Any phone you can plug into a regular POTS PSTN line from your Telco
should work with the TDM400. Don't expect the fancy function buttons to
work, however.
On Wed, 2004-04-07 at 14:49, Gregory Junker wrote:
What about the Partner phones and TDM400?
You can't plug Lucent's (Avaya's) DCP,
Hate to reply to my own message again, but I just figured it out.
Nothing wrong with asterisk really, just a bad configuration. Somehow
the queue line in extensions conf got changed by someone to:
exten = 81003,3,Queue(receptionistq|tTH||10)
Thats where the 10 was coming from. :) Could this
Take out the allow=all in your sip.conf and put in allow= for the codec
you want to use and disallow=all.
On Wed, 2004-04-07 at 15:18, Roger wrote:
I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone
are registered by the below information
*CLI sip show peers
I'd take a look at the VoiceTronic cards ( http://www.voicetronix.com/hda.htm ) which
can be used with * or their free software.. these cards can be configured as :
12 Loop-Start ports only.
8 Loop-Start AND 4 Station ports.
4 Loop-Start AND 8 Station ports (default configuration).
12 Station
I have a situation where calls come in via SIP from VoicePulse and get dropped
into a main menu.Voicepulse only works /w dtmfmode=inband and I only
allow ulaw/alaw as codecs.
When the call comes in and gets dropped to the menu, you can hit 1, 2 or 3 to
get to other people. Or you can
This is a fairly simple thing to do. You don;t say what type of phones you are using,
so I;ll assume SIP for the example:
Step 1:
Put
callerid=Darren 1234
for each phone definition in sip.conf, obviously replacing Darren with the user eg
Darren Nay or Joe Bloggs, then replace the 1234 with
On Tue, 6 Apr 2004, Mark Spencer wrote:
I've been considering the nature of Asterisk, its security, the bug
tracker, and more... And i've come up with an interesting idea: A
message of the version. The idea is that Asterisk has a compile time
32-bit unsigned int version which is incremented
This may be a little to far into PBX land but...
Anyone know of a place where there are good examples of how
to configure the Definity PBX stations with PRI? I currently have a T1
between a Definity and Asterisk. It is currently doing robbed-bit
signalling but
I would like to do PRI. I can
I guess I didnt place that part of
my message in the correct context. Presence is very handy, and I would
like to see the functionality added to Asterisk. What I meant by my
comment you quoted below, is that if I could attach my Asterisk server to the
Skype network, I would not care if I
Can someone take a look, tell me if this is a bug, a
possible resources issue, or my own damn fault?
http://bugs.digium.com/bug_view_page.php?bug_id=0001381
Thanks,
Derek
Dean Collins just sent out a message a second ago (responding to an earlier
posting regarding the new Skype PDA client). He said:
Presence based information is the biggest 'seller' in the IP PBX market at
the moment, being able to tell what/where a person is certainly driving a
lot of sales
Also, check out www.citel.com This company claims to have SIP adaptors for
Avaya's digital PBX phones. If they work as advertised, you can keep your
Avaya/Lucent phones, throw out your legacy PBX, and connect them all to
Asterisk! However, I doubt they have all the display integration working
On Wed, 2004-04-07 at 15:44 -0500, Eric Wieling wrote:
Don't expect the fancy function buttons to
work, however.
That's specifically what I was asking about...
Has anyone tried to decipher the ETR signaling protocol? Or is it such a
closely guarded Lucent/Avaya secret as to make the formula
On Apr 7, 2004, at 4:23 PM, Darren Nay wrote:
My question is. Is there a way to make asterisk aware of the
calling-from (callerID) number so that it will automatically detect
the number and then go directly to asking them to input their
password.
From show application VoicemailMain try:
I'm emailing this as the customer in this case, since my carrier appears
to be completely unable to solve this. A brief rundown of the problem:
- We have several voice lines going through the ADIT, of course into a
VoIP type of arrangement.
- Voice traffic will become choppy, even drop calls
It cant be that hard to do considering Siemens
are offering a cordless handset that can connect to skype.
I guess its just a matter of
bridging the 2 together.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Hall
Sent: Thursday, 8 April 2004 7:20
AM
Doesn't work for me. Connects to Asterisk but says All extensions are busy
right now when I try to do anything. Here's what an extension looks like.
Any suggestions? Thanks!
