I found the price. $450 :-/
Kevin P. Fleming wrote:
Evert Meulie wrote:
That unit looks VERY promising! Thanks! :-)
Would anyone happen to know an approx. price for a unit like this?
Anyone? I bet the manufacturer of the unit would know a price for it,
and it's probably even
Hi all!
I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which act as a gateway, but what I'd really like is a for example an Asterisk module that can route calls to Skype, perhaps the same
principle as IAX2?
I'm assuming more people are interested in this, but... does it
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
This is not an indication of a memory leak. The size of the asterisk
process:
ps `cat /var/run/asterisk/asterisk.pid` -o vsize -o rss
Is this what you were talking about?
[EMAIL PROTECTED] ~]# ps `cat No such file or directory-o
Hello users,
What is the meaning of this message?
Dec 19 09:19:28 WARNING[15112]: interface.c:215
decodeMP3: Junk at the beginning of frame 54414700
Thank you
Dov
___
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Asterisk-Users mailing
On Sat, Dec 17, 2005 at 07:44:29AM -0600, Rich Adamson wrote:
ok rick all of my conf...
asterisk 1.2.1
zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side
are xten with SIP.
my extencion.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
L.S.
I'd transported a pretty new snom 220 in a weekendbag, and the lcd display
did not survive.I know, I know...
If a kind soul has an otherwise broken 220 you would make me very happy
with a display circuitboard. Or maybe a link to a secondhand shop?
Cheers!
Loek atsign Gijben.nl
try separating the values of tx/rx gain in zapata.conf
ex:
txgain= -2.5
rxgain= 10
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
Best Regards
Greg Cirino
[EMAIL PROTECTED] Virus Spam Free
and you can't do better than that!
http://www.cirellemail.com
Cirelle Enterprises Inc.
25
hi
i have GXP-2000 ( grandstream ) and and i am trying to press key fron phone
keypad when i hear greating message and asterisk asks me select one
extention ( i have backgroud function in my extentions.conf ) ,
with grandstream asterisk dosnt receive anything from ip-phone , but with
On Monday 19 December 2005 06:44, Cirelle Enterprises wrote:
try separating the values of tx/rx gain in zapata.conf
I think you missed the part where this is pure-voip, no zap channels
involved. :-)
-A.
___
--Bandwidth and Colocation provided by
On Mon, December 19, 2005 11:33, Evert Meulie said:
Hi all!
I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which act
as a gateway, but what I'd really like is a for example an Asterisk module
that can route calls to Skype, perhaps the same
principle as IAX2?
I'm assuming
Hello,
Polycom ip soundpoint support ACD login/logout .
Can we configure asterisk with polycom ACD support?
Regards
Harry
___
Nouveau : téléphonez moins cher avec Yahoo! Messenger !
Hi Fernando,
Have you tried connecting asterisk directly to the R2 link? Many of
these converter boxes are a pain to work with, as they don't give you
much information when things are not right.
Regards,
Steve
Fernando Romo wrote:
Dear comunity:
I use a r2mf converter called SignalPath
My point exactly.
I'll take a look at that script though, if I could automate that each
night then it might be fine, tag the imports, clear out those then
re-import again.
Thanks
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
On 12/18/05, Douglas Garstang [EMAIL PROTECTED] wrote:
Hi Tyler.
We're registering users with OpenSER, which also routes the calls to a series
of Asterisk systems. The really tricky part is allowing different phones
entering through different Asterisk systems to reach other. Currently, the
Hey,
That is not what I meant
I L O V E ASTERISK, every other PBX I have had to deal with, always had
some limitation, I am only using 1.0.7 and really have found nothing
limiting.
We run ISDN 30 line, Reception get 440+ calls per day, dial out 23,000 calls
per month all fully integrated
[EMAIL PROTECTED] wrote:
Polycom ip soundpoint support ACD login/logout .
Can we configure asterisk with polycom ACD support?
For the uninitiated among us (myself included) what is ACD login/logout
support?
Thanks,
Matthew
___
--Bandwidth and
ok rick all of my conf...
asterisk 1.2.1
zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side
are xten with SIP.
my extencion.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
CONSOLE=Console/dsp
i have GXP-2000 ( grandstream ) and and i am trying to press key fron phone
keypad when i hear greating message and asterisk asks me select one
extention ( i have backgroud function in my extentions.conf ) ,
with grandstream asterisk dosnt receive anything from ip-phone , but with
Matthew wrote:
For the uninitiated among us (myself included) what is ACD login/logout
support?
