Jonathan,
Have you tried:
same => n,Set(TIMEOUT(absolute)=3540)
You could override later if required.
Thanks,
Dan
On 17 Apr 2017 04:49, "Jonathan H" wrote:
The following setup prevents callers from going over 59 minutes:
--
On 24 July 2012 15:18, Hans Witvliet aster...@a-domani.nl wrote:
On Sun, 2012-06-03 at 23:23 -0400, Tom Browning wrote:
Any tips on solving the following performance conundrum:
Asterisk 1.8.12.2 running on HP DL360 G5 and G7s
tcpdump running to capture UDP 5060/SIP signaling to .pcap
How about...
exten = _X.,1,NoOp()
exten = _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})
exten = _X.,3,Set(NULL=${REALTIME(agents,device,${DEVICE})})
exten = _X.,4,Set(usernamepair=${CUT(NULL,\,,1)})
exten = _X.,5,Set(username=${CUT(usernamepair,=,2)})
exten = _X.,6,NoOp(DEVICE is ${DEVICE})
exten =
On 26 February 2010 12:38, Trevor Peirce tpei...@digitalcon.ca wrote:
Charles Wang wrote:
The sip.conf of MYE1 likes below:
[MYPBX]
type=peer
host=mypbx.abc.com http://mypbx.abc.com
nat=no
disallow=all
allow=g729
canreinvite=yes
qualify=no
context=default
Or, alternatively using the 's' priority...
exten = 845,1,Verbose(3, Incoming call from ${CALLERID(all)}) ; this will
be priority 1
exten = 845/12345678,n,Goto(blacklist) ; the n will make this priority 2
exten = 845/23456789,s,Goto(blacklist) ; the s will make this also priority
2
exten =
In the 1.6.1.* branch the line type=peer seems to be required on each
user...
d
2009/9/19 Örn Arnarson o...@arnarson.net
Sorry I wasn't more specific.
The error message is just the standard 'Can't find that extension'.
The problem is, however, that asterisk parses users.conf (and doesn't
Check your VICIdial logs and try to debug the VICIdial side of things... It
could be something along the lines of agent hits hangup, web interface goes
to add hangup command into the manager queue, fails due to a lock on the
table so call stays up... This wouldn't be an asterisk issue...
You need
2009/8/15 John Novack jnov...@stromberg-carlson.org
Received this on the console
-- IAX2/76.21.238.129:4569-4986 requested special control 20, passing it
to SIP/magicjack-08225a58
Did a Google search, but reached a dead end
Can anyone explain?
Something need to be changed in my
With that phone what you really probably want to do is just configure them
all with the same details...
i.e.
# Line 1 appearance
line1_name: incoming
line1_shortname: Incoming (Line1)
line1_authname: incoming
line1_password: password
# Line 2 appearance
line2_name: incoming
line2_shortname:
2009/8/11 Chuck Coleman p...@2cci.com
I have 6 Cisco 7940g phones and I would like to add them to my Asterisk
2.6.2 box. My SNOM 320 work just fine but I cannot get the Cisco’s to
register. They pull the TFTP P0S3-08-9-00 just fine. I change the NAT to *
no* but it still does not register.
2009/8/8 Dan Pilcheck pilch...@gmail.com
Hello all,
This is a VICIDial server and I am looking to send calls to VM box
2100 after 3 minutes of sitting in the queue(via the VICIDial AGI).
This would be inserted between exten = s,8,Background(open) and exten
= s,9,AGI.
From what voip-info
Perhaps it's only basic in certain parts of the world... I know I've never
experienced a voicemail system with such a feature...
I'm not saying having the option would be bad... but... I'd prefer voicemail
to get some more common, more requested of me, features first and that's
personally where
2009/8/4 Faheem faheem_...@yahoo.com
how to implement CLONED LINE Feature in asterisk
Hey, I want to implement Clone Line feature in asterisk. I am using
SPA-2100.
The feature should work in this way.
There are two ports in the SPA-2100 both are registered with asterisk with
same
2009/8/5 Mike asterisk-us...@norgie.net
On Tue, Aug 04, 2009 at 03:35:22PM -0500, Doug Bailey wrote:
This code is designed to handle Message Waiting Indication (MWI) incoming
on FXO
line. This data could very well be embedded in your CID spill as part of
an
MDMF message that also
point here is in chain_sip.c what are variables or structure that need
to maintain so that we can consider all registered users as active users.
Thanks!
Faheem
--- On *Wed, 8/5/09, D Tucny d...@tucny.com* wrote:
From: D Tucny d...@tucny.com
Subject: Re: [asterisk-users] how to implement
2009/7/31 pepesz76 pepes...@o2.pl
Dear All,
I'n trying to make a simple call forwarding, however I have small
problem when evaluating an expresion.
Here is my extensions.conf
...
