That website is not owned by the providers of this mailing list, as
far as I know. The WIKI is an independently-owned resource where
anyone can post anything (on topic) they like. There's no screening
except for the fact that anyone can also remove, change or comment
on anything they don't
I have recently posted a message regarding hung SIP channels when using the Monitor()
command. Well, I was wrong.
The channel hanging wasn't caused by the Monitor command. They also hang when they
aren't monitored. The cause seems to be our PSTN gateway provider. When for some
reason their
Hi guys,
I am new to Asterisk. It looks great, but I have a couple of open
questions such as
- What softphones are recommended and provide Call Center level
functionality?
- What hardware (especially E1 (Euro S2M) class) are supported?
I would like to search the list archives, but can't find a
Hi,
I'm not sure where you come from... but asking for help and ranting
about something is usually the wrong way around... and therefore I will
just point you to the wiki and go help yourself.
Too brittle? It won't work with final... look you probably just came
looking around the corner and
I have tested positively mbrola with lliaphon for the french language.
I'm using it within an agi script.
for lliaphon go :
http://www.culte.org/projets/biglux/install/lao/lliaphon.shtml
for other languages try :
http://tcts.fpms.ac.be/synthesis/mbrola/mbrtts.html
Regards
Eric
I have to disagree with a lot of your statements:
Leo Ann Boon wrote:
Hi all,
I'm having serious problem getting zaphfc to work on my box. I d/l'd
bri-stuff-0.1.0RC3/RC4a and followed the instructions to the dot.
Everything builds fine. But, when 'make load' the whole machine will
freeze.
Yes, it should, as you may define in zaptel and zapata which channels
are B channels, and which ones are D. You will just have to figure out
which channel the Legend is putting the D channel. It will probably be
either 9 or 24, but it is hard to say which, knowing Lucent.
On Wed, 1 Sep 2004
Hi,
I just setup two voicemail boxes - but after they record the wav
file, they don't seem to email it even though i have the correct format
in my voicemail.conf file. Aterisk doesn't give me any feedback
besides the recording of the voip file... and then stays there.
Secondly, I can't get
Oliver Breidenbach wrote:
Hi guys,
I am new to Asterisk. It looks great, but I have a couple of open
questions such as
- What softphones are recommended and provide Call Center level
functionality?
- What hardware (especially E1 (Euro S2M) class) are supported?
I would like to search the list
First thing to check is that the kernel sees the card. If you do lspci
you should see the zaphfc card assigned an IRQ. If you don't see this
then your card is not being detected.
If your card is there, then try replacing nz in your zaptel.conf file
with uk as I know that works and I think these
Hi,
I'm having a few issues with X-Lite from home. Setup is as follows:
Asterisk --- Work Router ---(internet)--- Home Router --- Home Pc with
X-Lite
The router at work has a port forward to 5060 on the asterisk ip. X-Lite
then connects to the server through this port, it can log in and
Hi, we have a CCM3.3 environment previously installed. Now
we have * up and running, and have been able to place calls from CCM-controlled
skinny-phones to Asterisk-controlled X-Ten-clients. This is done with a simple CCM-config
(add H.323-gw, add routepattern pointing to it). My question
Hello Eric,
thanks for the information
i have everything installed .. but can't seem to get the voice to get
into asterisk
i tried
System(/usr/local/play_ola /tmp/dino.ola)
as a test .. but it times out ..
any solutions ?
--
Best regards,
Dannymailto:[EMAIL
Hello,
Do most debian users make packages from the cvs HEAD?
I do a regular old compile for a 2.6 kernel and it
works fine. Is it bad form or dangerous? thanks.
JW
__
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Yahoo! Mail is new and improved - Check it out!
Hi,
I have been looking for info on * and the BT Easicom 1000 without much
luck when i found a post to this list from Andy Powell saying that he
had the phone working quite well. Before i go buy a shedload of these
things I would like to know what problems/sucesses people have had with
these
Andres wrote:
The quick and dirty way:
In rtp.c, function ast_rtp_write, in the switch statement,
AST_FORMAT_G729A case, change the smoother creation to something
larger. E.g.:
rtp-smoother = ast_smoother_new(40);
Keep in mind that you must set this into something
I use the Background command from the agi. The audio file
has to be in the proper directory, i.e. /var/lib/asterisk/sounds/fr
here's excerpt from a bash script
# phonetize
lliaphon $filetxt # creates the ola file
# pronounce
$mbrolaprog -e -f 1.0 -t 0.6 -I $mbrolainitfile
Has anyone out there used the PipeMedia/PipeCall PSTN gateway?
