Hi,
Phone vendors (Snom, Thomson-Technicolor, ...) are on the way to support
TR-069 (see http://en.wikipedia.org/wiki/TR-069).
Has someone experienced with TR-069 ?
What do you think of this protocol set ?
Regards
--
_
--
Hi,
sorry if the question seems silly, but for some reason, all these phones,
modems, ATAs use TR-069 instead of SNMP ?
I have experienced it, but as user, for a small time.
Regards
On 16 February 2010 09:54, Olivier oza_4...@yahoo.fr wrote:
Hi,
Phone vendors (Snom, Thomson-Technicolor,
16 feb 2010 kl. 08.54 skrev Olivier:
Hi,
Phone vendors (Snom, Thomson-Technicolor, ...) are on the way to support
TR-069 (see http://en.wikipedia.org/wiki/TR-069).
Has someone experienced with TR-069 ?
What do you think of this protocol set ?
And the SIP forum is about to release
On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri lenz.lo...@gmail.com wrote:
Yes but in any case you can enter all of the strings that reasonably match
- even if you have variable-length numbers, you will be able to determine
Hello list,
debugging SIP, I found many empty lines like:
--- SIP read from UDP://XXX.XXX.XXX.XXX:5060 ---
-
The IP address above corresponds to one of my accounts, which
is behind a firewall.
Is that normal, maybe some firewall that
16 feb 2010 kl. 09.43 skrev Tzafrir Cohen:
On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri lenz.lo...@gmail.com wrote:
Yes but in any case you can enter all of the strings that reasonably match
- even if you have variable-length
Hi,
did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It
looks related to your issue.
Best regards, Marcus
On Fri, Feb 12, 2010 at 12:04 PM, Armin Schindler ar...@melware.de wrote:
On Fri, 12 Feb 2010, Armin Schindler wrote:
I had a look at
netstat -nuap
and it
16 feb 2010 kl. 10.40 skrev Alexandru Oniciuc:
Hello list,
debugging SIP, I found many empty lines like:
--- SIP read from UDP://XXX.XXX.XXX.XXX:5060 ---
-
The IP address above corresponds to one of my accounts, which
is behind a
call transfer
call transfer from reception to other extensions.
Question: Details of Extensions
On Tue, 2010-02-16 at 17:25 +0530, cool dude wrote:
call transfer
call transfer from reception to other extensions.
Question: Details of Extensions
Reception - 2000
Sales - 2001
Accounts - 2002
any call comes it should be received by extenion 2000, n if person
wants to talk to
On Tue, 16 Feb 2010, Marcus Hunger wrote:
Hi,
did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It looks
related to your issue.
Oh thanks, I missed that one.
It really looks related. I have added a note.
Thanks,
Armin
Best regards, Marcus
On Fri, Feb 12, 2010 at 12:04
Tuesday, February 16, 2010, 12:55:12 PM, cool wrote:
call comes it should be received by extenion 2000, n if person wants to
talk to Sales, receptionist should put the caller on hold than connect
to Sales i.e exten 2001, while on hold the caller should hear music on
hold,now sale exten can
Okay, so my issue isn't really a technical one but more of needing
advice on the best way to program this. I have a user in Colorado who
works from home but frequents our office in Colorado. All of our
remote users connect to a server in Dallas the users at the HQ in
Colorado connect to a
Tilghman Lesher wrote:
On Monday 15 February 2010 18:01:11 VinÃcius Fontes wrote:
He probably means AgentCallbackLogin
While it has been deprecated, that hasn't been removed, either. If
an
enterprising person would like to try to fix it, I don't have an
objection.
Wasn't
Do a qeuee, add each as a station in the quee..
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Anness
Sent: Tuesday, February 16, 2010 10:04 AM
To: asterisk-users@lists.digium.com
Subject:
I apologize, my English isn't better than the last year
So,
- why Queue has some options like 'caller can continue in his dialplan' (Dial
has g and F options for callee/caller) and not he same option for 'callee' ?
- no option in queue to send message (for only calle/caller) when bridged as
On Mon, Feb 15, 2010 at 11:05 AM, Amit Patkar | Avhan Technologies
Pvt. Ltd. a...@avhan.com wrote:
I have a server with Quad Core Xeon 2.4GHz and 4GB RAM. I want to use it for
PSTN-IP gateway. What is the maximum call handling capacity I can achieve
with this server?
You can handle a lot of
Is anybody currently using the SMS application? We've had some reports of the
current SMS application not working, and we're really not equipped on this
side of the pond (USA) to work on figuring out what the problem is, or even if
there is a problem at all. I've attempted to get the SMS
callee in queue-land is always an agent who picks up the call instead of
getting it transferred. In some concepts, queue is more like a conference
than a transferred/dialed call. Item 3 could be accomplished with an
AGI/AMI command.
