On 12/02/11 04:02, Bryant Zimmerman wrote:
I am running 1.8.3 and my BLF lights have stopped working. The hints
appear to be intact when I use core show hints. But none of the phones
are getting the BLF updates. This has happend in the past and I have
had to restart my server. What could be
On 09/12/10 07:06, Zeeshan Zakaria wrote:
I think this may be because ...
So you think, don't know. Maybe you knew if you knew the FreePBX
code, or bothered to look into it.
For God's sake, stick a sock in it. Others are attempting to help. You
are not.
--
I don't know exactly what help you expect to receive in this forum.
Asterisk itself has nothing to do with VPNs of any kind, and you should
take your questions regarding the setup and configuration of them to the
appropriate place.
On 09/09/10 18:26, Deepika Nijhawan wrote:
I am not
On 08/11/10 18:46, Tino wrote:
Thanks Gareth for your quick reply.
It is the lateset version and i think i need access to Dahdi
interface. Is there any disadvantages other than this.
If you need access to cards installed in the machine, you can forget
running Asterisk under VMware. VMware
On 06/19/10 15:19, Kamonwat Sookkara wrote:
Dear Asterisk friends,
Please help me to clarify my doubt. After monitor SIP and RTP
traffic with Wireshark. I found that both SIP and RTP traffic between
2 sip clients must be passed through Asterisk.
Is it possible that 2 sip clients
On 05/21/10 09:07, Leif Madsen wrote:
Danny Nicholas wrote:
If I'm going to bother with 1.6.2, I'll wait a few months for 1.8. But in
the spirit of your question:
(1) dialplan conversion
(2) loss of functions like Gosub
Can you be more specific about what 1) and 2) mean?
Exactly what is the problem you've having with CallerID? Are you not
receiving it, or are you not able to send it? Which carrier are you
using and what make and model card is the line connected to?
For incoming calls on ISDN-10/20/30 lines, no special configuration is
required to receive caller
), W:0, C:1, A:0, SL:0.0% within 0s
Members:
Local/1...@agents (dynamic) (Not in use) has taken no calls yet
No Callers
I am completly lost. :(
On Sun, 14 Mar 2010 01:08:53 +1100, Rob Hillis wrote:
Your best option is likely to be to create a separate
Your best option is likely to be to create a separate context that calls
both numbers, like so...
[agents]
exten = 1,Dial(SIP/0317998975SIP/0317998985)
...then add Local/1...@agents to the queue.
On 03/14/10 00:03, Magnus Benngård wrote:
Hi!
We have alot of users who are having 2 phones, 1
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
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 02/15/10 20:00, Randy R wrote:
Olle, this may be a stupid question, but shouldn't a native santitize
function be urgently added to the code base in all versions or change
the dialplan compîler to ignore dangerous characters?
Whilst I agree with this, the unfortunate attitude we seem to
I think he's referring to the fact that you seem to be looking to put
together the telephone equivalent of a spam service.
I'd be advising rm -rf / as well.
On 02/06/10 16:19, Thomas Perron wrote:
karl,
does it make you feel good ?
wow. pathetic.
On Fri, Feb 5, 2010 at 11:00 PM, Karl
On 01/16/10 04:27, Bruce Nik wrote:
Hi Guys,
Other than than yum repository (which fails when installing freepbx
with it) are there any automated install scripts out there that would
install Asterisk 1.6 or 1.4 onto a CentOS LAMP system?
If the script install FreePBX that would be a BONUS.
On 01/17/10 01:15, Tzafrir Cohen wrote:
Try PBX-in-a-Flash. Undoubtedly it won't do everything you want out of
the box, but I suspect it will do /most/ of what you want out of the box.
But will not let you debug that install script. I tend to distrust
running such a hidden script
On 01/15/10 17:54, randall wrote:
hi all,
i noticed that a lot of VOIP phones have a double network interface
allowing you to use only 1 LAN cable for both the phone and your
desktop, a really nice feature that can save a lot of cable, but most
are 10/100 connections while i have a
On 01/10/10 05:03, --[ UxBoD ]-- wrote:
Hi,
I use VoIPTalk as my provider and unsure of a minor issue. When people call
me they get a US ring tone instead of UK. Is this a Asterisk configuration
issue or one for VoIPTalk ? I am running 1.6.2.0 and IAX2 trunks.