Anything good/bad to say about it?
I'm considering using them for a new customer. They seem to have good rates,
good provisioning tools and (better still) give commission on usage to
dealers.
--
David Gurr
Congruity Ltd.
Annoying :(
S.
-Original Message-
From: Mail Delivery System [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 02, 2004 3:06 AM
To: [EMAIL PROTECTED]
Subject: Mail delivery failed: returning message to sender
This message was created automatically by mail delivery software (HiveMail).
Hi
Another beginner's question:
Can I gain root if I have write access to asterisk's config files?
--
Tzafrir Cohen +---+
http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend|
mailto:[EMAIL PROTECTED] +---+
Hello list,
I found a webmin module at asterisk's ftp directory.
But i could't install it.! Does someone have a fix
for it ?
Regards,
-Jefferson Carvalho
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi all,
Maybe I'm being thick here, but I've had a look through the mailing list and
the Wiki, and I can't seem to see details of anybody else with this
problem
With the following line:
exten = s,1,HasNewVoicemail(201)
I am getting the following error:
-- Executing
Can anyone help me out here
When I make spandsp-0.0.1 I get the following error message(s) on Fedora
Core 2
Help or suggestions / fixes would be great
I have searched the threads and found and tried all sorts of
suggestions
---
In file included from /usr/include/tiffiop.h:45,
Can anyone suggest a good, easy way to implement fax support on
asterisk?
I have tried spandsp, still having problems getting it to compile...
Fax Manager, still appear to need spandsp
Same with hiyla
Any one got fax support working on * on FC2 ?
___
Hi,
I'm trying to dial out on a vonage line connected to a zap channel
using stuff like:
exten = 200,1,Dial(Zap/2/${EXTEN})
but it doesn't work - when i dial in the extension, i can see on a phone
connected to the same line that it's gone active - but there's no
dialtone. also tried adding
I use tips from Voip-info but doesn't work yet.
canreinvite=yes in sip.conf
rgds
Sato
Do you Yahoo!?
Win 1 of 4,000 free domain names from Yahoo! Enter now.___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi,
need a quick help ... it should be easy but ...
exten =_9898,1,Answer
exten =_9898,2,VoiceMailMain([EMAIL PROTECTED])
Accepting overlap call from '342' to '9' on channel 0/2, span 3
-- Executing Answer(Zap/8-1, ) in new stack
-- Executing VoiceMailMain(Zap/8-1, @domain) in new
A customer of mine has 3 TDM400P cards in a box running asterisk.
On each card he has four FXO modules.
I have set up the dialplan to dial via group 1 for an outgoing call.
Channels 1-12 are in group 1.
If he plugs a telephone cable into socket 2 or 3 etc, but not 1,
This is frequently caused by the evil exten = _.,1,Blah. Asterisk tries
to run several letter extensions at different times. exten = i is
run when you dial an invalid extension. exten = o is run when a
caller presses 0 while leaving a voicemail. _. is a common gremlin
usually attracted to
On Thu, 2 Sep 2004, Storm D. J. Petersen wrote:
Hi,
Thanks, I'll try to do a GSM Bridge call today. I understand your answer
for why voicemail - as in it does not require realtime processing, but what
about the echo back test? When I use echo back tests on other * servers or
FWD I
On Thu, 2004-09-02 at 02:21, Oliver Breidenbach wrote:
I would like to search the list archives, but can't find a searchable
archive.
There's a searchable archive field on the Digium mailing list page. I
don't like that one, it doesn't seem to work as well as I would expect.
To search the
[EMAIL PROTECTED] wrote:
Accepting overlap call from '342' to '9' on channel 0/2, span 3
-- Executing Answer(Zap/8-1, ) in new stack
-- Executing VoiceMailMain(Zap/8-1, @domain) in new stack
As you can see there variable CALLERID is empty, why ?
Sending a question again doesn't mean
I'm pretty sure someone must have done this before but I couldnt find any
trace of it on the web so I thought I would drop a note about how I ended up
doing it. I have also posted this info on voip-info.
Warning : This is not very elegant and I'm currently trying to write a patch
in order
On Thu, 02 Sep 2004 07:28:21 -0500, Eric Wieling [EMAIL PROTECTED] wrote:
On Thu, 2004-09-02 at 02:21, Oliver Breidenbach wrote:
I would like to search the list archives, but can't find a searchable
archive.
