Am Sonntag, den 30.03.2008, 16:56 +0800 schrieb mark morreny:
Dear friends,
I am wondering if there is any efficient way of extract the country
code, area code, and local code into 3 different variables from one
DNID that can look like 001630233-4333 or 0086213345333?
International code
Am Sonntag, den 30.03.2008, 09:54 -0400 schrieb Mike Trest - Personal:
Ayman,
One solution is to write an AGI scrip to parse the number and read
back in Arabic semantic order. for the last two digits and for
certain special numbers like 11 , 100 , 1000, ... .I must bring
out my old
this and now I cannot reproduce the problem. Gotta love * :)
Thanks, guys!
--
martin | http://madduck.net/ | http://two.sentenc.es/
a gourmet concerned about calories
is like a punter eyeing the clock.
spamtraps: [EMAIL PROTECTED]
digital_signature_gpg.asc
Description: Digital signature (see http
by asterisk (which
they support), but asterisk is going a bit lala here, isn't it?
First of all, why does it even bother with 3 and 111, given how
I disallowed them? And second, why does it *dare* to announce more
than what is available to the peer to which it relays?
--
martin | http://madduck.net
that will run on amd64, or which come with source
code?
Thanks,
--
martin | http://madduck.net/ | http://two.sentenc.es/
i wish there was a knob on the tv to turn up the intelligence.
there's a knob called 'brightness', but it doesn't seem to work
been compiled for 64bit. Sure, I can
run an entire 32bit system on 64bit hardware thanks to backward
compatibility, but I actually run a 64bit machine with native 64bit
code.
And no, this is not for performance reasons and there wouldn't be
any benefits. I just can't run 32bit software.
--
martin
Just a test, please discard
Looks like something is eating my messages on their way :-(
Martin
--
http://mujblog.atlas.cz/___
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asterisk
/asterik/zapata.conf
signalling=fxs_ks
context=your_incoming_context
channel=1
channel=2
channel=3
signalling=fxo_ks
context=your_dial_context
channel=4
Martin
- Original Message -
From: John Novack [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk
Yes, this codec is UNLICENSED and might be incompatible with legal versions.
Use it for testing purpose only might (or not) be legal, you must obtain
appropriate licenses if you plan any 'real' use of it.
I recommend to use gsm codec instead :-)
Martin
- Original Message -
From: Eric
configuration of cidstart= cidsignalling= etc...
This was recently discussed here, try to search archive
Martin
And configure your mail client to use plain text :-)
- Original Message -
From: Mark Quitoriano
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: 23. brezna 2008
Download an appropriate binary from
http://asterisk.hosting.lv/
and just drop into /usr/lib/asterisk/modules/
add allow=g723 to your sip.conf as necessary and restart asterisk...
Im only not sure how legal is this, you will probably need to obtain
licenses for all concurent channels...
Martin
Am Donnerstag, den 20.03.2008, 08:59 -0600 schrieb Aaron Fransen:
Holy Mackeral. Ignore that last message. I still do NOT know how to
route calls with the same extension being used in two locations,
however the issue I've resolved is getting Cisco CallManager and
Asterisk talking together
Am Donnerstag, den 20.03.2008, 16:59 +0200 schrieb Tzafrir Cohen:
And what happens if at the time of the shutdown there was a
___
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ROTFL
Trafrir, you made my day.
(BTW: I
Am Montag, den 17.03.2008, 15:08 +0800 schrieb Pete Kay:
Hi,
I am new to Asterisk and I am having a setup problem that I am trying
to resolved for the last couple days without any success. I am pretty
much desperated on this issue and I don't know why. Can someone
please kindly help me to
Am Montag, den 17.03.2008, 21:38 +0800 schrieb Pete Kay:
Hi,
Here is the SIP debug output for the playback test. Thank you so much
for your help.
Hi Pete,
[Mar 18 05:33:08] -- Executing [EMAIL PROTECTED]:1]
Answer(SIP/2000-081e0738, ) in new stack
[Mar 18 05:33:08]
Am Montag, den 17.03.2008, 13:59 + schrieb Alan Williamson:
Afternoon one and all.
I am having some interesting fun with our Asterisk setup.
We have two CISCO handsets (7960) sitting on the same network (NAT).
Each phone can successfully originate calls.
Each phone can be called
My asterisk crashes after fax is received using app_rxfax. But its not svn
version, its binary from Debian/Lenny ...