This is almost always
On 12/23/09 12:23, Russell Bryant wrote:
Wasn't this scheduled for 1.6.2?
I don't believe so, but I could be mistaken. It's certainly not in 1.6.2
as best I can tell from looking over the source code :-)
Nope, it's not in 1.6.2. It went into trunk after the 1.6.2 feature freeze.
Leif Neland wrote:
I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B)
return busy when just one extension is busy.
Forgive me for the question, but /why/ do you want this behaviour?
Isn't the whole point of dialling multiple extensions so that a call has
a greater chance of
Indeed it does. You add contacts and set the softphone number to
extension@server
Leif Neland wrote:
Philipp Kempgen skrev:
Leif Neland schrieb:
Mostly to debug/test BLF, is there a softphone or another app. which can
subscribe to hints on Asterisk?
X-Lite?
Peter Evans wrote:
On Tue, Nov 11, 2008 at 02:35:19PM +0800, 邱磊 wrote:
hii guys:
i get the message from the asterisk:
Started music on hold, class 'default', on
Local/s...@skype-web-callback-dial-263to263-1775,1
[2008-11-11 14:32:41] WARNING[1781]:
Tilghman Lesher wrote:
Regardless of how you think it should work, the poster above described
precisely the way it works. If your end boundary is 12:00, it will evaluate
as true all the way up until 12:01:59. If you don't want that, another poster
has suggested using 11:59, which will work
Chris Bagnall wrote:
First things first. You are running /very/ old versions of firmware -
particularly on the 300 and 320. Upgrade them. I've been running
7.3.14 for some time without a problem, though it appears that 7.3.23 is
now out.
I concur about upgrading the software, but
oi geli wrote:
Hi,
I am using SNOM phones with Asterisks for few years. They used go occasionaly
NR, and I would reregister them from the phone web interface. But it started
doing frequently for last few months.
Here are SNOM Phone and the firmware version;
snom190-SIP - Version-Code:
800ms is horrendous lag for a VoIP connection. If I were you, I'd be
investing some time in finding out why the lag is so great. Even if I
do a ping to a UK address, I'm getting pings of no more than 300ms from
Australia. Unless you've got multiple satellite connections in the path
(in which
Low bandwidth is another possibility, but I'd have though that any
connection slow enough to generate that much latency wouldn't be usable
for VoIP in the first place.
Ishfaq Malik wrote:
Cheers Rob, I was thinking it was due to a low bandwidth connection at
the other end but from what you're
Alex Samad wrote:
Voltage isn't the issue - the difference is in the impedance. Australia
I get this in my dmesg when I load up the rdm410 modules
[1083334.103487] Freed a Wildcard
[1083336.171371] ALAW override parameter detected. Device will be
operating in ALAW
[1083338.040522]
Yes, although not for connecting to the PSTN - I've used one for
connecting to a legacy NEC PABX.
Voltage isn't the issue - the difference is in the impedance. Australia
uses complex impedance (220+820Ohm resistors with a 120nF capacitor)
whereas the US uses a straight resistor.
Alex Samad
Bart Coninckx wrote:
Hi,
I'm using a ISDN-30 E1 line from KPN Belgium.
The challenge is to get a correct CallerID on outgoing lines.
When I put this in my dialplan:
exten = _0.,1,Set(TEMPVAR=${CALLERID(num):1})
exten = _0.,2,Set(CALLERID(num)=144622${TEMPVAR})
exten =
jonas kellens wrote:
Do you understand what is happening ?
I don't understand what this sentence means :
SIP/3starsnet-08d70ea8 is making progress passing it to
SIP/twinkle-08de0490
Pretty simple really. Your SIP trunk 3starsnet is making progress with
the call and Asterisk is passing that
Dan Pilcheck wrote:
The call will go over the server fine, but when the Call Center server
answer, the CLI returns:
NOTICE[4296]: chan_iax2.c:7398 socket_read: Rejected connect attempt
from 10.0.10.20, request '2...@2xxx' does not exist
What context are the phones in the extension range
Cary Fitch wrote:
We have a bunch of SNOM 360’s we are not using. I agree they are not
intuitive to the user. They work ok in general. I would part with 15
or so at an attractive price, one or more,
I like the Grandstream 2000 series. Easy to use, easy to set up, good
web page.