The Polycom phones can send XML NOTIFY messages to signal to the server
the agent is logged in/out/paused. I know of no documentation on the
messages (although they don't look that hard to
When ACD is used the queues and agents are configured
so agents have to send agent id and password to become
available in a queue .
Harry
--- Matthew matthew@zeut.net a écrit :
[EMAIL PROTECTED] wrote:
Polycom ip soundpoint support ACD login/logout .
Can we configure asterisk with polycom
On Mon, 2005-12-19 at 07:21 -0600, Kevin P. Fleming wrote:
Matthew wrote:
For the uninitiated among us (myself included) what is ACD login/logout
support?
The Polycom phones can send XML NOTIFY messages to signal to the server
the agent is logged in/out/paused. I know of no
So we have to add a context like this to login/logout
agents.
I add 4 agent in a queue with roundrobin strategy .
What's going on if the first available agent don't
answer the call ?
Asterisk-1.2
[agents]
;Agent Login
exten= 501,1,AgentCallbackLogin(||[EMAIL PROTECTED])
;Agent Logout
exten=
Adam Goryachev wrote:
Could chan_sip simply start executing the DP at a particular
extension ?? or would that require the existence of a channel, which
there isn't really since it is just XML not RTP???
chan_sip _could_ do anything at all. However, since these are not INVITE
requests,
We proudly announce that VoiceOne has just been released as an ALPHA version
at http://www.voiceone.it
Main features:
* Client/Server architecture based on Web services
* Relies on Asterisk Realtime Architecture (ARA)
* SIP extensions management (support for Zap/IAX/mISDN soon added)
exten = 9,1,NoOp(ISDN: Pickup outside line (early B3 connect) for: ${CALLERIDNUM})
exten = 9,2,SetCallerId(${THORCOM_MAIN}) exten = 9,3,Dial(CAPI/g1//b) exten = 9,4,Hangup
use this string with BT
extn = 9,3,Dial(CAPI/g1//bo)
Should provide correct progress
Hello,
I have a single line to receive fax and voice.
I add faxdetext in zapata.conf and
[pstn]
exten = s,1,Answer
exten = s,2,Queue(MyQueue|tn||100)
exten = s,3,Hangup
exten = fax,1,Dial(Zap/g2)
However when fax tone is detected both phones in queue
and the modem of Hylafax server answer the
I'm trying to connect to another PBX via an T-1 interface. I have a
T100P card.
On the CLI I get the error Everyone is busy/congested at this time
(1:0/0/1) When I try to dial out of the T-1 line from an SIP softphone.
I have posted this question a few times here and at the asterisk forum,
Which standard for ACD login/logout ?
--- Kevin P. Fleming [EMAIL PROTECTED] a écrit
:
Adam Goryachev wrote:
Could chan_sip simply start executing the DP at a
particular
extension ?? or would that require the existence
of a channel, which
there isn't really since it is just XML not
On Monday 19 December 2005 14:23, Francesco Peeters (Asterisk) wrote:
On Mon, December 19, 2005 11:33, Evert Meulie said:
Hi all!
I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which
act as a gateway, but what I'd really like is a for example an Asterisk
module that
hi
i have tested it with sip info option in grand stream as DTMP relay and
dtmfmode=rfc2833 and it works , that's ok but problem is this i cant ask
users change their DTMP on their ip phones, so i should use auto on asterisk
to detect who is comming with which DTMF mode,
i change
Hi,
We're working with asterisk 1.2.0, hardware sip phone (Thomson st2020 by
example), and sip soft like x-ten or snom 360 (who can both manage
many lines). We are also using the queue with round-robin strategy and
dynamic members.
When the hardware phone is busy, the call is redirected to
Michael,
Does the zttool program show the PRI as working correctly?
Can the PBX push calls into the Asterisk system?
Also, what type of PBX is it, and is it providing the clock etc.. For
the T1 connection?
