear duing a conversation. The party
on the remote end would not hear anything.
I can do a redirect into a meetme room with the person on the remote
You can use setgroup and getgroupcount (deprciated for the function) and
trigger an acton using the system Application
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Damon Estep
Sent: Monday, September 12, 2005 8:04 PM
To: Asterisk Users Mailing
Are you using the Linksys
router as your PPPoE termination or are using the Netopia??
Alex
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B. Asterisk Users
Sent: Sunday, September 11, 2005 3:46 AM
To:
asterisk-users@lists.digium.com
Did it take an interrupt??
Whats does /proc/interrupts say??
Did you check your span= settings in zaptel.conf??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Sunday, September 11, 2005 5:48 AM
To: asterisk-users@lists.digium.com
Title: Message
What does you za configs look like..
I do not think that it is a problem with SIP but rather a
problem with the way you are 'grabbing' zap channels...
What is you zapata.conf file and how are you dialing out in
the extensions.conf???
From: [EMAIL PROTECTED]
I always use _ALERT_INFO
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brian Capouch
Sent: Wednesday, September 07, 2005 12:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Working example of
How is your line provisioned?? (EW, PRI, Trunks, etc.)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Wright
Sent: Monday, September 05, 2005 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
I had the problem on a very old version of the TDM card
(brown card) I contacted Digium and after a few WTF's they sent me a shiny
new blue card that to this day is still blue!!!
Contact your reseller or Digium directly they always stand
behind their products.
I would double check for
I have the same setup. With Paetec and in Miami also..
You can call me to discuss if you like.
305-503-3000 ext 122
Alex
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA
Sent: Wednesday, July 20, 2005
1:18 PM
To:
Try prepending two _'s like this.
exten = 5000,1,SetVar([EMAIL PROTECTED])
exten = 5000,2,Goto(mailexten,s,1
It allows the variable to be exported.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Frank Schoep
Sent: Monday, July 11, 2005 4:40 AM
Try answering the line first.
Exten = 500,1,Answer()
exten = 500,2,Dial,Zap/g1/3105551010
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Syed Akbar
Sent: Friday, July 08, 2005 12:01 AM
To: 'Asterisk Users Mailing List - Non-Commercial
I am not sure about E1 but it _should_ be the same. The Dialed Number is
usually transferred in 'a whole block' as the Telco passing the call to you has
already routed that call to you. What type of switch are you connected to??
Could your switch be expecting a ACK of some sort from *??
Why not play the message BEFORE you call the Dail application. This
would also give the caller a chance to terminiate the call by hanging up
BEFORE your techs even get the call..
Hint: use the playback application
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Try _.4445454
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Modesitt
Sent: Wednesday, June 29, 2005
4:17 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Extension
Matching.
Is there a way to match the last 7
digits of
.
That worked perfectly,
this behavior must have changed recently because I tried that 6 months ago and
it did not workJ
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Wednesday, June 29, 2005
2:30 PM
To: Asterisk Users Mailing List -
Non-Commercial
.
Okay I take that back it
kinda works, but the behavior is erratic. Sometimes it matches just the 7
digits and sometimes it matches every number that enters the context J
Chris
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Alexander Lopez
Sent: Wednesday, June 29, 2005
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
trixter http://www.0xdecafbad.com
Sent: Sunday, June 26, 2005 4:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt
On Sun,
300 phones should not be a problem if you design the system correctly.
If they are all analog sets with no transcoding your should fine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Don Brearley
Sent: Wednesday, June 22, 2005 10:51 AM
To:
What signalling are you using, PRI, RBS,
What model of TNT are you using???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Baird
Sent: Wednesday, June 15, 2005 10:55 AM
To: asterisk-users@lists.digium.com
Subject:
See:
http://lists.digium.com/pipermail/asterisk-users/2004-September/063348.h
tml
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kumara
Jayaweera
Sent: Tuesday, June 14, 2005 1:46 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RJ45
I will also host a mirror. I am located in Miami, Florida.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian C.