Christian Stredicke wrote:
Check out the snom 300 or the snom 820...
Good lord... talk about two extremes... :) The Snom 300 is pretty good,
but the 320 is much better and costs around a *third* of what the Snom
820 does.
Stick with the older model snoms. So far I've seen nothing about
Alex Samad wrote:
I have been looking at a snom 300, which seems okay. the display goes a
bit haywire occasionally - not sure why yet.
Are the 320 worth the extra money ?
IMO yes, though it really depends on what you want from the phone.
The Snom 320s handle transfers considerably better
Jeff LaCoursiere wrote:
We are still talking about a $175 phone. How about the Polycom IP 320?
$85 at 888voipstore. Can't go wrong with Polycom for voice quality.
True, Polycom's are brilliant for voice quality, but unlike the Snom, a
Polycom /will/ reboot on the drop of a hat /and/ take
The clue in the log is no authority found. Something in the
configuration at the other end doesn't match the configuration at this
end - almost certainly the username and password.
Why are you including the IP address when dialling the trunk? If your
peers are set up with IP addresses (which
Sounds like you're looking at the wrong variable. You should be looking
at CALLERID(num).
peace keeper wrote:
Hi there,
I am using the Asterisk as the PBX, and need to know the caller ID for
the incoming call,
but when I show the caller Id, it gives the Zaptel channel that
recieves the
Louis-David Mitterrand wrote:
Hi,
Is anyone here using OrderlyStats with asterisk in a call center
setting? If so what what is your experience with it? Is that software
really free for asterisk users?
Or is there a better option out there?
The short answer is OrderlyStats isn't really free
carl Lougher wrote:
Ok cheers.
Any idea when 1.6 goes stable for prod?
Theoretically it already has, however as was the case with 1.4, I
suggest you tread very carefully when it comes to migrating to 1.6.
___
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Sebastian wrote:
Anyone thought about something like outgoing queues?
Many people have. I know QueueMetrics has methods for this kind of
thing, and I'm fairly sure that Vicidial does as well.
___
-- Bandwidth and Colocation Provided by
Darrick Hartman wrote:
Rob Hillis wrote:
Daily? No. However, after implementing a weekly restart of Asterisk,
I've found the instance of lockups and CPU utilisation spikes have
decreased significantly.
Unless you're using some unstable modules, there really should be no
need
Darrick Hartman wrote:
If I were to do things again, I'd be running Astlinux on a net 5501 with
an integrated hard drive (for voicemail/IVR and so on) Only time I've
ever had to reboot my Astlinux box at home (on an ALIX-3) is when it's
time to upgrade Astlinux.
That's what we like to
Kurian Thayil wrote:
On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:
Daily Asterisk restart
Do you think its mandatory in production env?
Daily? No. However, after implementing a weekly restart of Asterisk,
I've found the instance of lockups and CPU utilisation
Benny Amorsen wrote:
Michael mich...@networkstuff.co.nz writes:
pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in
extension 'PHONE NUMBER' in context 'phones'
[Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto:
Priority 'outgoing|PHONE NUMBER'
Michael wrote:
pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in
extension 'PHONE NUMBER' in context 'phones'
[Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto:
Priority 'outgoing|PHONE NUMBER' must be a number 0, or valid label
PHONE NUMBER = the
Sebastian Milioto wrote:
2. What E1 card should I buy for Asterisk? Is the physical interface
(conectors) E1 identical as T1?
The connectors are identical, however the protocol isn't. However, just
about all the T1 cards I'm aware of support E1 as well - usually
selected by a jumper on the
Michael wrote:
On Fri, 03 Apr 2009 12:32:03 you wrote:
Like:
exten =
5226001454,1,Dial(SIP/3615221401SIP/3615221402SIP/3615221407SIP/52260014
0 5,20)
That is what I am currently doing - though is there a cleaner way?
The only cleaner way is to define the group in [globals]
Loic Didelot wrote:
Hi,
I am trying to connect a doorbell to a Xorcom device. And the setup is
quite simple. But when I push the doorbell all I see on the asterisk cli
is:
-- Starting simple switch on 'Zap/11-1'
[Apr 2 13:00:40] DEBUG[8771]: chan_dahdi.c:6180 ss_thread: not enough
digits
Paul Hales wrote:
I would love to see the agent login/logout stuff working - but that's
just me.