-Jonathan
Jonathan O'Connor
Senior System Administrator
Inoveris LLC
Direct Line
Why oh why would you want to install *, which runs on Linux, on a
machine made by a company that does NOT support Linux? Both IBM and HP
do a pretty good job of supporting Linux. So do other Linux oriented
companies like PenguinComputing.com
Digium cards have historically been a little finicky in
Yeah, the zttool program shows the PRI as having No Alarms. It is
an Infinity system by Amtelco. I haven't actually tried making a call
from the other pbx, but I did have my vendor (Amtelco) look at it and
they verified that the span was up and everything was working
correctly. The asterisk
Well, I formated the system installed slackware 10.2 compiled a new
kernel with 2.4.31, modprobe usb-uhci, install zaptel, no problem,
modprobe ztdummy, modprobe zaptel. Got this in lsmod
ztdummy 1656 0 (unused)
zaptel219520 14 [ztdummy]
usb-uhci
I unsubscribed but I am still getting
emails to this account.
Please remove [EMAIL PROTECTED]
I think that the moderator added my reply
to address to the list.
Thanks for your help.
-Jason
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Sampson
Sent:
The parameter in zaptel.conf that sets up timing etc
is:
span=1,1,0,esf,b8zs
The first 1 means this is span 1.
The second one defines the timing of the link. For asterisk to provide the
timing use 0 instead. For instance my Asterisk box,
hooked directly to my Avaya G3 uses:
My other pbx vendor told me they supported pretty much all of the
switchtypes and that the system would automatically detect the correct
one. I've tried Qsig and National and both seem to bring the span up
fine.
I just switched to span=1,0,0,esf,b8zs
to have asterisk provide the timing. That
No matter what I try, I can't seem to pass variables to the local channel
when using the Originate command through PHPAGI. Here is a small snippet of
the PHP code:
/***
**/
$channel = 'Local/[EMAIL PROTECTED]/1';
$exten =
The only other thing I can think of is that your
contexts etc... need checked.
It would be very helpful to know if calls can come into
the system from the PBX, that would be the only way to know the span is alive
and well truely. Once you know that then its down to the contexts and
When I used MixMonitor, and I hangup the channel, Asterisk exit whit error in segment. Why? Thanks
Yahoo! Messenger: chiamate gratuite in tutto il mondo ___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
Hi all,
I've got an * sitting behind a linux iptables firewall. I have an
account with teliax. After entering the registration information
accurately and restarting *, iax2 show registry shows a registered
status on that connection.
However, whenever I try to place a call, I get a No one
nr k wrote:
HI all
How to configure voice mail in asterisk . pls do the needful.
regards
ramakrishnan.n
here is the wiki page on voicemail
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
Tom
--
This message has been scanned for viruses and
dangerous
Lists wrote:
I have gotten the tftp server working and the 9133i is doing a firmware
update and finds the aastra.cfg file as well as the 00XXX.mac file. The
issue is that I can't figure out what is wrong in the configuration files
that it is not loading the extension, proxy, etc. info.
Could
($dialstatus=ANSWER) replace with $dialstatus eq ANSWER
perl treats = as an assignment operator. For comparison, you need eq or == .
-apu
On 12/18/05, Code Lover [EMAIL PROTECTED] wrote:
Hi all,I wanted to execute one of mySQL query when the call is answered itried with the following code but it
do i need any ports open inorder to use send mail from behind a router
[EMAIL PROTECTED] wrote:
Send Asterisk-Users mailing list submissions to
asterisk-users@lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
As it turns out I can dial from the Infinity PBX into the Asterisk
box. So it must be something to do with contexts or configs I guess.
So when I set up the ZAP trunk in AMP it automatically did it as
ZAP/g0. Well I just assumed that was the first one. Most spans
numbering that I deal with
from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html
The scam isn't new, and its certainly not limited to home 800 numbers.
The same basic principles were used by some of the 900 number folks
a few years ago as well.
My fear wasn't that someone would stuff phony charges on my
We can call out almost every number. But when calling to numbers with automatic
attenders the
asterisk returns a NO ANSWER as dial status, like the number doesn't exist. Can
any one help as?
We have no idea about was it's happening.
We are runnig an Asterisk 1.2 with a TE210p digium card.
thx
I have a question about how to set up some variables. I have
an extension that my teachers are going to dial to record their information
messages for the public. I need to know what I need to put in step two for my
teachers to enter 6 digits on their phone and have that saved as a
On Mon, 2005-12-19 at 10:13 -0800, Wolfgang S. Rupprecht wrote:
from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html
The scam isn't new, and its certainly not limited to home 800 numbers.