_
From: asterisk-users-boun...@lists.digium.com
I had a customer ask me this question today, and I was surprised to say I
didn't have an exact answer for them. They have a relatively small support
queue for their business (three agents, and rarely more than one person in
line at any given time in the queue, if all agents are on a call), Their
What is the current best practice to avoid transcoding on an outgoing call
to a
party whose codec preference is not known in advance?
In other words, incoming calls are easy since codecs are negotiated from
least-known (the remote party) to most-known (my endpoint) and my codecs can
simply be
Hi,
We are using asterisk version 1.0.0.
For queue'ing we use the rawplayer script to play a music file in the
background.
Now we see that after a while all the sessions on our Linux environment
will be taken by the rawplayer process.
An example of such a session is (done with ps
I've Audiocodes MP-114 registered per-endpoint (2x FXO / 2x FXS) but when call
comes on pstn- it goes to context fax-incoming
in sip.conf:
[pstn-]
type=friend
context=incoming
...
[pstn-9998]
type=friend
context=fax-incoming
...
the device register per end point just fine, so it
On Tue, Feb 16, 2010 at 1:43 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri lenz.lo...@gmail.com
wrote:
Yes but in any case you can enter all of the strings that reasonably
On 16 Feb 2010, at 17:36, Arjan Kroon | Mobillion wrote:
We are using asterisk version 1.0.0.
Wow.
S
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On 02/17/10 03:39, Warren Selby wrote:
I had a customer ask me this question today, and I was surprised to
say I didn't have an exact answer for them. They have a relatively
small support queue for their business (three agents, and rarely more
than one person in line at any given time in the
I using the asterisk yum repository at
http://packages.asterisk.org/centos/5/current to install a vanilla asterisk.
All went well on my development server, which is a fully up to date CentOS5
machine. But now I am trying to do the same with my production server, which is
an OEL5 machine with
On 02/17/10 05:01, Steve Howes wrote:
On 16 Feb 2010, at 17:36, Arjan Kroon | Mobillion wrote:
We are using asterisk version 1.0.0.
Wow.
Yeah, that about sums it up. A little googling reveals that Asterisk
1.0 was announced on January 14th, 2005 - over five years ago.
I would
On Tue, Feb 16, 2010 at 3:01 AM, Olle E. Johansson o...@edvina.net wrote:
16 feb 2010 kl. 09.43 skrev Tzafrir Cohen:
On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri lenz.lo...@gmail.com
wrote:
Yes but in any case you can
stephen.hindma...@bt.com wrote:
rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec
snip
error: Failed build dependencies:
kernel-devel = 2.6.18-164.11.1.el5 is needed by
dahdi-linux-kmod-2.2.1-1_centos5.2.6.18_164.11.1.el5.i386
Add a --target=i686 to your rpmbuild
On Tue, Feb 16, 2010 at 10:53:16AM -0700, Steve Murphy wrote:
(is there some escape mech in the syntax to let you say \NA\NCY? I
haven't checked).
[N]A[N]CY . Or, if we have it your way, [N][A][N]C[Y]
But, there's no reason we can't add other matching chars
for handy things. A = alpha
Hi Karl,
that's funny you are asking this, am also currently looking at how to
solve the g722 codec negotiation riddle, in my particular case to play
nicely together with a KonfTel 300 IP conference phone.
In other words, incoming calls are easy since codecs are negotiated
from least-known
OK Danny, but of course, i don't mean 'exactely the same options'.hum i
don't really understand why it's possible for the caller to continue in his
dialplan (= cmd Dial option g) and it's not possiblef or the callee (as option
'F transering' on DIAL).
Reading the source code in
Audiocodes is not distinguishing/identifying correctly which device calls comes
through IN; I think it just take the fist identifier base on internal
registration. So the call goes to a wrong context.
--
Joseph
On 02/16/10 10:48, Joseph wrote:
I've Audiocodes MP-114 registered per-endpoint
Hello,
I have several voicemail accounts that share the same IMAP mailbox (same
imapuser), and I'd like to store their messages in separate IMAP
subfolders. I tried to set 'imapfolder' settings on a per voicemail
account basis, but it does not seem to work. All messages keep being
stored in the
At 9:04 AM on 16 Feb 2010, Steve Anness wrote:
[...]
exten = 12109,1,Dial(iax2/castle-rock/109iax2/colo/17128,20)
This has one flaw, if for whatever reason his home phone isn't
connected, like it isn't now, than when I call 12109 I get his home
voicemail as it picks up right away.
How
Hi!
But, there's no reason we can't add other matching chars for
handy things. A = alpha chars Y = alphanum chars, G = Graphical chars,
Pretty please!