Fertig
Sent: Tuesday, June 14, 2005 2:12 PM
To: Matt; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Can you try a 'more standard' boundary?
Like 255.255.0.0 or 255.0.0.0 ???
If so it (polycom) may not understand CIDR.
You may want to look at implementing a VLAN...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Charlie
Watts
Sent: Tuesday, June 07,
This may or may not work due to timings slips that you
may experiance with the Digium Cards. Your are correct in assuming this
scenaro.
I did the same (pre-asterisk) with an Adtran
Atlas. It is rock solid and works great. What modem access bank are
you using, there has been some talk about
That should work but you need to have the asterisk box
setup to do pri-net on the connection to the PM3. I would add the did dialed so
that the PM3 knows about it for radius accounting..
exten = 1234567890,
1, Dial(Zap/g2/${EXTEN})
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
We use Wireless b/w two office in Miami We are using the Proxim stuff
and it is solid. Two Asterisk servers doing Iax b/w them should (will)
work fine. What is the interface into the 3gsi?? Do you have a card
part number to post, that would help in determining what you need to do.
Alex
Just make sure that the Carier you use on the PSTN side will support
Modem comunications, some cariers use VoIP for the long haul
(surprise) This makes it VERY difficult to get a good connection.
Also make sure that yur clocking on the T1 is solid, Minute frameslips
cause a multitude of
The good thing about gsm files and the fact that they are
headerless is that you can simply cat files together. You just need to find the
right sound files to do so.
Then program your
dialplan to play the message before sending the person to voicemail. I would
zero out the unavailable and
I have looked for other FXO SIP Gateways and there are not many to
choose from. I found another made by clipcom, but that was about it,
other than a small asterisk server.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent:
It works for me. Do you have reinvites enabled. I do not. That may
explain why * is sending a redirect.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Boehm
Sent: Wednesday, May 04, 2005 11:35 AM
To: asterisk-users@lists.digium.com
Subject:
Funny thing is that Faxes over IP (SIP ATA186) and Fax Over Public
Internet (FOPI) have worked fine since day one. I even have faxes on
DSL lines at my house working glitch free. I have been scared to
'retire' the old OS as it has worked so well.
# ./zttest-mod -v
Objective: to read 8192
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling aka ManxPower
Sent: Tuesday, May 03, 2005 12:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Freak incidents, who's to blame?
Ryan Courtnage
I ran this test on a machine with P3/700 an got same results. See
provious post. Is anyone keeping track of this???
Alex
# ./zttest-mod -v
Objective: to read 8192 bytes from TDM card in 1.00 seconds.
Opened pseudo zap interface, measuring accuracy...
8192 bytes in 1.023984 seconds
8192
From : http://www.techfest.com/networking/wan/t1.htm
T1 has a number of other defined alarm and control signals. The alarm
signals have different color designations and are used to indicate
serious problems on the link. These alarm signals are defined as:
Red Alarm
This is a local equipment
On Artisoft PBX systems I used to use a nifty program call IMS Music on
hold (http://www.nch.com.au/ims/)
It would play loops of music and mix canned scripts for voice overs. IT
would allow you to set music on hold messages by time date and
frequency. It is a windows program but it has a free
What's the diffeance???
I just logged im and saw the same screens.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek
Whitten
Sent: Tuesday, May 03, 2005 7:48 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Nufone
Nufone is now
Polycom phones and Snom phones supoprt paging.
As far as your Overhead paging all you need is an FXO port on your
system. The * system will work perfectly with this. Even allowing the
zones to be set from the dialplan so your users won't need to learn any
new 'paging codes'
Email me off -list
As ny 10 year old step-daugher says I don't get it..
Can't you just do a redirect if you specify the channels, * doesn't care
if they are bridged together or not. You may end up with zombie
channels if the other leg does not drop, but you could do a soft hangup
and take care of that..