The same basic principles were used by some of the 900 number folks
a few years ago as
No, I don't know why. But, the dtmf mode used between the phone and asterisk
stays the same regardless of where you call. That part of the call doesn't
change. Once asterisk has the dtmf digit(s), the next channel (eg, zap, iax)
will use whatever it deems to be the correct dtmf mode (unless you
What hardware/?
Sorry I have missed part of the message?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Evert Meulie
Sent: Monday, December 19, 2005 4:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Does hardware like this
I've got an * sitting behind a linux iptables firewall. I have an
account with teliax. After entering the registration information
accurately and restarting *, iax2 show registry shows a registered
status on that connection.
However, whenever I try to place a call, I get a No one is
I have the same problem !!
:-(
2005/12/18, Mohammad Shokuie [EMAIL PROTECTED]:
Hi there,
Any one confronted a crash in asterisk when using mixmonitor app. When i'm
using the mixmonitor app on a briged call as soon as the called party hangs
up the call asterisk crashes and the process
I am trying to configure zapata.conf to handle distinctive ring.
Everytime someone calls my main number, I get a ring pattern of 0,0,0
which works consistently. The problem is that every time someone calls
one of the other phone numbers (same number each time), I get a
different ring pattern
I think the broken pipe issue is related with the mpg123 player,
try disabling moh and see if it behaves the same way
On 12/19/05, Maximiliano J. Goldsmid [EMAIL PROTECTED] wrote:
I have the same problem !!
:-(
2005/12/18, Mohammad Shokuie [EMAIL PROTECTED]:
Hi there,
Any one
On Mon, Dec 19, 2005 at 06:36:16AM -0600, Rich Adamson wrote:
ok rick i will check this directives and write you again.. thanks
ok rick all of my conf...
asterisk 1.2.1
zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side
are xten with SIP.
The paper is definitely interesting and I commend them for their effort
but it doesn't represent a complete understanding of the Skype protocol
to the extent that an Asterisk server could speak the Skype protocol.
They say that much of the Skype protocol is encrypted and needs to be
inferred
Is there a way to use the manager interface to originate a call between
an outside phone number and an inside queue?
I've thought of maybe just doing it by originating the call between the
outside phone number and an outside DiD that gets routed to a queue, but
that seems messy.
Thanks for your
yikes,
25 and 110 will allow mail - but please without the whole digest attached.
And wouldn't your question be more useful with a better subject field? now
no one will see me addressing a question I know an answer to.
___
--Bandwidth and
Mark Hulber wrote:
The paper is definitely interesting and I commend them for their effort
but it doesn't represent a complete understanding of the Skype protocol
to the extent that an Asterisk server could speak the Skype protocol.
They say that much of the Skype protocol is encrypted and
Everyone should simply uninstall Skype and switch to the Gizmo project
because it interfaces quite nicely with Asterisk.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
Does anybody know of any VoIP provider in Switzerland (or other Euro country
not far from it) that could give me a DID with VPN termination. What I need
is to have a SIP or IAX connection encrypted inside a VPN (ipsec preferably)
to make and receive calls. Fax support would be a huge plus.
For testing purposed I want to simulate a CO line into my channel bank
using an FXS ATA. I thought I could use a Sipura 1001 ATA to feed to the
FXO module of the Adtran 750 channel bank, FXS feeds so why does this
not work? Does the channel bank rely on seeing something different that
an
note:
What I have found is that when many people say bad, they really mean
different.
(from training staff on Asterisk systems)
PaulH
- Original Message -
From: James Sturges [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
I have four DIDs. 2400,2401,2402, and 2403
There is no phone attached to 2400 but the other three DIDs do have
phones attached
All the four DIDs have their own voicemail and voicemail works on all
the DIDs. When you dial 2400 it rings the other three numbers. If no
one picks up, it goes to
Hi,
Has anyone used a Digium PRI card in an IBM eServer x346? I know that
Digium's website lists the x345 as having problems, but I am restricted
to buying only IBM eServers for this possible project.