There's a searchable archive field on the Digium mailing list page. I
don't like
Hello Andreas,
Thursday, September 2, 2004, 2:28:33 PM, you wrote:
AS [EMAIL PROTECTED] wrote:
Accepting overlap call from '342' to '9' on channel 0/2, span 3
-- Executing Answer(Zap/8-1, ) in new stack
-- Executing VoiceMailMain(Zap/8-1, @domain) in new stack
As you can see there
Good day all
Is there anyone who has experience with ISDN BRIDDI?
I want to know if asterisk can distinguish between the different numbers?
I want each number to play a different intro/answering message?
Please Help
Thanks
Altus
___
Asterisk-Users
FYI. Reading is free; if you don't have an account it is trivial to
sign up, and they're very politically correct, as might be imagined,
about using email for selling purposes.
http://www.nytimes.com/2004/09/02/technology/02caller.html?hp
___
On Thu, 2 Sep 2004 14:51:10 +0200, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
Is there anyone who has experience with ISDN BRIDDI?
I'm currently using a BRI with ISDN4Linux.
I want to know if asterisk can distinguish between the different numbers?
Yes, it can differentiate between
Hi Everyone
The prepaid billing system is properly installed in the asterisk box.
We are trying to connect from one SJPhone to another SJPhone using asterisk
server.
The call procedure are as follows:
1. From one SJPhone the user dials sip:destination user id@Asterisk IP
address.
2. Then sip
Hi all, I'm having sudden MWI problems. Everything else on the phone
works fine though.
I have three Cisco 7940s.
Asterisk server is behind a firewall running NAT. (192.168.1.202/24)
Phone #1 - On the same subnet 192.168.1.250. Everything works great.
Phone #2 - On a different subnet,
I only did this with ports and the using context how do I do the number
I want each number for a different company?
On Thursday 02 September 2004 15:10, Darryl Ross wrote:
On Thu, 2 Sep 2004 14:51:10 +0200, Altus Snyman [EMAIL PROTECTED]
wrote:
Good day all
Is there anyone who has
Check out the following url:
http://www.mail-archive.com/index.php?hunt=asterisk
Darren Wiebe
[EMAIL PROTECTED]
el Flynn wrote:
Oliver Breidenbach wrote:
Hi guys,
I am new to Asterisk. It looks great, but I have a couple of open
questions such as
- What softphones are recommended and provide
hi,
i'm under the impression that this feature is not available in asterisk,
consider this scenario:
- you are the operator. you answer a call from outside and you want to
transfer it to one of the extensions. after you transfer, if the person
you transferred the call to, doesn't pick up or if
On Thu, 2004-09-02 at 05:05 -0700, Alberto Sato wrote:
I use tips from Voip-info but doesn't work yet.
canreinvite=yes in sip.conf
and no t or T in the Dial command.
--
Dave Cotton [EMAIL PROTECTED]
___
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[EMAIL
Brian Capouch wrote:
FYI. Reading is free; if you don't have an account it is trivial to
sign up, and they're very politically correct, as might be imagined,
about using email for selling purposes.
http://www.nytimes.com/2004/09/02/technology/02caller.html?hp
Bugmenot.com:
Login details for
star38.com .25 connection .07-.13 per min What a bargin
On Thu, 02 Sep 2004 16:16:26 +0200, Stefan de Konink [EMAIL PROTECTED] wrote:
Brian Capouch wrote:
FYI. Reading is free; if you don't have an account it is trivial to
sign up, and they're very politically correct, as might be imagined,
Hi All,
Has anyone had experiences with Asterisk
and ISDN Gateways or cards? I am looking to setup a home phone system using
asterisk and am looking to convert my two existing no's to a single isdn2e
line.
I was wandering if anyone could recommend
which is better to use with asterisk - ISDN
I know someone who was looking into it but they decided not to make
the investment at this time versus other options they had available.
Prices did look decent though.
On Thu, 2 Sep 2004 11:10:37 +0100, David Gurr
[EMAIL PROTECTED] wrote:
Has anyone out there used the PipeMedia/PipeCall PSTN
On Thu, 2 Sep 2004 [EMAIL PROTECTED] wrote:
Has anyone had experiences with Asterisk and ISDN Gateways or cards? I am
looking to setup a home phone system using asterisk and am looking to
convert my two existing no's to a single isdn2e line.