Extension
NameExt 2003/Name
Number2003/Number
DeviceSIP/2003/Device
Contextfrom-sip/Context
On Wed, 2004-04-07 at 15:44 -0500, Eric Wieling wrote:
Don't expect the fancy function buttons to
work, however.
That's specifically what I was asking about...
Has anyone tried to decipher the ETR signaling protocol? Or is it such a
closely guarded Lucent/Avaya secret as to make the
Eric Wieling wrote:
Take out the allow=all in your sip.conf and put in allow= for the codec
you want to use and disallow=all.
Holy crap it worked!
sip.conf
disallow=all ; disallow all codecs
allow=ulaw ; Allow all codecs
allow=alaw ; Allow all codecs
Right, I know that the voice part is POTS because I have a standard
cordless phone plugged into our Partner system.
Hmm, wouldn't ETR be covered under a patent and not a copyright? And has
17 years been up yet?
And if someone is selling devices that convert to/from ETR, then the
protocol spec
Hi,
I had exactly the same symptoms today with a co-located * connected to a
Public Switch here in the UK. The problem was solved by insisting that the
Telco turned on CRC4 at their end and then, after an 'init 6', layer two
settled down on both systems.
I was taught that if you are connecting
Ok its probabally something really eaisy im missing. I've searched the
archives and voip-info.
Asterisk is trying to send the email notification for voice mail. But the
log says Invalid sender. Sender = [EMAIL PROTECTED] and not
[EMAIL PROTECTED] as assigned in conf file.
VM Config:
YES PLEASE.
Wonderful Stuff! In my opinion just what the project needs. I deployed and
supported many GPL and commercial SmoothWall (firewall) installs and was
forced to poll a web page from time to time to see if any of my customers
needed an urgent security patch applying...not a satisfactory
Title: [Asterisk-Users] Presence
I have to agree.
A large number of people are looking for this feature. I have written a web script that can show Agent logged into the system.
I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product.
There is
Shad Mortazavi wrote:
I think integration/gateway between Asterisk and Jabber would be a
amazingly wonderful product.
firefly, while not 100% bug free I think it has this feature, although I
haven't played with it enough to work out how to show someone as being
online...
--
Best regards,
You should be trying to telnet to the ip address of your asterisk server.
Christian Hoffmeyer
YottaDot Solutions
Huntsville, AL
(w) 256.859.4508
(c)256.655.0321
(iax) 700.859.4508
Ask me about Asterisk
- Original Message -
From: Jain, Sonal [EMAIL PROTECTED]
To: [EMAIL
- Original Message -
From: John Vogel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 07, 2004 5:37 PM
Subject: RE: [Asterisk-Users] WAMi - Windows Asterisk Manager
Doesn't work for me. Connects to Asterisk but says All extensions are
busy
right now when I try to do
They modified iax to include the presence packet but only works on their
customized firefly network. I was thinking along the lines of a software
app for those of us who use hardware phones but still want to keep TXT
chat and presence and perhaps integrated into 1 of the iax soft phones
as well to
Hi,
I'm trying to get CDR recording via MySQL working. I've setup my
database and compiled the asterisk-addons from cvs and setup the config
file, but when i start asterisk i get this:
[cdr_addon_mysql.so] = (MySQL CDR Backend)
== Parsing '/etc/asterisk/cdr_mysql.conf': Found
Apr 8
William Suffill wrote:
They modified iax to include the presence packet but only works on their
customized firefly network. I was thinking along the lines of a software
app for those of us who use hardware phones but still want to keep TXT
chat and presence and perhaps integrated into 1 of the iax
Sounds like an error in your config file. Want to paste the contents
in? Thanks...
Sean
-Original Message-
From: Jeremy Bogan [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 07, 2004 8:06 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] MySQL CDR
Hi,
I'm trying to get CDR
Sounds like an error in your config file. Want to paste the contents
in? Thanks...
Sorry:
;[global]
;hostname=localhost
dbname=asterisk
password=password
user=asterisk
;port=3306
sock=/tmp/mysql.sock
userfield=1
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
;[global]
Never mind i'm an idiot. I've commented out [global], whoops!
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Remove the semi-colon in front of [global]
HTH,
Ryan Thrash
On Apr 7, 2004, at 8:15 PM, Jeremy Bogan wrote:
Sounds like an error in your config file. Want to paste the contents
in? Thanks...
Sorry:
;[global]
;hostname=localhost
dbname=asterisk
password=password
user=asterisk
;port=3306
At 8:29 PM -0400 on 4/7/04, Shad Mortazavi wrote:
I have to agree.