I would like to use the TE411P
Harry
Have it send an email to everyone that there is a voicemail in the box, then
anyone can log into the voicemail system and retreive it
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
I geuss this explains why you should test :)
It should work, you miscofigured something.
On 12/19/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
For testing purposed I want to simulate a CO line into my channel bank
using an FXS ATA. I thought I could use a Sipura 1001 ATA to feed to the
FXO
The 8 byte chunk size is used in zaptel because:
1. For lowest cost echo cancellation in software you need the lowest
possible delay between outbound and inbound data streams, which demands the
minimum possible buffering.
2. The 8 byte chunk per channel is the source of the millisecond timer
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:
Hello,
I have two questions.
First, Can I redirect calls with a regular sip-phone like ZyXEL Prestige 2000W,
typing something like #0#SIPNUM# or another key-combination?
I have a queue configured. How can I configure it to jump to voicemail when
pressing like, # when in the queue? and is
Hi,
I'm using a td400p
card with an FXO portand asterisk 1.2.1 in South Korea and asterisk isn't
detecting whenPSTN callers hangup.
I've gone through
all the settings related to hangup detection and none work. I've
tried:
hanguponpolarityswitch=yes
callprogress=yes
busydetect=yes
C F wrote:
I geuss this explains why you should test :)
It should work, you miscofigured something.
I am testing, it is a test setup. The ATA rings a phone connected to it,
and the Channel bank answers a call on an incoming CO line. Any ideas?
--
Chris Mason
NetConcepts
(264) 497-5670
We can provide this with IPSEC DES - 3DES or AES encryption
We are based in Strasbourg ( near Basel - Switzerland )
Best regards
Thierry
tél: +33 (0)3 90 40 06 75
fax: +33 (0)3 90 40 06 75
email: [EMAIL PROTECTED]
web: http://www.widevoip.com
-Message d'origine-
De : [EMAIL
Hello,
Does it make a difference for the NAT traversal capabilities of Asterisk
if the users are registered directly to Asterisk or if they are
registered to a SIP proxy which just relays the calls to Asterisk?
Are there any cases where Asterisk would be able to traverse the NAT if
the user is
Hi Chuck,2005/12/17, Chuck Bunn [EMAIL PROTECTED]:
If you do not have QOS assigned to the SIP protocol it is quite possiblethat there are packets time outs and the packets are discarded. Is itpossible to test the network during the evening or at a time whentraffic is at it lowest?
It took me some
Well this seems doable although can extensions be contexted? Meaning
that can you have an extension of 100 for company A and the same for
company B?
Scott.
On 6/30/05, Iqbal [EMAIL PROTECTED] wrote:
Hi
If I have ser sending calls to asterisk, is there a way to get a
different block called
Thanks for this information!
Lan
- Original Message -
From:
ram
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, December 16, 2005 8:50
PM
Subject: Re: [Asterisk-Users] A2billing
Trunk
Hi
check this URL will help you for
Hi everybody!
Jonathan wrote:
Hi,
I'm using a td400p card with an FXO port and asterisk 1.2.1 in South
Korea and asterisk isn't detecting when PSTN callers hangup.
I've gone through all the settings related to hangup detection and none
work. I've tried:
hanguponpolarityswitch=yes
could you please tell how it interfaces with Asterisk? Could I receive calls
into Asterisk? send calls out?
I've just downloaded it and am searching (unsuccessfully) for these on
Gizmo's site/software.
- Original Message -
From: Kerry Garrison [EMAIL PROTECTED]
To: 'Asterisk Users
Yes you can send and receive calls via Asterisk.
http://voipspeak.net/index.php?/content/view/19/28/
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAIL PROTECTED]
Hi All,
The setup I have is an asterisk box with a sipura spa 2000 for the fxs ports
and a TDM400P with 2 fxo ports.
Anyway, I recently upgraded from 1.0.9 to 1.2.1 and have found a very annoying
problem.
One of my dialplans includes this statement in order to distinguish between
the 2
On Mon, Dec 19, 2005 at 02:56:30PM -0800, Kerry Garrison wrote:
Yes you can send and receive calls via Asterisk.
http://voipspeak.net/index.php?/content/view/19/28/
so let me understand.
