Am Montag, den 23.07.2007, 06:44 -0700 schrieb satish patel:
Dear all
I have configure asterisk with 100 SIP PHONE ( SNOM )
but now thing is that my boss need phonebook feature find extention
number by Pbook so i have read about it there is a feature in asterisk
but it is
Am Montag, den 23.07.2007, 16:21 -0400 schrieb Michael J. Liberatore:
I noticed in 1.4.x I can no longer use n+101 ? I use this all over my
dial plan and wouldn't even know how to replace it. Like when trying to
call out and a channel is busy, would I need to do all if then's??? How
can I
Am Montag, den 23.07.2007, 14:33 -0400 schrieb Matt:
Hi,
What dialplan option do I need to send a call out like this:
NPA-NXX- local calls
1-NPA-NXX- - long distance
Won't 'national' send it out NPA-NXX- no matter if it's long
distance or not?
I do not understand your point
Am Dienstag, den 24.07.2007, 11:26 -0500 schrieb John Faubion:
To prevent further missunderstanding please do not refer the SI-120
as a snom
phone. If you need support please contact snom India.
Tim,
If it is sold by snom India, and one is to contact snom India, I can
certainly see
Am Mittwoch, den 25.07.2007, 12:13 -0700 schrieb bilal ghayyad:
Hi BaharatSamaria;
Thanks for your kindly email.
Are (Xlite and phoner) IAX or SIP? From where I can
download them (Xlite and phoner)?
I googled for xlite. One of the first matches was a wiki page on
voip-info.org, which in
Am Montag, den 30.07.2007, 05:24 -0700 schrieb Vieri:
Hi,
I would like to know if one can set the outgoing
caller ID within Asterisk when calls are going out
through:
1) an analog POTS line (I suppose not)
2) a telco BRI line (I don't think so)
3) a telco PRI line (maybe)
4) a voip
Am Montag, den 30.07.2007, 14:29 -0700 schrieb Lee Howard:
http://www.asterisk.org/node/48327
I mean, really... you're kidding me, right?
It is not at all April 1st... however, I see the point in having a
simple demo app. Wether you call it helloworld or hellomarc, the
difference is not too
Am Dienstag, den 31.07.2007, 07:39 -0500 schrieb Asterisk guy:
1and1 dedicated server's service has been down for a few hours ,
unable to reach them by phone or email. do anyone know what is going
on there ?
There were rumours they had trouble with an outdated version of the
web
Am Mittwoch, den 01.08.2007, 16:32 +0530 schrieb Benjamin Jacob:
Hello good ppl,
A couple of questions for multiple pbxes
1. Is it possible to support multiple pbxes in one Asterisk box(using
contexts, etc.)?
2. Can we use the domain field in sip.conf to specify the different
domains for
Am Montag, den 06.08.2007, 18:09 +0200 schrieb gincantalupo:
Hi,
I'm trying to use a Detewe TA 33-clip but there is no rj11 connector on
it...only a TAE connector.
I'd like to create an adapter so I need to know which TAE pins to
connect to RJ 11 pins.
Is there anybody who knows where I
Am Dienstag, den 07.08.2007, 16:51 +0200 schrieb Olivier:
Hi,
Where can I find relevant information concerning callto:// tags ?
Is it standardized or browser specific ?
How within your browser, can you specify the software and parameters
to used when clicking on such callto:// tags ?
I
Am Dienstag, den 07.08.2007, 07:47 +0200 schrieb Olivier:
So no proper logoff between logins, right ?
As I will apply free sitting in school environment, chances are phones
would then remain logged-in several hours or days between another user
logs in.
My thoughts are focused on finding
Am Mittwoch, den 08.08.2007, 23:55 +0900 schrieb Balgansuren Batsukh:
Hello,
I installed Asterisk on Dell Precision workstation and configured with
sample configuration.
I have two TDM400 board and one with 4xFXO and second one 4xFXS module
installed.