; Unconditional Call Forward
exten = _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten =
2009/7/24 Stefan Schmidt s...@sil.at
Hello,
i´ve a question about the Meetme Options. How could i play a enter and
leave sound but without recording the user name first. I just want a
User joined conferenc and a user leaved.
With the i or I Option i have to record the name first.
Is
Looking at the code, asterisk doesn't know how to handle RPID in an INFO
message, so it just responds with an OK and goes on with it's business...
The fact that the message has the name of the called party, rather than the
calling party probably wouldn't help even if Asterisk did understand it...
2009/5/27 Atlanticnynex atlanticny...@gmail.com
(Accidentally posted this to asterisk-dev, should be here)
fgets is only returning one character... either when run as an AGI or
run as a test on PHP on CLI...
Example, enter , then fgets returns '3'.
It's not... There are problems with
2009/4/30 Justin Piszcz jpis...@lucidpixels.com
Hello,
I am using an SPA3102, all is working with asterisk 1.4, voice mail,
outbound calling etc, and it even passes the cid name/num to my analog
phone. However, when someone is calling me, I hear the beeps but the
caller-id information is
2009/4/22 Michael mich...@networkstuff.co.nz
I use GPL Ghostscript 8.6.2 to produce the TIFF files for faxing.
Does anyone know of a way, either while producing the file, or after, to
tell
how many pages have been produced? (without manually viewing the file)
tiffinfo? then count the
2009/4/22 michel freiha mich...@gmail.com
Hi all,
Does asterisk support the following scenario? I need when a customer who
own an endpoint registered on asterisk open the line, the asterisk will run
a specific AGI script inside the endpoint context?
You mean when they pick up the phone
2009/4/18 Tamer Higazi th9...@googlemail.com
Scenario:
I have a Asterisk PBX with a cologne chipset ISDN BRI card on it a DSP
cpu to take out the echo cancellation.
Communication is done through the chan_capi interface module.
If a call comes inside, and I forward it to the SIP account
2009/4/18 Tamer Higazi th9...@googlemail.com
D Tucny schrieb:
2009/4/18 Tamer Higazi th9...@googlemail.com
mailto:th9...@googlemail.com
Scenario:
I have a Asterisk PBX with a cologne chipset ISDN BRI card on it a
DSP
cpu to take out the echo cancellation
2009/4/19 Matthew Pounsett m...@conundrum.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi there. I've done some googling around to try and find an example
of what I'm trying to do, but it's one of those things that just seems
hard to find the right terms to search for. If there's
2009/4/19 Matthew Pounsett m...@conundrum.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 18-Apr-2009, at 17:17, John F. Ervin wrote:
Can't you handle that by defining an outbound route? set it to hit a
trunk or set of trunks when the correct dial pattern is detected?
Probably,
2009/4/15 Olivier oza-4...@myamail.com
May I ask : beside saving channels, what are the benefits of TBCT over
bridging calls inside Asterisk ?
I'm not aware of anything apart from saving channels...
What about caller ids ? I would say caller id should passed over to final
callee after
2009/4/15 John covici cov...@ccs.covici.com
Its not there and the link you gave me says its for sip originating
rather than calls to a sip channel.
on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote
It's been around awhile. I've used it in 1.4 Check out this link for
2009/4/14 jonas kellens jonas.kell...@telenet.be
[r...@asterisk asterisk]# netstat -a -n -p | grep 5060
udp0 0 0.0.0.0:50600.0.0.0:*
3047/asterisk
[r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060
tcpdump: verbose output suppressed, use -v or -vv for full
2009/4/8 Olivier oza-4...@myamail.com
2009/4/8 Alan Lord (News) alansli...@gmail.com
* DHCP Option 114 implemented.*
* DHCP Option 120 implemented.*
http://lists.digium.com/mailman/listinfo/asterisk-users
What does it imply ?
Provisionning from DHCP server ?
114 is for passing
2009/3/28 Jason Parker jpar...@digium.com
D Tucny wrote:
2009/3/26 John Morris aster...@zultron.com mailto:aster...@zultron.com
Hi, Axel.
Axel Thimm wrote:
How about merging in your changes/improvements/new packages with
ATrpms (and automatically later
I can't say it's always been like this, as I don't recall, but, Background
in 1.0 behaved like this, answering the channel if it wasn't already
answered and playing the sound file/s until they finished or an exten was
dialed...
in 1.0 the 'skip' option would cause playback to be skipped if the
2009/3/26 John Morris aster...@zultron.com
Hi, Axel.
Axel Thimm wrote:
How about merging in your changes/improvements/new packages with
ATrpms (and automatically later into rpmrepo.org)? That way we won't
have further fragmentation and a larger user base to test bits (which
will be
From your figures, it would appear that if you double the load you will be
potentially starting to see problems...