Martin
- Original Message -
From: Matt Riddell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: 11
fine, it crashes after transfer is complete
Martin
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
Am Montag, den 10.03.2008, 02:59 -0500 schrieb John Faubion:
But, just to clarify, please remember that using music as MoH
is considered a public performance, and if the pieces in
question do not include a buyout license *for the performance
Ok now I am curious, if a radio is playing
to create
channel of type 'sip' (CAUSE 3 - no route to destination)
This phone is known to give up if registration is not succesful after a certain
period, I believe its a bug in firmware but manufacturer don't...
Martin
- Original Message -
From: Jim Meehan
To: asterisk-users
I've tried this branch, but no luck again Dialing out work just fine,
but no incoming calls are recognized. With absolutely no message in logs
:-( I also have seen this mentioned here:
http://bugs.digium.com/view.php?id=12099
Martin
- Original Message -
From: sean darcy [EMAIL
You should stop asterisk first, otherwise modules are still in use
And then
/etc/init.d/zaptel stop
will remove modules.
Martin
- Original Message -
From: Paul Hales [EMAIL PROTECTED]
Sent: 28. února 2008 19:35
Subject: Re: [asterisk-users] Problems with removing zaptel
/etc
, ) in new stack
== Spawn extension (Martin, *903, 3) exited non-zero on 'SIP/210-081e9968'
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/210-081e9968, ) in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(SIP/210-081e9968, Zap/3) in new stack
-- Called 3
[2008-03-02 09:44:17] WARNING[5055]: chan_zap.c
(2.6.24 in my
case) ... Sometimes its hard to find working combination of kernel and
zaptel :-)
I will try the fix mentioned below first
Martin
- Original Message -
From: sean darcy [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: 2. brezna 2008 17:29
Subject: Re: [asterisk
A simple inrease of DEFAULT_BATT_DEBOUNCE to 32 (originally 4) work for
me... Probably a nasty hack but allows me to dial
(I also need DEFAULT_RING_DEBOUNCE increased to 256 to minimize problems
with caller ID, which is right above)
Martin
My incoming calls work just fine with 1.4.9.2
Am Mittwoch, den 27.02.2008, 16:30 +0100 schrieb Hans-Peter Straub:
Hello all,
i today have searched on the internet about a solution to let asterisk act as
a SMSC (Short MessageServiceCentre) to deliver SMSes directly to GSM Phones.
I only have found some cases with use of an extern SMSC
Am Donnerstag, den 07.02.2008, 00:18 + schrieb Ed W:
Can someone please explain how to match a + character in a dial plan (so
that I can swap it for the 00 country escape code).
In Europe at least the + is a common shortcut for the international
prefix (which is 00 in my country).
Am Freitag, den 18.01.2008, 04:21 -0800 schrieb bilal ghayyad:
Hi;
Via OpenVPN or port forwarding is known for me, but
via SSH is new for me, how I can do it and what is the
difference by SSH and OpenVPN?
In principle both use a packet stream (SSH is TCP, OpenVPN is TCP or
UDP) for
Am Donnerstag, den 10.01.2008, 12:31 +0100 schrieb Michiel van Baak:
On 11:22, Thu 10 Jan 08, Olivier wrote:
Hi,
Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support
roaming and handover and are these functions transparent for handset (then,
these functions are
Am Sonntag, den 06.01.2008, 21:05 -0600 schrieb Alejandro Kauffmann:
As long as this is an official rant thread
Good to know no new phones have hit the market since the last time this
question was asked and answered. It's also good to know
opinions about specific products don't change
Am Samstag, den 05.01.2008, 11:58 +0530 schrieb ram:
Hi
I understand what you are saying.
so once we see he is not input the pin more than 2times
he will be blocked for hour ( i will run cron job, after one hour
release them)
is this a good idea.
Hi Ram,
I do not think that
Am Samstag, den 05.01.2008, 06:40 -0400 schrieb William Herrera:
Hello to you all. Just got my first iP0020 phone and no matter what I
do to it when I try to call I get a busy signal even though Asterisk
and the phone web gui shows that the phone is “registered”.
Has any body had any similar
Am Samstag, den 05.01.2008, 13:31 +0200 schrieb Tzafrir Cohen:
On Sat, Jan 05, 2008 at 11:54:41AM +0100, Anselm Martin Hoffmeister wrote:
Using cronjobs is possibly a bad idea because you create load spikes, if
e.g. 5000 asterisk -rx commands are issued within a few seconds.
