Asterisk doesn't use PING to check the STATUS, it uses a SIP OPTION message.
Regards,
Marc
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: mardi 17 février 2009 14:06
To: Asterisk Users Mailing List - Non-Commercial
The Attrafax software that was mentioned at the beginning of the thread does
support Gateway mode.
Regards,
Marc
-Original Message-
Fabio Mosti wrote:
2009/2/16 Steve Underwood ste...@coppice.org:
You don't indicate the kind of setup you are using.
I use asterisk
Hello Douglas,
I don't know if this is exactly what you need, but the fromdomain and
fromuser in sip.conf (explained here:
http://www.voip-info.org/wiki-Asterisk+config+sip.conf) change the From:
header to [EMAIL PROTECTED]
Regards,
Marc
Douglas Garstang wrote:
Does anyone know how to
Hello,
I saw that the error:
SIP response 484 Address Incomplete
is converted into
DIALSTATUS = NOANSWER
HANGUPCAUSE = 16 (NORMAL_CLEARING)
shouldn't it be something like
HANGUPCAUSE = 1 (UNALLOCATED)
HANGUPCAUSE = 28 (INVALID_NUMBER_FORMAT)
or another cause, other than NORMAL ???
with Server C.
Best regards,
Marc Storck.
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To bad that prefixes like +220 (Gambia), +230 (Mauritius), +240
(Equatorial Guinea), +250 (Rwanda), +260 (Zambia), +290 (Saint Helena),
+350 (Gibraltar), +370 (Lithuania), +380 (Ukraine), +420 (Czech
Republic), +500 (Falkland Island), +590 (Guadeloupe), +670 (Timor
Leste), +680 (Palau), +690
Yes there has just been a new release of Asterix (the Gaul has x at the
end) .
JP Carballo wrote:
Obelix wrote:
Is there a .gsm file for announcing UK pounds and pence after the credit
remaining prompt, besides the dollar and cents file?
/Obelix
this may work on Grandstream phones... set the ring tone to number 3
which is empty, so no tone and set the ring tone number 1 or 2 to ring
on CallerID matching (e.g. everything starting with 678 will use ring
tone 2) ... I never tested it, but the configuration shows the fields,
so it may
I would be interested as well...
Why not post them somewhere?
Regards,
Marc
[EMAIL PROTECTED] wrote:
I'm game for using them /and testing them.
Ben..
Roman:
I created two bash scripts called Mail2Fax and Fax2Mail for use with the
asterisk sever.
They leverage the app_txfax and
Hello,
I tried to add the following SIP friend to SIP Realtime:
[sip-friend23]
type=friend
host=12.13.14.15
context=acme
disallow=all
allow=ulaw
allow=alaw
accountcode=sip-friend23
But only calls to that SIP friend work, calls from that friend are
instantly matched to the default context set
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MS Networks SA [EMAIL PROTECTED]
IT Service Providerhttp://www.msnetworks.lu
15, route d'Esch Phone: +352 2727 3030
L-4450
Marc Storck
MS Networks SA [EMAIL PROTECTED]
IT Service Providerhttp://www.msnetworks.lu
15, route d'Esch Phone: +352 2727 3030
L-4450 Belvaux Fax: +352 2727 3060
--- MS Networks powered service ---
http
The T38 service offered by this person has nothing to do with Asterisk,
they want you to use their own system, and pay them for service. As far
as I did understand, you need do install a custom firmware onto your ATA
(only a limited number of ATAs are supported).
Personally I don't think that
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Thanks Michael,
do they have an online ordering system, they don't seem to have a real
website
Regards,
Marc
Michael Graves wrote:
On Sun, 24 Jul 2005 21:20:05 +0200, Marc Storck wrote:
Hello,
I'm looking for US DID and US50/CA 800# Providers.
I found voiceconduits.com 8 month ago
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Hello,
I'm looking for US DID and US50/CA 800# Providers.
I found voiceconduits.com 8 month ago, there interface looks good, but
there are still not live, I believe they won't be any time soon.
I found sixtel, but order take eternities, they probably won't get my
orders right any soon.
So
and
experiences.. providing clues allows them to expand their experience and
build their confidence... It requires them to look at the details and
learn
to analyse them.
Regards,
Derek
- Original Message -
From: Marc Storck [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
,Dial(${ARG1},30)
Then I would fallback to voicemail (or something else) after the 30
seconds?
Angus
- Original Message - From: Marc Storck
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, July 24, 2005 10:06
.
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Hello,
I use the ${BLINDTARNSFER} variable for transfers from SIP accounts, but
this variable seems to be unavailable for IAX channels. Is this supposed
to be this way, is there another variable???
Many thanks for your help,
Marc
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proprietary protocol?
In some places IAX is refereed as open protocol.
How can proprietary protocol be open protocol?
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any
valid reason :-)
http://www.voip-info.org/tiki-index.php?page=Asterisk%20protocols
I think that should be corrected!
Documentation is here:
http://www.cornfed.com/iax.pdf
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Hello,
does someone offer DIDs from the areas of shanghai and/or bangalore.
Many thanks,
Marc
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I'm also looking for numbers from
HongKong,
Taiwan,
Japan and
Singapore
So if someone has some DIDs from this areas, I'm very interested to get
one or another from those DIDs.
Best Regards,
Marc
Marc Storck wrote:
Hello,
does someone offer DIDs from the areas of shanghai and/or bangalore.
Many
PRI E1. On incoming calls
the CALLINGTON variable is empty. I have the latest stable version of
asterisk. Do I have to use another variable or is the TON only support
in CVS?
Marc
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After applying the change to the init script, it seems to restart the
asterisk processes which get killed, but do you have a functional system
with this?