I made call to telephone line
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
Am Freitag, den 10.08.2007, 11:26 -0500 schrieb Peder @ NetworkOblivion:
That's great, now say you have 5 or 6 AA's and each one has 10 different
parts that you want to record (thank you for calling... for Steve
press 1 for dave press 2). Rather than having to record a long
message, I
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(${CALLERID(num)})}-10]});this
Am Mittwoch, den 15.08.2007, 10:14 -0400 schrieb Lee Jenkins:
Hi everyone,
I have been dealing with a certain issue with a particular customer site
for months now. The problem occurs when there is an error with caller
id as shown in the following:
WARNING[16036]: chan_zap.c:6309
Am Dienstag, den 14.08.2007, 09:06 -0500 schrieb Brandon Kruse:
I just use
exten = +12564286115,1,Goto(${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 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 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 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 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
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 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 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 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 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
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
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 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 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
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 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 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 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 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 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 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 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 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
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
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
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
ROTFL
Trafrir, you made my day.
(BTW: I
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
Am Samstag, den 28.06.2008, 08:15 -0500 schrieb [EMAIL PROTECTED]:
Hi List
I have build an IVR on Asterisk from 1.2 to now 1.4.18 and has already
processed more than 10million calls!
I have one big challenge which is reporting... it is the requirement to
have a web reporting module which
Am Montag, den 14.07.2008, 09:45 -0700 schrieb bilal ghayyad:
Hi All;
Anyone can advise for a method to have open vpn client to be installed on the
mobile, so it can open a vpn channel with Asterisk (I installed open vpn at
it) from the mobile, and then I can let fring use the open vpn
Am Dienstag, den 15.07.2008, 14:02 -0400 schrieb Fidel Garcia:
Need to have a different TONE for any internal call (EXT OR TRANSFER)
from an external (outside) call.
Any suggestions?
Fidel,
I do not know what kind of tone you mean:
The sound of a phone that signals a call coming from
Am Dienstag, den 22.07.2008, 14:53 -0500 schrieb Gregory Malsack:
Hello,
My boss is asking me to setup the asterisk server to record all calls.
(Simple). However, he wants to be able to enter a key sequence during
the call to stop the recording. Any ideas on how I would do that?
Hi
Am Dienstag, den 19.08.2008, 02:53 + schrieb Miguel Otamendi:
Please, I need help.
I have problem witch voicemail.
-- Executing [EMAIL PROTECTED]:3] VoiceMail(Zap/4-1, s) in new stack
[Aug 17 21:33:46] WARNING[11864]: app_voicemail.c:3061
leave_voicemail: No entry in voicemail
Am Freitag, den 29.08.2008, 09:16 -0700 schrieb Ira:
At 05:48 AM 8/29/2008, you wrote:
(so since they still liked the Snoms otherwise, my solution is to get them
to dial a star at the end of a number to select their 'home' account,
otherwise it goes out on their work account and the dialplan
Am Dienstag, den 07.10.2008, 01:42 -0700 schrieb Vieri:
Hi,
Currently I'm using Asterisk 1.2 and 1.4 in different setups. When a user
wants to pick up a call
within his/her pickup group, *8 must be dialed (or whatever you define in
features.conf).
[...]
I was thinking of configuring some
Am Mittwoch, den 18.04.2007, 13:18 +0200 schrieb Knud Müller:
Dinesh Nair wrote:
On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote:
I
think it can be done by using the dialplan and the database to store the
statistical information but maybe there is an easier way that integrates
Am Mittwoch, den 02.05.2007, 20:04 +0100 schrieb Goke Aruna:
Hello all,
I have a set up that answer my customer. and its working well,
however, the number of call to technical dept is what i want to reduce.
I want all call to get to voice prompt except that that enter when
minutes is
Am Freitag, den 04.05.2007, 00:48 -0400 schrieb Doug Crompton:
Can anyone suggest a source for a free robot dialer or examples? I need to
do some non-commercial auto dialing using Asterisk. Simple phone numbers
in a file, line by line format.
I found one called AstAutoDiaker but I was not
Am Sonntag, den 06.05.2007, 00:48 -0400 schrieb Salvatore Giudice:
Just forward them to 1-800-big-dick or some other 800 toll free phone sex
line. They can't tell they've been forwarded. They'll figure it out
eventually.
Whoa, that was _my_ coffee that's now on the screen.
I will urgently
Am Freitag, den 11.05.2007, 18:44 -0400 schrieb Jon Pounder:
just out of curiousity - anyone ever hijack pairs and get away with it ?
(do your own cross connects on the street and utilize some crossconnect
all within one branch of F1 cable out of the CO ?)