A large number of people are looking for this feature. I have
written a web script that can show Agent logged into the system.
I think integration/gateway between Asterisk and Jabber would be a
amazingly wonderful product.
On 05-04 14:35, Steven Sokol wrote:
TCP/TLS would be used for the SIP messaging which handles call setup,
teardown, and other non-Realtime functions. The voice stream will still be
handled via RTP which is a UDP-based protocol.
The reason for doing the call setup as TCP is to allow for TLS
Duane wrote:
William Suffill wrote:
They modified iax to include the presence packet but only works on their
customized firefly network. I was thinking along the lines of a software
app for those of us who use hardware phones but still want to keep TXT
chat and presence and perhaps integrated
[apologies for top-posting]
I am very interested in what providers typically take CNAM via ISDN.
I have some experience with PRI providers, but I've never heard of
one offering that service.
If you are a PRI provider in the lower 48 who takes CNAM and can pass
that off to the PSTN, please get
I'm not familiar with the protocol used in Firefly. If that was known
then it would be possible to add the functionality to * so anyone can
have the simple presences by dialing extensions in their dial plan or
crafted packets at a software level. Jabber is already deployed in my
organization so I
Quoting John Todd [EMAIL PROTECTED]:
[apologies for top-posting]
I am very interested in what providers typically take CNAM via ISDN.
I have some experience with PRI providers, but I've never heard of
one offering that service.
If you are a PRI provider in the lower 48 who takes CNAM
Currently the plan is to forward all PSTN calls on our 2 incoming PSTN
lines 2 remote toll free's via IAX2 to staff.
3 different delivery methods
1) Users local to the office where the lines come in with GS/PSTN phones
2) IAX2 to remote location * server then Cisco 7960 on that lan
3) IAX2 to
I am looking for a ISDN BRI card (u-INTERFACE) to connect * to a US 5ESS
switch (Qwest).
According to Qwest they support CNAME delivery on their 5ESS switches.
Does * chan_capi support CNAME ?
Regards.
Alfred.
___
Asterisk-Users mailing list
[EMAIL
At 9:50 PM -0400 4/7/04, Ray Burkholder wrote:
From: Ray Burkholder [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Quick Caller ID and Voicemail ?s
Quoting John Todd [EMAIL PROTECTED]:
[apologies for top-posting]
I am very interested in what providers typically take CNAM
It's probably sending the domain as the domain setup on the * server...
Change host to somedomain.com and see if that helps...
Thanks,
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Posted At: Wednesday, April 07, 2004 7:32 PM
Posted
I'm also looking for the same thing: ISDN-BRI U interface.
Thanks.
-brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alfred R. Nurnberger
Sent: Wednesday, April 07, 2004 9:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] ISDN BRI
Ralph,
My experience with the Adit (going on 3 years), it's a solid platform. I've
also had one backed into * using MGCP to CMG card for a bit with great
success.
The thing you said about the bandwidth usage and call quality sucks for both
300KB and 1.5Meg is the clue to this problem being a
On Fri, 2 Apr 2004 09:32:34 -0500, Adams, Gavin wrote:
We run at 1600x1200, 96 buttons would be useful.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nicolas Gudino
Sent: Friday, April 02, 2004 9:26 AM
To: [EMAIL PROTECTED]
I just wanted to ask about using Adtran boxes to support analog lines
into an Asterisk box. Right now x101p's are just too sensitive to RF
noise inside the PC. Going with an external chassis looks like a good,
albeit expensive option. It looks like I can use the Digium T1 card
into an
Title: SIP -- PSTN gateways
So what are people using these days for SIP or IAX to PSTN gateways.
1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow you to use your own SIP device (phone or something like *) instead of the interface hardware they usually provide?
On Wed, 7 Apr 2004, Michael T Farnworth wrote:
It appears that the final argument to all these functions (normally a 0
or 1) has been dropped, but it hasn't been fixed in chan_oss.c or
chan_alsa.c.
The easy fix is just to drop the final arguments for all these functions
and then to kick
On Wed, 7 Apr 2004, Brian Cuthie wrote:
So what are people using these days for SIP or IAX to PSTN gateways.
I'm setting up my own gateways. I'm getting a Cisco 2621XM with a HDV
module. I had high hopes for the Ovislink gateways, but they discard
proxied SIP requests, for some unknown
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