FYI, not sure if it's of use to you... but... The digium tc400b is a
transcoder card that can offload upto 120 channels of transcoding for g729
- ulaw... It's available as PCI only,
2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr
Hi, i managed to connect to Ericsson MD110 with PRI at last.
And made a successful call thru asterisk to ericsson.
But when i try to call from ericsson to asterisk i got an error on
asterisk side.
And i couldnt figure out why.
Here's my
2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr
Hi, i know i am asking a lot of questions lately in this forum..sorry
about that first of all. :)
Ok, here is the deal..
I am trying to make a hybrid system with an ericsson MD110 and asterisk.
Internally we have 4 digit phone extensions on
2009/3/20 D Tucny d...@tucny.com
2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr
Hi, i know i am asking a lot of questions lately in this forum..sorry
about that first of all. :)
Ok, here is the deal..
I am trying to make a hybrid system with an ericsson MD110 and asterisk.
Internally
2009/3/18 Kevin P. Fleming kpflem...@digium.com
Le'an Liu wrote:
My questions:
1. G.726 16/24/32/40 supported in asterisk-1.6.0.5?
No. Only G726-32 is supported in all Asterisk versions.
Perhaps the confusion in the voip-info page mentioned is due to the other
G726 rates being supported
2009/3/17 Marc Charbonneau timebandit...@gmail.com
I was looking at the aastra 9133i, however I was informed that this phone
is
no longer supported. What are good phones around the $100 - $125 price
point? (Need POE)
I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet,
2009/3/12 ssmax ss...@126.com
Hi all
i have just set up a asterisk in china, using DE410P and one E1 line
and get a phone number like: +86 020 87654321 from my sp
when somebody dial +86 020 87654321 , the asterisk will get the call in
number by ${EXTEN} variable, but it can only
2009/3/12 Danny Nicholas da...@debsinc.com
Greetings Listers,
I’m running 1.4.21.2 on SUSE 11.0 with and zaptel
1.4.12.1 on a TDM400P. Most of my calls work great, but occasionally we try
to connect to a customer or vendor external conference call and the call
A 2950 can be configured to limit the speed per port...
I guess the ISP here is operating this way because they are out of the way
and have limited bandwidth themselves, so, they are trying to split up the
bandwidth provided into smaller, more manageable chunks to avoid overloading
things at
The hotdesking section of the asterisk book may also be of interest...
d
2009/2/13 David Ruggles da...@safedatausa.com
Some googling lead me to this:
http://hans.fugal.net/blog/tag/astdb
Which looks like it has an answer.
Thanks all!
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
2009/2/13 Vikas topg...@gmail.com
My questions are:
1. The black wire coming into the Mc Manstel box is that a fibre optic
cable ?
2. What is the Mc Manstel box doing ?
3. What CISCO router do I need to buy to do bandwidth aggregation at my end
?
1) Yes
2) It's stopping you from poking
2009/2/9 邱磊 qiulei...@163.com
i reload the app_meetme.so in CLI:
- Reloading module 'app_meetme.so' (MeetMe conference bridge)
== Parsing 'etc/asterisk/meetme.conf': Found
All the sip message show that there is no fault, and i dont know why the
meetme application can't work.
i
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
These, the the pattern is ^\+[0-9]+
First I tried:
Set(pattern=^\+[0-9]+);
if (${REGEX(${pattern} ${${var}})})
but that
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
D Tucny schrieb:
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number
2009/2/4 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Tue, 3 Feb 2009, Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the
Please get this out of office reply disabled, or at the very least, fixed...
It currently seems to have generated a loop, sending out of office replies
to the out of office replies it's already sent to the asterisk-users mailing
list... It's bad that it sent a reply to the list anyway, but this
2009/2/2 Steve Underwood ste...@coppice.org
Bernd Felsche wrote:
Ian Cowley i...@moffat.co.uk wrote:
Beware PoE switches that can't handle Class 3 (15W) on all ports.
Most have fans because 24 (or 48) x 15W is hot!
That's the power supplied .. which'd be at the far end of the
2009/2/2 Daniel Harper dan...@harper.net.nz
I was wondering if anyone can help me with a problem we have at one of
our sites.
We have setup a Asterisk Trunk to a Avaya PBX, ie ...
Avaya - Asterisk (1.2.30) - External ISDN Network
BUT They also have a Polycom VSX 7000 that with some
Hi All,
There have been a number of comments recently about a shortage of
documentation on Asterisk, so I wanted to cover briefly the documentation
options available and suggest what they are useful for and how they can be
improved...
Documentation sources:
http://www.asteriskdocs.org/
- Not
This is still very off topic...