I've been tempted in the past, and
Am Donnerstag, den 17.05.2007, 10:40 +0200 schrieb [EMAIL PROTECTED]:
Hi all.
We have Snom phones which do have a defined key in order to drop incoming
call WITHOUT answering.
Pressing that key, a SIP/2.0 486 Busy Here message is sent back.
We have other phones (I.E. DECT Siemens C450IP,
Am Montag, den 21.05.2007, 23:16 -0500 schrieb Mike Hammett:
If it is easy, could you enlighten me? I have another thread on caller ID
matching, but I haven't received any positive responses.
In the context where your internal calls usually are handled, like this
(my internal phones have SIP
Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player:
Hello,
i just want to activate SMS service between my asterisk local sip
accounts and between asterisk and local sip accounts. How can i do
this thin? Also i tried smsq to an account but all i obtained is a
error message:
Am Dienstag, den 22.05.2007, 17:35 +0300 schrieb Jonson Player:
Thank you for reply. Can you send me some working configs? I'm still
confusing about this sms option.
Just to get you started, try this:
Find out which user asterisk runs as. Get a shell for that user.
Run (all in one line)
smsq
Am Dienstag, den 22.05.2007, 21:49 +0300 schrieb Cosmin Prund:
Googling arround I found a number of pocket pc softphones. Of those I was
only able to install SJ-something (removed it).
Is there one (pocket pc softphone) that works?
When I searched for one, about half a year ago, there were
Am Dienstag, den 22.05.2007, 20:37 -0500 schrieb Eric ManxPower
Wieling:
David Florella wrote:
Thank you knox. Finally, I have chosen this solution : find
/var/spool/asterisk/voicemail/default/*/Old/ -atime -7|xargs rm –f, executed
every night by the CRON. However, I would have preferred
Am Donnerstag, den 24.05.2007, 08:23 +0300 schrieb Cosmin Prund:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Remco Post
Sent: Wednesday, May 23, 2007 10:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Am Donnerstag, den 24.05.2007, 10:44 +0200 schrieb dima:
Hello, everyone.
I'm having a strange problem with my asterisk. After dialing and before
the other side picks up the phone I should hear the tones (I'm not sure
what are they called: p---pii) and in almost
Am Samstag, den 26.05.2007, 02:45 -0700 schrieb Crazy Boy:
Hi Friends,
I am planning to buy IMate PDAL mobile phone. This contains Wi-Fi
802.11b/g feature. So, Is it possible to get internet using my
wireless router in my office?
Most probably yes. The device runs windows, so it comes with
Am Samstag, den 02.06.2007, 11:34 +0200 schrieb [EMAIL PROTECTED]:
Hi,
Problem is:
I have a Dell 1950 server with 6 NIC's ( 1 for Voice / Asterisk rest of
them for other functions).
The Voice LAN is on the 172.16.3.0 (255.255.0.0) subnet. One the other
NICS there are different but also
Am Donnerstag, den 07.06.2007, 01:15 +0200 schrieb Patrick Zwahlen:
Hi everyone,
How do you send multiline SMSs using smsq or .call files ?
smsq --motx-channel=mISDN/g:bri/ 078 line1 line2
How can I have a carriage return between line1 and line2 ? I have tried
the regular \n and
Am Dienstag, den 12.06.2007, 09:57 -0400 schrieb Shad Mortazavi:
Dear Group,
I have a scenario where I would like to change the caller ID based on
the number dialled;
For example;
;Outbound UK and London Calls
exten=_8.,1,Set(CALLERIDNAME=0207100)
Am Samstag, den 23.06.2007, 09:52 -0300 schrieb Ronaldo Z. Afonso:
Hi all,
Does anybody know any USB phone that I can use as an IAX Client?
The USB phones I saw on the market just behave like an additional
sound card, with some control buttons perhaps, and those will not work
without a
Am Donnerstag, den 28.06.2007, 07:07 +0200 schrieb Adam KOSA:
Hi guys,
sorry for the long e-mail, i'm only trying to give as much information
as i think is relevant to my problem (console log, sip.conf and
extension.conf parts). I've sent this e-mail a couple of days ago, but
it bounced
Am Mittwoch, den 04.07.2007, 11:00 -0400 schrieb Noah Miller:
Is it just me? After the mail list server upgrade, the average delivery
time for messages to the users list is between 4 and 5 days. The Dev
list seems fine!