I was wandering if anyone could recommend
i have a traditional pbx attached to one line of NT1, and asterisk with hfc-s
to the other one;
when a call comes in, it is like asterisk captures it, passing it to the
channel configured on dialplan; in the facts, the call is not answered, but
NT1 doesn't ring pbx, as it would do if call had
I don't have the complete list at the ready, but *8 on a Zap channel
is (non-directed) call pickup.
I spent a lot of time searching for the call waiting problem months
ago, and I guess I can only blame myself if it's still not
well-documented on the Wiki. My experience is almost exclusively with
Thanks gentlemen for all your help and suggestions;
For some reason, this setup of ours started to work perfectly again this
morning and is still up without any of the symptoms and failures I
described. We'll have to figure this out with our telco rep - I am certain
this problem has not gone
On Thursday 02 September 2004 04:56, Michael Workman wrote:
On your web you have a link
http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard
To Setup Calling with Diamondcard.us and I signed up and paid the money
according to Stephen Karrington it was all automated... And it was
Has anyone here ever used or tried the ROBO-8712VLA single-board computer
with Asterisk (and therefore Linux)?
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
___
You don't read do you? Simple reading of the message shows that the
email account sysad is over quota. A little more disection of the
message shows it is [EMAIL PROTECTED] that is over quota. What is worse
is that the admin of the HiveMail software at bembang.com is so
incompetent as to either use
The proper thing for you to do is find or spam a real admin account at
bembang.com to fix their broken software and admonish the user for not
checking and cleaning mail more often.
They don't care... I've sent several messages to the postmaster and they go
unanswered... Obviously a bunch of
Hey there
I'm trying to get asterisk to leave messages on answering machines
So i have a pretty cool php notifying script (it notifys, it doesn't
spam!!) to phones and cellphones
Now all is fine if a human picks up, but if an answering machine picks
up, well the script plays, but only the
Answering machine detection is usually accomplished by analysing the timing
of the voice energy in the initial answer period. People usually answer by
saying: Hello, Frank Giwerski, Pencil sharpening department, or
something fairly short, whereas answering messages are usually longer.
So, I
Hey thanks, thats a great idea too
Basicly just check for a pause, if i don't get one quickly, then its an
answering machine
And both ideas are compatible, so i could do both at the same time
Chears
Scott Stingel wrote:
Answering machine detection is usually accomplished by analysing the timing
After having a very good experiance with the UIP200 phones we decided
to purchase a couple of UIP300 (2 line version) Only to releise that
these phones are H.323 not sip. What is the difference between h.323
and sip. What needs to be done to get this phone working with
asterisk?
Thanks,
Matt
Hi All,
A curious thing started to happen recently. My Asterisk server uses
IAX2 to several ITSPs for outgoing calls, but I still have two POTS
lines for incomming calls. One of those lines has the DSL circuit from
my ISP. When that analog line rings it forces * to drop any IP based
calls that
Hi,
after following up on my previous email about zaptel x100p having
trouble accessing a vonage dial tone, I think the problem is with
feedback and noise on the line - any remedies for this?
Thanks,
Imran
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Asterisk-Users mailing list
[EMAIL
We are having some problems with the audio during a call
dropping out for a second or so and then returning. This is not at the beginning
of the call and seems to happen randomly. We use primarily Cisco 7960s
and our * is a very current version of CVS head. The server itself is a Compaq
I would like to know where could I change the directory that the web
interface looks to when trying to play wave file. The current
location is as follows: /var/spool/asterisk/vm/1234/INBOX/msg.wav
When I use addmailbox command it places it under
/var/spool/asterisk/voicemail/default/1234/.
After having quite a bit of success with the UIP200 we decided to pick up a couple of UIP300 phones. After purchasing them we found that they were H.323 Ip phones. I searched on voip-info.org and have been unable to determine how exactly to setup this phone in asterisk. Any help would be
Sorry to be off topic of your post but did you have the problems
documented on the link below with the UIP200 ?
http://www.voip-info.org/wiki-UIP200
Thank you,
Steve Maroney
On Thu, 2 Sep 2004, Matt Hohman wrote:
After having quite a bit of success with the UIP200 we decided to pick
up a
Looks to me like you are telling Asterisk to outpulse 200 on the
vonage line.