Or am
ChanIsAvail
Show application Chanisaval
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Elmar
Haneke
Sent: Wednesday, April 27, 2005 10:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Determinating Phone status
Hi,
how can I
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Wednesday, April 27, 2005 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Determinating SIP Phone status
Elmar Haneke wrote:
Hi,
OK a simple AGI can do this, I know you didn't want one, but its only
three lines
Cat findzone.agi
#!/bin/sh
zone=`cat $1.dat`
echo SET VARIABLE zone $zone \\\n
Put the above script in your agi-bin (usually /var/lib/asterisk/agi-bin)
chmod 755 findzone.agi
Then in your dialplan do this:
Yall' (being a southern Yankee!) should checkout the app_dictate app in
the Mantis, It allows you to replay and gives you better control for
something like this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Wednesday, April 27,
That seams to be the same issue with SpanDSP. It seams that the high
interrupt rate is slipping. In the case of the SpanDSP issue it is drop
1 out of 50 packets. This is of course with the TDM cards (fxo/fxs) not
the Single or Quad span cards. I think it may be time to look at the Zap
vased code
Check externip= in sip.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Tyreman
Sent: Tuesday, April 26, 2005 2:45 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Remote Phones - No Audio In Either Direction
Hi,
After months of
Extensions, Usernames, context, and what the device 'thinks it is' are
all different.
As SIP goes Make your sip.conf entries like this:
[compa2000]
username=companyA_2000
context=contextCompanyA
[compb2000]
username=companyB_2000
context=contextCompanyB
That will give each device a unique
Dont; forget the Milliwatt
application in Asterisk
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Keith O'Brien
Sent: Monday, April 25, 2005 4:28
PM
To: [EMAIL PROTECTED];
Asterisk-Users@lists.digium.com
Subject: RE: [Asterisk-Users]
Cisco's
ignorepat is for Zapata devices. Sip devices sned the number to the
swith AFTER the SIP device feels it has dialed it. I am not a pro on the
GS phones, (never played with them) but I would cheak the documentation
on setting up a 'dialplan'.
I hope this sets you in the right direction.
Alex
Title: One touch voicemail on Cisco 7940/60
Program
your messages key to voicemail; with:
messages_uri: voicemail
in your SIP(MACADDRESS).cfg config file
And in extensions.conf:
Exten = voicemail,1,Wait(1) ; Wait
a minute to make sure audio is up
Exten = voicemail,2,
SER is a SIP proxy, Asterisk is a PBX, and application server.
SER passes calls from place to place and does not get in the audio path.
SER uses SIP, * is able to transcode, and convert Protocols.
You can build an IVR, VM, and PBX with Asterisk. SER is like a traffic
cop, where * is the car
Carlos,
Change one line to prepend es to the filename, (ie es/cerrado). See if
that works. It may be a simple fix where the language support is broken as far
as paths go.
If is works please report it as a bug on the Bug Tracker. http:/bugs.digium.com
Please include as much info as
It is a kludge but should work:
Action: Command
command: show channels
Then sort based upon the result and you should have the two final
variables you need.
SIP/8000-? And Zap/??
You can then Monitor the SIP channel or just grab the Zap. I would go
with the former as you could then
Try running the agi from a command line. ?When I tried it a while ago,
it complained about files that it could not find, and even Googleing
neither could I.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Dent
Sent: Friday, April 08, 2005 6:46 PM
To:
That brings up a good question of Wiki Housekeeping. With the constant
changes to CVS versions of asterisk. The Wiki gets old sometimes. Would
it do a good idea to set up a Wiki-Marshall???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Stingel
Try http://www.clarent.com/
They are now Verso
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Danny N
Sent: Thursday, April 07, 2005 10:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and clarent
You may be in luck!!!
The Adtan 600 line does have a DSX-1 module available. (you gotta love
Adtran!!)
http://www.adtran.com/static/docs/64200612L28.pdf
Now all you need are a buch of IP phones and your rocking
Trash the CB plan go Digital
-Original Message-
From: [EMAIL
Daylight Saving Time confused me as well!!!
I'll make it simple:
FXO ports connect to a phone company line, can be referred to as
Office
FXS ports connect to a phone device, can be referred to as Station
What kind of features do you want in a channel bank? Not many!! Good
channel banks are
Are both cars recognized by the system?? Check in /proc/interrupts and
see if BOTH cards are there.