I'm getting new messages within a matter of minutes. I dunno.
As
Am Freitag, den 29.06.2007, 14:23 -0600 schrieb Anthony Francis:
Andres Paglayan wrote:
On Jun 29, 2007, at 12:50 PM, Lenz wrote:
Hello list,
I am getting the list with days of delay, take for example this
message:
As you can see, the message was posted on June 25th and was sent to my
Am Montag, den 09.07.2007, 17:21 +0200 schrieb Matthias Huber:
When i send more than one messages shortly after the other, my log
(/var/spool/asterisk/sms ) looks like this
and only two of four messages arrive.
What am i doing wrong ?
I am using an AVM B1 PCI with chan-capi and 1.4.4.
Am Montag, den 16.07.2007, 09:44 -0500 schrieb Jeremy Mann:
Does anyone know if X-Ten or SJPhone support multiple cordless
handsets for multiple lines? I have an office with multiple roaming
users(nurses) that are in and out. I’d like to provide them
telephones, and my idea is to have a PC
Am Donnerstag, den 12.07.2007, 16:57 -0700 schrieb Russ McBride:
Newbie question(s):
From what I can determine it sounds like the SMS messaging isn't as
robust as it could be (?). I'm wondering if there's active work on
that right now or if it's more of an issue about PSTN carrier
Am Dienstag, den 23.01.2007, 05:41 -0200 schrieb Barzilai Spinak:
I've seen several examples that use extensions such as;
s-BUSY
s-NOANSWER
etc...
It's more or less evident what they do, but I've searched for some
FORMAL documentation everywhere and have found nothing.
Do they work for
Am Montag, den 29.01.2007, 11:58 +0100 schrieb Thomas Winter:
Hi,
If I develope an dialplan, some AGI and AMI functions for Asterisk and ship
it
as an complete product to an coustomer, do I have to put my developed code or
the complete product under the GPL?
IANAL, but in my understanding
Am Donnerstag, den 01.02.2007, 16:15 -0600 schrieb Larry Alkoff:
I wish to have my Grandstream GXP-2000 phones make a different
distinctive ring for internal calls ( Internal ) or if the incoming call
has no caller id 'NOCID'.
The Grandstream phones calls allow 3 distinctive rings
Am Mittwoch, den 07.02.2007, 21:57 -0800 schrieb Jason Kim:
Hi,
This is the configuration I want.
Hard Video phone---video---Soft Video Phone(PC)
^
|
audio
|
V
Audio Only Phone
Any idea?
You could see wether having a second call that does a
Am Mittwoch, den 14.02.2007, 07:17 +0800 schrieb Ronald Wiplinger:
Where can I get a starting point for setting up sms via VoIP and via web.
I want to send SMS from VoIP or web to VoIP phones and GSM phones.
1. how to set-up?
2. which smsc should I use? (what is the price?)
3. which
Am Dienstag, den 13.02.2007, 21:41 + schrieb Razza:
Hi all, is it possible to to dumb down a FRITZ!Box Fon
ata (http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_ata/index.html##)
and have the two FXS ports AND the ISDN interface register with Asterisk. In
much the same way a sipura
Hi list,
I bought two UTStarcom F1000 phones, pre-equipped with the latest
firmware, including WPA support. Those are configured to register to an
asterisk server on the internet (not LAN), and registration works.
Calling and being called also, with transfer and all bells and whistles.
After a
Am Montag, den 19.02.2007, 12:39 -0700 schrieb Robert Norton - SophMedia
LLC:
Hey Guys,
I’m curious if there’s an interest in a free, CallerID database? For
those of you in the same spot we are, our current provider only
provides us with the CND, excluding CNAM.
Would creating a public
Am Dienstag, den 20.02.2007, 14:54 -0500 schrieb Mike Lynchfield:
Well caching is the way to go., bu then again most of the current
solutions have this problem.
John smit has a DID.. 514 555 1234 and closes account.. did sleeps for
3 months and new client Jane doe takes it..
Now how long
Am Dienstag, den 20.02.2007, 16:33 -0700 schrieb Natambu Obleton:
I would guess that registration would be by the telco for the blocks
just like with reverse dns today, so then each telco would have a
local server to manage their 'reverse' cnam lookup and the people
in charge would be
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