Our Testsystem spits out some '100% CPU-Loaded mpg123 processes' and
asterisk was somehow dead.
Did I miss something?
Yes I found the same
Mark, that's what the command pri debug span 1 does, produce a lot of
output so you can see what is received and what is sent. Maybe you can
paste the output to pastebin.ca and tell us the link.
Regards,
Marc
Mark Phillips wrote:
Nothing happens. I get the same (non)error.
I get plenty of
I use DIDs for incoming faxes as well, but we have several users with
combined Telephone-Fax-Hardware. As humans make errors and are very
lazy, these users don't want to dial another prefix when they send a fax.
This is what I try to do:
exten = _XX.,1,SetVar(NUMBER=${EXTEN}) ;save the number
Hello,
does the Junghans QuadBRI Card and qozap module support Fax detection?
I want to use fax detection using the Answer() command and the 'fax'
extension. I used the example from the wiki, but I had no success so
far. Can someone please share his/her experiences/knowledge??
Many thanks,
Marc
Marc Storck
MS Networks SA [EMAIL PROTECTED]
IT Service Providerhttp://www.msnetworks.lu
15, route d'Esch Phone: +352 2727 3030
L-4450 Belvaux Fax: +352 2727 3060
--- MS Networks powered service ---
http
servers. Can someone help me???
Regards,
Marc
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Marc Storck
MS Networks SA [EMAIL PROTECTED]
IT Service Providerhttp://www.msnetworks.lu
15, route d'Esch Phone: +352 2727 3030
L-4450 Belvaux Fax: +352 2727 3060
--- MS Networks powered service ---
http
Sorry for replying into the wrong thread.
Regards,
Marc
Marc Storck wrote:
According to
http://www.itu.int/ITU-T/inr/forms/files/Applications-E-164.pdf Page 3
the 882 99 has been assigned to Telenor (http://www.telenor.com). So
e164.org may have a problem with that prefix, if the 882 99 is ever
for nearly three weeks now. Their voice mail boxes are full and
writing email to them does not get any returns. Thoughts or sightings are
appreciated.
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that number?
If this would have to be implemented I'm willing to fund a bounty!
Regards,
Marc
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L
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). But then the
debug output stops in the middle of 1 debug packet, to continue over 20
seconds later (if it continues).
The actual CPU load is load average: 0.00, 0.00, 0.00.
I cannot find the problem, maybe someone over can help me!
Thanks,
Marc
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), the same is
true if the users uses the redial button.
So my question is, how can make asterisk or the ZAP channel wait a
little bit longer before he claims a match against a number in the
dialplan...
Thanks,
Marc
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message to finish, and wait for tone
exten = 666,3,Dial(my-fax-number) ;after about 10sec.
exten = 666,4,Dial(1) ;to confirm selection
exten = 666,5,Hangup
exten = 666,6,Goto(s,1)
Any improvements are welcome.
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this works great for me, i use callerid= like this:
callerid=Marc Storck 35227273033
Matthew Boehm wrote:
OK. Here is the caveat I've found. The phones, in sip.conf, all have a
callerid= line because if they don't when they call someone the caller id
shows up ONLY as their extension.
For instance
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Marc Storck
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Hello,
thanks for the answers!!! You mentionned to use the switch command. I
read about it in the WIKI, but I couldn't find enought information to
understand what it is actually doing. Can someone point me to the right
direction?
Marc
Steven Critchfield wrote:
On Sat, 2004-09-11 at 21:41, Marc
is there any in-depth information available about the switch command???
Marc
Steven Critchfield wrote:
On Sun, 2004-09-12 at 06:42, Marc Storck wrote:
Hello,
thanks for the answers!!! You mentionned to use the switch command. I
read about it in the WIKI, but I couldn't find enought information
Hello,
I want to link several * boxes together. Some of them are dedicated as
user servers (SIP and IAX clients connect to them) and some are used
as PRI servers (where the PRIs are hooked onto).
I think TDMoE is the only channel type where you can group different
Interfaces into a single
did you try to add
canreinvite=yes
to
[net2phone3]
??
Marc
[EMAIL PROTECTED] wrote:
Hi!
Net2Phone is getting a common SIP status code, 404 Not Found, when
trying to place a call to our Asterisk server. We're hoping someone on
the list can shed some light on why this is happening. We can
You can re-register the codecs one time using other NICS. after that
one time you need to contact Digium to be able to re-register, but the
process is very easy!
At 21:33 18.07.2004, you wrote:
Anton Tinchev wrote:
The readme says that the license uses all network cards MACS
What happens
2 different things,
you should be able to join a channel even if nickserv didn't authenticate
you yet!!!
but this is OT ;-)))
Marc
At 20:12 19.06.2004, you wrote:
Steve Underwood wrote:
Hi,
I figured it out. Most IRC channels requiring some authentication give a
minute's latitude to allow for
configure your asterisk to use e164.org and make use of EnumLookup
then try to call +352 818 595, if your call goes to [EMAIL PROTECTED]
then you can call me for free over the net!
Marc
At 03:33 23.05.2004, you wrote:
Dean Collins wrote:
Tony, as per you inference that e164 are up to
asterisk only supports IAX2, SIP and TEL, it will only use IAX2 and SIP
entries however
so it is used to route via the Net if it cannot find a route via the
Net or the link isn't working it will go to the next priority in your
dialplan and do whatever you want, it doesn't re-configure
try to ask in english you may get an answer a whole lot faster
Regards,
Marc
- Original Message -
From: Administrator [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 06, 2004 6:11 PM
Subject: [Asterisk-Users] Fehler beim starten...
Hallo,
nachdem mir bis jetzt noch
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