Someone's already suggested you look to somewhere for centos help... if you
had, you'd have found this...
http://www.centos.org/modules/newbb/viewtopic.php?forum=39topic_id=10098
Which is an RPM containing an update driver, so you wouldn't have to mess
about
on CentOS you can use lshal to return which driver is in use...
e.g.
lshal -s |grep pci_ |xargs -n1 lshal -l -u |grep -E
udi|info.product|info.linux.driver
get's a list of items, filters on pci, gets the long output for those pci
devices and outputs lines containing udi, info.product and
Perhaps this would help...
http://blog.michaelfmcnamara.com/2007/10/dhcp-options-voip/
Gives details on the dhcp option string needed for the phones and explains
that without it the phone will not accept a DHCP response...
d
2009/1/24 Joseph syscon...@gmail.com
Thanks for the input.
Yes, I
2009/1/22 Laurent Bonny laurent.bo...@gmail.com
Hello,
I am trying to connect an asterisk 1.6 to a trunking plate forme. With
asterisk 1.4.x I added to sip.conf a line asking for registration in the
form of:
register =
2009/1/23 Olivier oza-4...@myamail.com
Hi,
I need to locate a Asterisk server and a PoE-enabled switch on someone's
desk.
I've seen this Netgear ProSafe FS108P with 4 10/100 and 4 10/100 w/PoE
ports silent enough ?
Any recommendation ?
Specs on netgear.com say it's silent... 'Acoustic
2009/1/23 Lee, John (Sydney) john@compuware.com
There's nothing special about analogue phones in China, they are fully
interchangable with analogue phones elsewhere... Perhaps you have a
configuration problem, or, hardware problem on the Rhino Channel Bank,
perhaps the ports are
2009/1/23 Dean Collins d...@cognation.net
Looks like www.packet8.com has been hacked L
The phone service is offline as well.
Anyone else on this list using packet8?
Not using packet8, but, the website looks normal to me...
What are you seeing?
d
2009/1/21 Lee, John (Sydney) john@compuware.com
I am testing analog phone and fax machine plugged into Rhino Channel
Bank which is connected to TE412P card. This site is in China.
I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4
I ran into a problem which is
something wrong which I don't
know what it is.
Zeeshan
On Tue, Jan 20, 2009 at 1:58 AM, D Tucny d...@tucny.com wrote:
That's not my experience...
e.g.
SIP Phone show register
LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: REGISTERED
line APR state timer expires
2009/1/20 Olivier oza-4...@myamail.com
2009/1/20 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Tue, 20 Jan 2009, Olivier wrote:
GTalk seems to fill the bill of requirements, though, I don't think it's
available on Nokia mobile phones ..
My Nokia
2009/1/20 Olivier oza-4...@myamail.com
One thing to note about fring, the device establishes a connection using
fring's proprietary protocols to fring servers, fring then establishes SIP
connections from those servers... So, even if connected to the office Wifi
connection, you could
If your provider provides any signalling to indicate answer, such as a
polarity reversal, this could be detected easily...
; Use a polarity reversal to mark when a outgoing call is answered by the
; remote party.
;
;answeronpolarityswitch=yes
This isn't very common though... alternatively, there
2009/1/20 Olivier oza-4...@myamail.com
Hi,
Is anyone using Fring as a SIP client to an Asterisk server ?
Yes, testing it...
A prospective customer of mine is asking to integrate its iphones with an
Asterisk server and after googling, I still have some unanswered questions :
1. Which
2009/1/20 Zeeshan Zakaria zisha...@gmail.com
Hi everyone,
I googled this followed the instructions, but it hasn't work for me yet.
I have universal setting in SIPDefault.cnf and phone specific settings in
SIPXX.cnf. But it doesn't get registered.
I need to register it on two
2009/1/20 John Todd jt...@digium.com
On Jan 19, 2009, at 6:29 PM, D Tucny wrote:
2009/1/20 Olivier oza-4...@myamail.com
Hi,
Is anyone using Fring as a SIP client to an Asterisk server ?
Yes, testing it...
A prospective customer of mine is asking to integrate its iphones
2009/1/19 Thomas Stein thomas.st...@knowledgetools.de
Hello.
Does someone know what order field means in followme.conf? The Doku says:
number= number to call[2nd #[3rd #]] [, timeout value in seconds [,
order in follow-me] ]
So an example would be:
number= 123124125,10,?
It would be
register to the second box, only to the first
one. Why? god knows...
__Yehavi:
2009/1/20 D Tucny d...@tucny.com
2009/1/20 Zeeshan Zakaria zisha...@gmail.com
Hi everyone,
I googled this followed the instructions, but it hasn't work for me yet.
I have universal
2009/1/17 Carlos Chavez cur...@telecomabmex.com
I have this call:
SIP/protel-525512047 default 90445528885371 1 Ringing
AppDial (Outgoing Line) 90445528885371 264:24:2
(None)
I cannot use the soft hangup commando from the CLI because I do not
70 matches
Mail list logo