Also see what /proc/pci tells you.
If you are still having trouble contact Digium (only if you bought the
cards from them, if you bought the clones, you may be Out of luck!)
-Original
Try modprobe wctdm.
It looks like the drivers for the card are not loaded. You will also
need to edit /etc/zaptel.conf and /etc/asterisk/Zapata.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Hobbs
Sent: Monday, April 04, 2005 10:46 PM
To:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
G
Sent: Wednesday, March 23, 2005 10:43 AM
To: Asterisk
Subject: [Asterisk-Users] Group channel rotation for outgoing call?
Hi,
If I have a PRI with all channels grouped in group=1, I
Comedian is probably a play on 'Meridian Mail' by Nortel.
It makes for a great laugh when you drop in a replacment for the Nortel
VM system with Asterisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson
Sent: Monday, March 21, 2005 12:51 PM
Build a script, use curl or wget parse output and use the variable to
trigger events either via gotos of via the agi-script you wrote.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matias G.
Sent: Friday, March 18, 2005 11:27 AM
To:
I think you just did
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanishka Somaratne
Sent: Friday, March 18, 2005 11:15
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] reply a
post
Hi
how do i reply a
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohammed
Firdosh Nasim
Sent: Tuesday, March 15, 2005 11:08 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] How to register two SIP phones ( e.g.
WindowsMessenger) from different subnet
One MORE
Adtran Atlas 550. I have used it in service for over 3 years and it is
ROCK SOLID!!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
Sent: Saturday, January 29, 2005 9:17 AM
To: Asterisk Users Mailing List - Non-Commercial
Do you have SIP phones??
I do not have busydetect or busycount enabled.
However, I still get the same drops.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Goutam
Shaw
Sent: Thursday, January 27, 2005 12:16 PM
To: Asterisk Users Mailing List -
I think not since the audio is coming from the festival command not
asterisk.
I would try this,
Exten = s,1,SetVar(FILENAME=FESTIVAL-${EPOCH})
Exten = s,2,AGI(festival-bg|${FILENAME}|Text for festival to speak)
Exten = s,3,BackGround(${FILENAME})
The AGI script would take two args, the
Title: Re: [Asterisk-Users] E911 Testing !
The PBX craging can happen on any system. Most key system don't have UPSs or battery backups. The larger ones do but the smalll 4 x 8 systems usually don't. The best practice would be to install a POTS line and adjust your dialplan to route 911
Title: Re: [Asterisk-Users] Looking for a wireless phone... wifiortraditionalwireless ?
This senario works great! I have a Snom phone coupled with a Linksys Wireless game adapter. When on a event trip I was able to have the snom ring where ever there was a signal.
So the proof of concept
Title: Re: [Asterisk-Users] PRI concentrator
Look at the Atlas by Adtran.
-Original Message-
From: [EMAIL PROTECTED] [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Sent: Thu Jan 13 15:24:13 2005
Subject: [Asterisk-Users] PRI concentrator
Hey
Do you need my help???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Tuesday, January 11, 2005 10:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Installing * on fedora 3
G'Day List,
Can someone help me
I am also running a pretty recent version
albeit not todays CVS, but CVS-HEAD-11/20/04-11:29:52.
D you have problems b/w the Ciscos or only
when going out to the PSTN??
I have 35 7960s with a PRI and no problems
that you speak of. I do get an occational dropped call but that may be
is off and I dont use meetme or anything, its never been an issue.
I dont/have not used any extra-curricular Asterisk patches. Most of them confuse or concern me anyway.
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Alexander Lopez
Sent: Monday, January 10, 2005 12:13
OK here it goes..
Caller ID is two parts or actually three:
Part 1 Number only
Part 2 Number + Name
Part 3 Whole lotta stuff (also known as ADSI)
Here is the US, I cannot speak for other countries.