One nice feature of skype is the excellent (for the user; i
understand the sysadmin may see this as a
You may want to check this out:
http://www.digium.com/asterisk_handbook/agentlogin_queues.html
A number of our clients use it to analyze outgoing calls from within
QueueMetrics.
Yours
l.
In data Mon, 19 Dec 2005 20:57:54 +0100, Jim Miller
[EMAIL PROTECTED] ha scritto:
Is there a
I don't know exactly how it works, but since it appears to just be SIP, I
would have to assume a STUN setup. I haven't bothered to sit there and watch
the packets go by to see what its doing under the hood.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
On 14:56, Mon 19 Dec 05, Kerry Garrison wrote:
Yes you can send and receive calls via Asterisk.
http://voipspeak.net/index.php?/content/view/19/28/
Is there any change you can provide the sip.conf and
extensions.conf stuff this generates?
I'm not an amp user, nor do I want to use it just to
I've got my Handytone 486 registered fine with SIP. Using an analog phone attached to it, I can dial 2 digit extensions in the main context just fine. I have a DID mapped to it from the outside which I can also dial and ring through to fine. For some reason thoug, a 7 digit or 10 digit dial string
I'm pretty sure ALERT_INFO became _ALERT_INFO, so change your line to read:
SetVar(_ALERT_INFO=bellcore-r4)
and it should work again.
Moj
Nikhil Yogesh Jogia wrote:
Hi All,
The setup I have is an asterisk box with a sipura spa 2000 for the fxs ports
and a TDM400P with 2 fxo ports.
Anyway,
On 12/19/05, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
I'm pretty sure ALERT_INFO became _ALERT_INFO, so change your line to read:
SetVar(_ALERT_INFO=bellcore-r4)
and it should work again.
Actually, having just checked then chan_sip.c source, ALERT_INFO
should be working. How
Asterisk -r makes no mention of any activity when this occurs
so it seems that Asterisk is not even generating the busy
signal. Is the Handytone capable of doing this and if so, why
would it be?
Make sure early dial is disabled in your HT486 config. I've never been
able to get it working
Yes call progress tones (busy) can be generated by ATA devices. Not
sure if the 486 has a digitmap or not, may wish to checkit out if so.
I also believe it has an internal status webpage and syslog info you
could look at for clues.
On Dec 19, 2005, at 5:34 PM, Craig Bruenderman wrote:
On 12/19/05, BJ Weschke [EMAIL PROTECTED] wrote:
On 12/19/05, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
I'm pretty sure ALERT_INFO became _ALERT_INFO, so change your line to read:
SetVar(_ALERT_INFO=bellcore-r4)
and it should work again.
Actually, having just checked then
Greetings all,
A couple of clients have recently decided they'd like extensions to their
office PBXs at their homes, so they've duly been provided with preconfigured
phones which register with the Asterisk server at their offices (public IPs)
quite happily.
Every 3-5 days it seems that these
Harry McGregor wrote:
Hi,
Has anyone used a Digium PRI card in an IBM eServer x346? I know that
Digium's website lists the x345 as having problems, but I am restricted
to buying only IBM eServers for this possible project.
I would like to use the TE411P
Harry i am using 3 deferent IBM
On Mon, Dec 19, 2005 at 03:18:18PM -0800, Kerry Garrison wrote:
I don't know exactly how it works, but since it appears to just be SIP, I
would have to assume a STUN setup. I haven't bothered to sit there and watch
the packets go by to see what its doing under the hood.
thanks - luigi
On 12/19/05, Jolly M. Recto [EMAIL PROTECTED] wrote:
Harry McGregor wrote:
Hi,
Has anyone used a Digium PRI card in an IBM eServer x346? I know that
Digium's website lists the x345 as having problems, but I am restricted
to buying only IBM eServers for this possible project.
I
Greetings,
We have just completed a successful test of DNS SRV failover with
Polycom phones, but feel that the failover was a little sluggish (up to
15 minutes for some phones).
We are using the default Polycom settings for Retry Interval and other
related settings and I'm wondering if
Harry,
We have a ibm x346 with a one port Digium E1 card in it.
Ilan
On 12/19/05, Harry McGregor [EMAIL PROTECTED] wrote:
Hi,Has anyone used a Digium PRI card in an IBM eServer x346?I know thatDigium's website lists the x345 as having problems, but I am restrictedto buying only IBM eServers
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