Why would
Am Samstag, den 05.01.2008, 19:50 + schrieb Gres +:
please can anyone help me knowing if i can install Linux and Asterisk
on HP servers
Gres,
you will have to find out if _YOU_ can do that.
Generally speaking it is very well possible.
For a quick start, you might want to try an
Hey folks,
Is anyone working on Asterisk (or other) presentation proposals for
OSCON 2008 in Portland, OR? Here's the link, in case:
http://en.oreilly.com/oscon2008/public/cfp/13
I'd love to see more Asterisk content there!
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic
Dear Rickygm,
Am Donnerstag, den 03.01.2008, 20:19 -0600 schrieb troxlinux:
hello list, happy new year to all, also to digium for their great work
with asterisk .
I want to make an automatic call marking an extension from my dial
plan , an example that when marking the extension 100, tell
Am Samstag, den 15.12.2007, 16:55 -0800 schrieb Philip Prindeville:
I've got the following set up:
Someone calls into my PBX on a single number (via SIP trunk from my
carrier), and the get a voice menu of extensions.
On one of the extensions, it rings a bunch of internal SIP hardphones,
Am Mittwoch, den 12.12.2007, 09:14 -0500 schrieb Lutgring, Sam:
I have a strange issue with CLID that I would appreciate if someone
could point me in the right direction. When a call comes in (either
from another SIP user on the same Asterisk box or from the ISDN PRI)
the Caller ID Name is
compared to includes in something like GCC.
:)
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
___
--Bandwidth and Colocation Provided by http://www.api-digital.com
.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
Sent: Sunday, December 09, 2007 2:21 PM
Am Freitag, den 07.12.2007, 17:53 -0300 schrieb
[EMAIL PROTECTED]:
Hi all,
I don't know if this is the right list to ask, since
I'm using Trixbox version 1.0.0.28, that has asterisk
1.2.17.
I'm trying to configure the ring timeout value for my
local extensions (when dialing from one to
info with Google.
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
Office: 888-865-0065 Ext. 202
Mobile: (585) 721-8679
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing
Am Freitag, den 30.11.2007, 15:08 -0800 schrieb Philip Prindeville:
bump...
Philip Prindeville wrote:
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no default
Am Montag, den 26.11.2007, 22:39 -0500 schrieb [EMAIL PROTECTED]:
On Nov 26, 2007 9:52 AM, Alberto Pastore [EMAIL PROTECTED] wrote:
I also found the Pirelli DP-L10 dual phone to be an excellent sip client
with good roaming support and discrete battery saving capability.
(Used in a 14-cell
Am Dienstag, den 27.11.2007, 18:00 +0100 schrieb Robert Rozman:
Hi,
I have an older phone with touch screen from Philips. It have it connected
to Sipura 3000 FXS port and majority of features work ok.
But phone also has touchscreen and web browser that I'd love to use for
accessing my
Am Mittwoch, den 21.11.2007, 13:47 -0500 schrieb [EMAIL PROTECTED]:
I just bought an Aastra 480i CT for a client who needed cordless
capabilities in their office. I'm trying to set up the base station and
cordless handset in my office first. I'm able to connect the phone to
my Asterisk
Am Montag, den 19.11.2007, 13:45 +0100 schrieb Olivier:
Hi,
My Gigaset S450ip allows 2 simulatneous calls when each incoming call
are targeted to different phones.
When both calls target the same extension, the second one is forwarded
to voicemail.
I couldn't check yet SIP messages but
Am Dienstag, den 13.11.2007, 09:29 -0500 schrieb Il Neofita:
Hi
I have the same problem
On Nov 13, 2007 9:10 AM, marcotasto [EMAIL PROTECTED] wrote:
Hi Neofita, Doug and All.
I think I've the same problem but I don't know if it's related to the bug
suggested below.
I try to explain
Am Montag, den 12.11.2007, 15:14 -0500 schrieb Il Neofita:
Hi
additional information if I am going to wait at least 3 seconds after
the voicemail starts to give me the instruction I am able to listen my
messages.
But why I need to wait?
On Nov 12, 2007 2:28 PM, Il Neofita [EMAIL PROTECTED]
Am Dienstag, den 06.11.2007, 11:49 + schrieb Tony Mountifield:
I know that I can record the contents of a call by calling Monitor()
or MixMonitor() from the dialplan just before invoking Dial().
I have a potential customer who wants only the first minute of each
call recorded (for
.