When party A places a call to Party B. Party A's Telco picks up the
number, either from a
switch thru
the STP's to a SCP which has the calling name database. The TCAP query
returns back to the launching
switch the caller name. LIDB is for operator services etc. CNAME is a
TCAP
database lookup, much
like 800 number translations.
Tom C.
- Original Message -
From: Alexander Lopez
for
calling card, operator
services, etc. These are all seperate databases stored for use in an
SCP
connected to STP's.
So is there a relationship between CNAME and LIBD, no.
Tom C.
- Original Message -
From: Alexander Lopez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
Make sure you has a span defined for each port on the TE410P. With out
signaling it would not take interrupts.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl H.
Putz
Sent: Monday, January 10, 2005 12:38 AM
To: Asterisk Users Mailing List -
You are using a PRI based config for POTS lines. It will no worky. Post
your zap*.conf files.
I'll take a look at them for you..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Altus
Snyman
Sent: Monday, January 10, 2005 1:24 AM
To: asterisk
Subject:
List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TE110P error
On Mon, 2005-01-10 at 01:33 -0500, Alexander Lopez wrote:
You are using a PRI based config for POTS lines. It will no worky.
Post
your zap*.conf files.
I'll take a look at them for you..
How do you plug analog lines
Title: Re: [Asterisk-Users] virtual pbx
Asterisk IS sleady there! Understand the dialplan and the various settings in voicemail.conf and you got it.
-Original Message-
From: [EMAIL PROTECTED] [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Title: Re: [Asterisk-Users] Best gateway to use for *?
I have to date NOT had a problem with the Digium HW. You just got to pick the right Mobo.
-Original Message-
From: [EMAIL PROTECTED] [EMAIL PROTECTED]
To: [EMAIL PROTECTED] [EMAIL PROTECTED]; Asterisk Users Mailing List -
Title: Re: [Asterisk-Users] can the dialtone be changed after pressing 9?
Yes you can but it only works for zap devices. IP based would be a function of the hardware.
-Original Message-
From: [EMAIL PROTECTED] [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Title: Re: [Asterisk-Users] Ringing an extension on multiple phones
There are several options here.
You can set up a queue and have the phones ring un the order you like.
Setup an additional extension on every phone.
Set up an AGI script that allows them to login to the receptionist
As per Caleer ID spec. Caller ID info is tranmitted between first and
second rings. This is to allow the phone to 'wake up' and receive the
Caller ID information. If you were to pick up the phone right before it
rang the first time or shortly after the first ring stopped you will not
get caller
Look at canreinvite= in the sip.conf.
If you remove Asterisk from
the stream them you are using Asterisk more like a Proxy and less like a PBX.
If this is the case and you want to support tons of users look at
something like SER. Asterisk is not a Sip proxy but rather a PBX and Media
What started out as a good thing for the community has veared it ugly
head and will come back to bite us in the ass. I give my respect to the
two companies that decided to put themselves 'out there' and attempted
to bring 'real world' certifications of knowledge in an area that is
unregulated,
Agreed, You have a strong point about the Monopoly aspect of the whole
thing. My .02 would be to have this be a Digium product. Heck, Mark DID
invent the thing and HE holds the copyright to it. I have faith in Mark
and what he can do when he gets back from France.
When I taught Sun and SCO
OK with CVS-HEAD-12/22/04-09:47:32 incoming (to ata) sip calls on a
ATA0186 now seam to work fine. However transfers still do not work.
With CVS-HEAD-12/22/04-12:46:47 transfers still do not work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tracy
Discussion
Subject: Re: [Asterisk-Users] Cannot transfer with Cisco or Snom
On Wed, Dec 22, 2004 at 01:33:35PM -0500, Alexander Lopez spake thusly:
OK with CVS-HEAD-12/22/04-09:47:32 incoming (to ata) sip calls on a
ATA0186 now seam to work fine. However transfers still do not work.
With CVS
Russell,
What kind of zap cards do you have??
If T1, is it PRI or RBS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russell
Horn
Sent: Wednesday, December 22, 2004 6:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
401 - 496 of 496 matches
Mail list logo