If I remember my vonage service correctly, everything (except 911) was
an 11-digit call.
Perhaps it'd look better as:
exten = _1NXXNXX,1,Dial(Zap/2/${EXTEN})
-Original Message-
From: [EMAIL
If you have a single line phone, or better yet, a butt-set, hook it up
to the TELCO side of the 66 block, and remove the bridge clips. This
effectively isolates the customer equipment from the telco circuit.
Then, using that butt set (or phone), go off hook and see if the static
is still there.
On Mon, 30 Aug 2004 17:15:42 -0400, Michael Graves [EMAIL PROTECTED] wrote:
I'd really like to see a show of hands with regard to people's
experience with FXO interfaces. I own a few X100p cards and have had
nothing but problems with them.
2 X100 FXO here and no problems with them here in
You can also search the archives here:
http://www.mail-archive.com/asterisk-users%40lists.digium.com/
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, September 02, 2004 8:28 AM
Just a hint, a friend workin with me in an Asterisk project traveled
recently to Mendoza (Argentina) and called to a bank using a big and
expensive PBX that anwered:
Meridian mail ... mailbox?
sounds very much like the
Comidian mail ... mailbox?
in Asterisk. Just a coincidence??
Bye
Luis
Chris
I have an FXO card in a channel bank, which is run into a Digium TE405P.
Works great.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Wilson Pickett
Sent: Thursday, September 02, 2004 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial
Kevin Walsh wrote:
Andreas Sikkema [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
There are a few codecs, other than G.729, that you may not have
heard of. These include GSM, iLBC and SpeeX, to name a few.
Paying for G.729 licenses, however cheap they may appear, only
encourages the
William Suffill [EMAIL PROTECTED] lazily top-posted:
star38.com .25 connection .07-.13 per min What a bargin
Was there a point to that, or was that just spam? You didn't quote any
context for your statement, so I have no idea what you are referring to
or answering.
--
_/ _/ _/_/_/_/
On Thu, 2004-09-02 at 20:54 +0100, Kevin Walsh wrote:
William Suffill [EMAIL PROTECTED] lazily top-posted:
star38.com .25 connection .07-.13 per min What a bargin
Was there a point to that, or was that just spam? You didn't quote any
context for your statement, so I have no idea what you
Yes and no... We had to use the nat=never, We had soem issues with the phone freezing on call waiting this was fixed by the newest firmware 4.59a. But overall a good phone, we just need two lines.
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
On 2 Sep 2004 at 6:32, Rich Adamson wrote:
A customer of mine has 3 TDM400P cards in a box running
asterisk. On each card he has four FXO modules.
I have set up the dialplan to dial via group 1 for an outgoing
call.
Channels 1-12 are in group 1.
If he
Title: Message
I have just spent
the morning playing around with a Polycom IP600's microbrowser. Everything
is working pretty well. In answer to the question of what type of XML it
runs, it appears to be more or less XHTML-compliant. I have created a
basic set of web pages allowing users to
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Greg Blakely
Sent: Thursday, September 02, 2004 2:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Hard Ground (On Ring)
If you have a
Hi,
Is there any phone numbers with answering machine wich can record my
voice and play it back to me?
It would be very helpful for asterisk testing, but im not shure such
service exsists at all.
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
___
A customer of mine has 3 TDM400P cards in a box running
asterisk. On each card he has four FXO modules.
I have set up the dialplan to dial via group 1 for an outgoing
call.
Channels 1-12 are in group 1.
If he plugs a telephone cable into socket 2
Joshua M. Thompson wrote:
Looks like an Asterisk box and a simple CGI script to me.
Is this possible out there without a SS7 gateway? Or do you need just a
friendly channel supplyer that allows you to send any callerids thru
their switches?
Stefan de Konink
Is there any phone numbers with answering machine wich can record my
voice and play it back to me?
It would be very helpful for asterisk testing, but im not shure such
service exsists at all.
Do you mean somthing like:
; record a temporary GSM file
exten = 3920,1,Wait(1)
exten =
Send me a copy of your solution! This would be a handy tool to add
external VM to an existing PBX.
Thanks!
Greg Blakely wrote:
Thank you. That will probably get me to where I need to go.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dunc
Sent:
Yes, something like this.
I would like ability to call such system via PSTN to test my * setup and
my ITSP termination
Something like this:
My* -- MyITSP -- PSTN -- System with extensions you tell about
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL
Has anyone written an AGI script with the AGI_BACKGROUND that will
allow on-demand recording of conference calls?