The Asterisk-Java library has a StatusAction and StopMonitorAction,
if Java is a language candidate for an application you might write.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original
Am Freitag, den 02.11.2007, 12:12 +0200 schrieb Atis Lezdins:
On 11/2/07, Michiel van Baak [EMAIL PROTECTED] wrote:
On 09:38, Fri 02 Nov 07, Tony Mountifield wrote:
Does anyone from Digium want to comment on why this Eloqua stuff has been
used, instead of just allowing Apache to serve the
:)
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
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asterisk-users mailing list
To UNSUBSCRIBE
In case it helps, I've fixed that problem before by making sure my DHCP
server gives out the correct time AND offset fields, and updating the
Polycom firmware to a recent version :).
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
Am Mittwoch, den 31.10.2007, 16:47 + schrieb Gordon Henderson:
Not strictly asterisk related, however...
No GSM!
How odd is that, given that it's a GSM mobile phone...
Maybe the GSM codec is implanted to the GSM chip and that one does
alaw, ulaw...
Anyway, my quest for the ultimate
Am Montag, den 29.10.2007, 15:54 + schrieb Gordon Henderson:
On Mon, 29 Oct 2007, Abdul wrote:
Hi,
Is it possible to have multi listening bindport in asterisk?
Now days mostly ISPs are Blocking the standard 5060 port so we want to
keep option if 5060 is blocked we can ask our
asterisk to ignore it and pass it along?
Am I stuck with absolutely no features that depend on # or * if I want
users to use those digits on a remote PBX?
Thanks :)
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
Am Dienstag, den 23.10.2007, 22:21 -0700 schrieb satish patel:
there is no special requiremnt to use g.729 but day to day my sip
client incressing thats why some time i got breaking voice or voice
quality not much better i think in LAN there is lots of brodcat on
lan
If your LAN is congested
Am Mittwoch, den 24.10.2007, 11:19 -0500 schrieb Peder @
NetworkOblivion:
This is semi-related, but I have a Tmobile MDA and I couldn't play the
files either. The issue was not a codec issue, it was an email encoding
issue. If I sent the message to an email account and it was then
Am Dienstag, den 23.10.2007, 02:56 -0700 schrieb satish patel:
Dear all
i have asterisk connected with avaya through E1 back-2-back
now when i configure my sip client with g.729 codec then i m not able
to put call from asterisk to avaya and when i user g.711 it is working
fine so i
Am Samstag, den 20.10.2007, 22:58 -0700 schrieb Philip Prindeville:
Erik Anderson wrote:
On 10/20/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
If you are trying to use non-complied (XML) profiles... don't even
bother wasting your time.
Why is that? I'm using the xml-style
Am Mittwoch, den 17.10.2007, 21:57 +0200 schrieb Michael Iedema:
Greetings everyone,
today I spent the last part of my day trying to find a parse error
inside this snip:
http://pastebin.ca/740081
If there's anyone who can shed some light on why my GosubIf condition
is throwing a parse
Am Montag, den 15.10.2007, 16:38 +0300 schrieb Cosmin Prund:
Behalf Of Anselm Martin Hoffmeister wrote:
Subject: Re: [asterisk-users] About .call files when the congestion is
on myside
Am Montag, den 15.10.2007, 11:26 +0300 schrieb Cosmin Prund:
Hello everyone.
I’m
Am Montag, den 15.10.2007, 11:26 +0300 schrieb Cosmin Prund:
Hello everyone.
I’m working on an application that needs to automatically send faxes.
To send the faxes I create .call files but the .call files mostly fail
because my lines are always congested within business hours! Is there
Am Samstag, den 13.10.2007, 15:01 +0200 schrieb Turbo Fredriksson:
I'm using Swedish on version 1.4.13. The full part of the
log is:
[Oct 13 12:51:16] WARNING[7810] file.c: File digits/ett does not exist in any
format
[Oct 13 12:51:16] WARNING[7810] file.c: Unable to open digits/ett
Am Dienstag, den 09.10.2007, 19:50 +0100 schrieb WipeOut:
Hi,
Ok.. I know dual NAT is a problem for SIP..
ie. UA - NAT - Internet - NAT - Asterisk
What about Multi-NAT where a dedicated public IP is mapped to the
private IP of the asterisk box..
ie UA - NAT - Internet - Multi-NAT -
Am Dienstag, den 09.10.2007, 14:23 -0500 schrieb Ricardo Melendez:
Hi, I would like to develop a “click to talk” app to interface with
asterisk, anyone know about some SDK/frameworks to implement this.