Philipp
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In concept, how I would do #3 is to make an AGI that has a timer and warn
intervals, like this
- exten = 100,1,AGI(timer.agi,3600,300) to set a timer for 60
minutes and play a message at 5 minute intervals.
In a post last year, Jared Smith posted a snippet to timeout a call. It
Hello My friends,
Today my asterisk stop working and i could see the following messags in
/var/log/asterisk/messages at the time that asterisk stop working:
[Feb 16 13:23:40] NOTICE[8230] chan_sip.c: Peer '324' is now Reachable. (2ms
/ 2000ms)
[Feb 16 13:24:41] NOTICE[8230] chan_sip.c:
Just curious to know how most of you deal with segmentation faults /
core dumps
These are a few of the things I have seen in regards to people dealing
with random crashes:
1. Apply Base OS updates
2. Recompile with DEBUG_THREADS and DON'T_OPTIMIZE turned on and look
for the cause of the seg
I have read the posts about the security issue and from what I
understand there should be a check to make sure that the characters
used are actually allowed. I wrote a very straightforward and not so
rocket science kind of macro that will do the job I guess. Just two
parameters, one with
I have this:
[menu]
exten = _X.,1,answer()
exten = _X.,2,wait(1)
exten = _X.,n,GoTo(ivr,s,1)
[default]
include = record
include = incoming
include = menu
[local-dial]
exten = _1XX,1,Verbose(. In local-dial context, dialing exten: ${EXTEN}
.
exten =
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
I have tried to replicate this, but with no luck. If I use a SIP-client
that supports '', I still get a reject from asterisk.
I am writing a filter for lua (pbx_lua), but it's a bit hard when I
cannot reproduce and test this. I've tried with both
Hello List.
I am puzzled and how asterisk listens to calls or connections from clients.
When I do a netstat -nat I don't see asterisk listening on port 5060. Now, I'm
testing a server with three network interfaces: two to the internet doing
load balancing and the other to our LAN. I would
At 2:50 PM on 16 Feb 2010, Landy Landy wrote:
Hello List.
I am puzzled and how asterisk listens to calls or connections from
clients. When I do a netstat -nat I don't see asterisk listening on
port 5060. Now, I'm testing a server with three network interfaces:
two to the internet doing
On Tue, 16 Feb 2010, Landy Landy wrote:
I am puzzled and how asterisk listens to calls or connections from
clients. When I do a netstat -nat I don't see asterisk listening on port
5060.
man netstat. See what -t means.
Now, I'm testing a server with three network interfaces: two to the
On Tue, Feb 16, 2010 at 4:38 PM, meetmecall i...@meetmecall.nl wrote:
I have read the posts about the security issue and from what I
understand there should be a check to make sure that the characters
used are actually allowed. I wrote a very straightforward and not so
rocket science kind of
I am puzzled and how asterisk listens to calls or connections from
clients. When I do a netstat -nat I don't see asterisk listening on port
5060.
That is because you need to do 'netstat -nau'. Only very recently
Asterisk has learned to do SIP over TCP.
Philipp
--
Hi!
I know I can do it with iptables and block incoming connections to
ports 5060-5070 from the internet but, wondering if it can be
confiruged in asterisk.
Iptables would be the right place, though.
Still: Look at 'permit'/'deny' in sip.conf, or use dialplan magic to
check on IP addresses
On Tue, Feb 16, 2010 at 5:26 PM, Steve Edwards asterisk@sedwards.comwrote:
See http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
Search for bindaddr.
--
Thanks in advance,
-
Steve Edwards
Order of configurations does make a difference so you may want to try with
the same order as the one that works. Saw it on the voip-info.org wiki
somewhere, but can't get you the link at the moment.
In general, if you will be authenticating based on IP, you should leave
username/secret out.
Some
I didn't know about the function but from what I understand from the
show function FILTER output it doesn't validate a string but it
cleans the string from not allowed characters. So
TRIM(1234567890,01243567505) results in 01243567505. If the length
of the output string is shorter then
See http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
Search for bindaddr.
Or udpbindaddr for 1.6.2+...also,
tcpbindaddr, tlsbindaddr if you plan
on adding TCP/TLS SIP support to asterisk.
Thanks to everyone who replied for clarifying.
--
Thanks for the input.
I know that order in extension.conf makes a difference but I did not know that
it applies to sip.conf as well.
I would like to find this article you have mentioned on WIKI what should I look
for :-/?
--
Joseph
On 02/16/10 18:50, uzzi wrote:
Order of configurations does
On Tue, Feb 16, 2010 at 6:28 PM, meetmecall i...@meetmecall.nl wrote:
I didn't know about the function but from what I understand from the show
function FILTER output it doesn't validate a string but it cleans the
string from not allowed characters. So TRIM(1234567890,01243567505) results
in
I use IdeaSip with IPKall.
How may channels are open when we use IdeaSip ?
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