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To UNSUBSCRIBE or update
On Thu, 2004-09-02 at 22:44 +0200, Stefan de Konink wrote:
Joshua M. Thompson wrote:
Looks like an Asterisk box and a simple CGI script to me.
Is this possible out there without a SS7 gateway? Or do you need just a
friendly channel supplyer that allows you to send any callerids thru
their
Hello,
We recently installed a Wildcard TE410P, but we are having problems to make
it work reliably with a German E1 (Primaermultiplexanschluss PMX DSS1). Our
carrier (Hansenet in Hamburg, Germany) is using Nokia/Lucent switches.
The card is only able to set up the first B-channel (although it
I've tested this change already, and it's actually more broke than
fixed. I have a T100P card connected to a TA750, with 20/4 FXS/FXO.
When dialing TO one of these ports, it always looks busy to
asterisk... I've backed off to version 1.330 of chan_zap.c and it
works fine...
I'll try to post
You can do that easily --
exten = 250,1,DigitTimeout(10)
exten = 250,2,ResponseTimeout(20)
exten = 250,3,Answer
exten = 250,4,Read(callto,pls-entr-num-uwish2-call,10)
exten = 250,5,Read(callfrom,enter-phone-number10,10)
exten = 250,6,NoOp(${callto})
exten = 250,7,NoOp(${callfrom})
exten =
I've just been doing some tests using the manager API to originate an
outgoing call via a X100P and connect the call to an extension:
Action: Originate
Channel: Zap/1/01234567890
Context: local-extensions
Exten: 6000
Priority: 1
I've noticed that extension is getting called as soon as the
You cannot take the primary clock from all 4 channels. So you must
change your lines in zaptel.conf to:
span=1,1,0,ccs,hdb3,crc4
span=2,2,0,ccs,hdb3,crc4
span=3,3,0,ccs,hdb3,crc4
span=4,4,0,ccs,hdb3,crc4
also why specify the groups double ?
switchtype = euroisdn
signalling = pri_cpe
group = 1
Hi Tony -
I am needing the same thing. In short, the answer is no. the only way
you can reliably detect answer supervision is :
1) Line reversal from the telco or PBX. I think BT will provide
reversal on answer if you request it as some payphones need it.
2) Meter pulse. This starts from
On 2 Sep 2004 at 15:34, Rich Adamson wrote:
A customer of mine has 3 TDM400P cards in a box running
asterisk. On each card he has four FXO modules.
I have set up the dialplan to dial via group 1 for an
outgoing call.
Channels 1-12 are in group 1.
I may be way off here, but don't you need to change that third line to:
make clean;make;make install
And I am wondering why such an old CVS(07/05/04)?
Lyle
- Original Message -
From: Norman Tomlnis [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL
Thanks, I made the changes, but that did not help?!
This is what Asterisk is telling me when I try to dial out. Incoming calls
are not signaled at all. When our carrier resets the E1, the first channel
sometimes works for a short time:
*CLI -- B-channel 0/1 successfully restarted on span 1
Hmm I have seen that behaviour before. Are you sure hansenet uses crc4 ?
On Fri, 2004-09-03 at 00:00, Henrik Pfluger wrote:
Thanks, I made the changes, but that did not help?!
This is what Asterisk is telling me when I try to dial out. Incoming calls
are not signaled at all. When our carrier
Yeah, I posted a new bug. My configuration isn't that wierd, I didn't
think. Asterisk - T100P - TA750... I'd still like to see the
'battery detection' working for my situation. I'm trying to get his
attention on IRC at the moment, as he suggested on the bug tracker...
Rob
On Fri, 03 Sep 2004
Yes, they told me so. When I turn it off I only get endless messages:
ep 3 02:28:37 WARNING[98310]: chan_zap.c:1880 pri_find_dchan: No D-channels
available! Using Primary on channel anyway 16!
== Primary D-Channel on span 1 up
Sep 3 02:28:38 WARNING[98310]: chan_zap.c:1880 pri_find_dchan: No
On 1 Sep 2004 at 23:57, [EMAIL PROTECTED] wrote:
Is it possible to allow distinctive rings work for FXS ports as well?
I need a certain FXS extension to ring a distinctive double ring.
I modified zapata.conf appropriately for dring1,dring2 and it just
Seems to ignore my updates.
Do
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