I have not ever used such an application, but there are several
solutions commercially
Am Montag, den 08.10.2007, 11:07 +0200 schrieb Vincent:
Hello
Now that I received an OpenVox PCI card
(www.openvox.com.cn/products_detail.php?genre_id=9id=28), I'm ready
to try and set up a voice server with Asterisk.
We need the following features:
1. When customers call in, they should
Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
Office: 888-865-0065 Ext. 202
Mobile: (585) 721-8679
___
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--Bandwidth and Colocation Provided
Am Mittwoch, den 26.09.2007, 11:08 +0200 schrieb Bastian Friedrich:
Hi,
I am currently setting up a voice mail/IVR machine with asterisk (1.4.10
at the moment). During testing and evaluation, all was fine; in the -
slightly different - production environment, the IVR contexts do not
Am Dienstag, den 25.09.2007, 07:37 +0200 schrieb Guenther Sohler:
Hallo Group,
I have basically set up a small asterisk system,
which ahs 4 peers:
* registers at 2 Sipgates
* 2 hardware phones connected to it
Both Hardware phones can phone outwards(cheaper sipgate is selected with
Am Dienstag, den 25.09.2007, 11:01 + schrieb dadsadsadf dsadasdsa:
Hola Jonathan
Te cuento un pokillo lo q intento hacer por si me puedes orientar en algo o
de algun sitio donde pueda mirar
Existe una especificación de Microsoft de lo que llaman
Dual-Forking, que básicamente consiste
Am Donnerstag, den 20.09.2007, 08:30 +0200 schrieb Guenther Sohler:
Hallo Group!
My Name is Guenther Sohler and I registred to this group, because
I think asterisk could be interesting for me.
Hi Guenther, this place probably is the right one. Welcome!
I have got a small server at home
Am Mittwoch, den 19.09.2007, 15:25 +0200 schrieb Christoph Adomeit:
Hi there,
I experience the same problem here with asterisk 1.2.24 on
an E1 Line, only 2 of 3 sms are sent, this happens always and
is reproducable.
Did someone find out more about the problem ?
Especially I do not
of order. In
addition, if I use Wireshark's voip call player, the outgoing side of the
call stutters and is delayed compared to the incoming side of the call.
Is this normal? Why would the PBX be sending packets already out of order?
Thanks!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road
it a long time ago. When you included pricing,
your email became commercial, an advertisement, and spam.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED
Am Mittwoch, den 19.09.2007, 11:02 +0530 schrieb Jim Boykin:
Can someone suggests a good and resonable cost voip provider with
business unlimited plan in USA and allows simultaneous outgoing
calling.
My experience with business unlimited is that they very well know which
customer uses more
I tried what you sent me and what Barry sent me and it seems to have
fix the problem. I don't know what was wrong but I rebooted the
server after doing both command even if it gave me no messages and
now everything seems to work fine.
Thanks
Martin
On 19-Sep-07, at 6:21 PM, Philipp
asterisk give me error messages and try restarting
until I log in as root and type service zaptel start then it load
zaptel correctly and everything seems to be fine from there. I was
wondering what I have to change to make sure zaptel load correctly at
startup?
Thanks
Martin
Am Dienstag, den 18.09.2007, 17:33 -0400 schrieb James FitzGibbon:
On 9/18/07, David Gomillion [EMAIL PROTECTED] wrote:
I've stayed out of this thread for a long time, and really
didn't read the past comments, so if I'm repeating someone,
I'm sorry. I've been thinking
Am Montag, den 17.09.2007, 05:09 -0700 schrieb satish patel:
Dear all
I have setup of asterisk 1.4.11 Now i want soft phone
which one support file sharring + video + voice call with asterisk SIP
is there any soft phone which support this all feature ??
Yes, there is such a soft
Am Montag, den 17.09.2007, 15:50 -0400 schrieb [EMAIL PROTECTED]:
Hello,
I have a small LAN network where I am running a Jain-Sip softphone on two
user pc's.
These softphones are connected through Asterisk(Trixbox). Although the
phones do
work in providing an audio conversation, there
Am Dienstag, den 11.09.2007, 17:11 +0530 schrieb Benjamin Jacob:
Thanks Anselm. This does clears a few things for me.
Tho, I couldnt find the patterns you mentioned in the docs(do point me
to the location if you know of it).
I started on
Am Dienstag, den 11.09.2007, 19:09 +0500 schrieb Rizwan Hisham:
The whole point of doing this is because if the user gives away his
username/password to his friends or relative and allows them to use
his account, that way we r gona have a lot more traffic in our
asterisk server.
Also we
Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
Hi,
my ATA has two phones attached and the possibility to set different
accounts.
I put two account of my asterisk server, however, it is able to call
only with the second one in order to the sip.conf and the first it
gives me 403.
Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita:
On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED]
wrote:
Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
Well, it seems there are differences between those accounts
Am Mittwoch, den 05.09.2007, 22:58 -0700 schrieb satish patel:
Dear all
I have FAX machine connected with audiocode SIP device
i am trying to send fax and when negosiation going on and i start send
fax button then my after half page it got stuck in fax machine so is
there any
Am Donnerstag, den 06.09.2007, 10:16 +0530 schrieb Benjamin Jacob:
Hello ppl,
Any way to specify alphabetical exten patterns in the dialplans on Asterisk?
All my users would have alpha/numerical ids. I don't want to add a line
for every user in my dialplans.
I searched around, but couldn't
Am Donnerstag, den 06.09.2007, 02:07 -0600 schrieb Al lists:
Hi list,
I'm trying to get some ideas on this subject.
Normally astersik sends emails with voicemail attached trough local
MTA.
As far as i know there is no way for asterisk to authenticate to an
external mailserver to relay these
?
Thank you!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
Office: 888-865-0065 Ext. 202
Mobile: (585) 721-8679
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--Bandwidth
Am Sonntag, den 02.09.2007, 23:25 -0700 schrieb fateme fatah:
Hi:
When should we use unnumbered priorities(n) in extensions.What is
the different between these 2 forms of extensions.conf? and ,Are both
true?
extensions.conf:
form1:
[Conferencerooms]
exten = 333,1,Answer
exten =
Am Montag, den 27.08.2007, 08:55 -0400 schrieb Jody Gugelhupf:
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before
i was using asterisk
1.4 and had the same problem, it concerns an italian voip/sip provider called
eutelia/skypho, my
problem is the following one:
Am Montag, den 20.08.2007, 13:57 -0400 schrieb Joe acquisto:
Excuse me if I recently posted on this, but I cannot find it, in my, or the
list archives.
Is it possible, when transferring a call, to set the user ID to that of the
outgoing number instead of the incoming number?
I believe the
Am Dienstag, den 14.08.2007, 09:06 -0500 schrieb Brandon Kruse:
I just use
exten = +12564286115,1,Goto(${EXTEN:1})
exten = 12564286115,1,noop(It worked.)
I believe that should work
That rewrites the callee number, not the CALLERID, so no, it would not
work for Todd's original problem.
BR
Am Donnerstag, den 16.08.2007, 12:08 +0100 schrieb Gordon Henderson:
On Thu, 16 Aug 2007, Diego Iastrubni wrote:
DUD! THIS KICKS ASS!
(I know I am getting into trouble, but hey! it's already in our PBX!)
Heh... Well I updated it and added some lyrics (and the guys from the
website
Am Mittwoch, den 15.08.2007, 10:14 -0400 schrieb Lee Jenkins:
Hi everyone,
I have been dealing with a certain issue with a particular customer site
for months now. The problem occurs when there is an error with caller
id as shown in the following:
WARNING[16036]: chan_zap.c:6309
Am Sonntag, den 12.08.2007, 21:16 -0400 schrieb C F:
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-10]});this
Hello Daryl,
See
http://www.asterisk.org/doxygen/1.4/res__agi_8c.html#c631d48f46d51d4b057
b31807baa1f10
The AGI application will answer the channel if it isn't already
answered.
You probably need to do whatever you want to do in the dialplan, and
keep using DeadAGI.
Martin Smith, Systems
Am Donnerstag, den 09.08.2007, 20:12 -0500 schrieb David Bandel:
Folks,
I'm trying to implement a simple loop in a dialplan. The object is to
set a counter, run through some IVR options, increment the counter,
return to the start, then finally fall through to an operator or
voicemail.
Am Freitag, den 10.08.2007, 09:02 +0200 schrieb Olivier:
Hi,
My question is more what should be done than how should it be
done.
I could say :
If you were a teacher, teaching and preparing your courses once a
week (as you can't be called while teaching, can you ?)
Well, yes. It always
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