I'd like to see the damn web interface become usable.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Rilawich Ango wrote:
My configuration is simple as below.
SIP phone - asterisk - CISCO - T1
Do you mean the hum noise is created by electric-magnetic field?
Asterisk can do nothing to eliminate it?
That would be my bet. No, Asterisk can't do anything to remove EM
noise. That's up to
Anthony Plack wrote:
Hey all,
I have a potential project which calls for a very small form-factor computer
like this:
http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp
However, I am needing an FXS port integrated into a small footprint computer.
About two and a half years ago, I upgraded a small call centre from
corded handsets to X-Lite with Plantronics CS60 USB headsets.
X-Lite lasted about two or three months before we ditched it in favour
of Eyebeam. X-Lite disables too many features to be useful. With the
Plantronics headset,
Yes. Grandstreams suck.
Oguzhan Kayhan wrote:
Hello,
I configured both asterisk and grandstream 2000 accourding to howtos on
the web..
And everything seems working fin.
But if i reload asterisk grandstream stops working with BLF.
I need to restart the phone to enable BLF again.
Any
Asterisk wrote:
You can simply:
exten = _0.,n,Goto(${DIALSTATUS})
(before the playback)
Use the labels as the destinations - eg.
exten = _0.,n(BUSY),Noop()
exten = _0.,n(CONGESTION),Noop()
I've never seen that before, does that definitely work in 1.4.x? If so,
cool...
Duncan Turnbull wrote:
Hi All
I am looking at a replacement for a hotel PBX which requires at least 60
analogue extensions.
I tend to use Sangoma equipment but haven't tried this many analogue
extensions before. I am interested in anyone's experience of which
server platform literally
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 (Spandsp) with a IAX2 trunk (ethernet connection) to
another asterisk (zap).
client-asterisk (Spandsp)-asterisk (zap)-fax
To quote the
Which line of code is generating this log entry?
[Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing
[91...@from-pstn:3] Goto(Zap/31-1, s|a2) in new stack
...because this appears to be where your problem lies.
joek...@gmail.com wrote:
Hi all,
I have a connect between a siemens hipath
...except that Macros are now deprecated and will most likely be removed
in 1.8.
Robert Broyles wrote:
Hmm, this is all very interesting.
Looks like using a Macro and the 'M' Dial() option is the way to go for
now if you need the answer confirmation.
Administrator TOOTAI wrote:
[MyPeer]
host=xxx.xxx.xxx.139
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142
permit=yyy.yyy.yyy.yyy/255.255.255.255
On incoming calls, when the peer address is the one terminating with
.139 everything is OK.
If I
Michael wrote:
Change it to the following:
exten = _10,1,Dial(SIP/10,10)
exten =_10,n,Background(vm-nobodyavail)
exten = _11,1,Dial(SIP/11,5)
exten =_11,n,Background(vm-nobodyavail)
The only time I am aware of that you can leave out the prefix underscore is
for exten = s and exten = i
No,
forums - sigma wrote:
having deployed a fair amount of phones I have the following observation
(and these observations are worth what you paid for them :-) )
1. Linksys 942, my preferred mainstream desk phone, a bit more expensive
than the Polycom IP330. Be careful as there are two SKUs
Michael wrote:
My experience with Grandstream is that are one of the better 'cheap' ones,
but
cheap non the less.
I am yet to run into a worse IP phone than the Grandstreams - although
having said that, I should say that I've always steered clear of most of
the Chinese no-name brand phones.
Michael wrote:
I bought it. The SPA962 went on ebay within 3 months of me buying it.
I have a few grandstream 286's I like to use for traveling and placing
in remote areas of an installation.
3 months... that long?
Again I'm surprised. I've had no problems at all with the Linksys
dubravko caric wrote:
Hi all,
I'm testing Linksys SPA922 phone and I have strange issue. when call
is finished on the phone I see CallEnded and normal silence for cca.
5 seconds and then I get fast busy for cca. 20 sec. So, this isn't
automatic hangup as on other phones I have tried
David fire wrote:
hi
i need an open source callcenter manager system like queuemetrics but
opensource any one know any?
i prefer to search before start a new one
You'll be pushing to find something even close to QueueMetrics' quality
available in open source. The closest I'm aware of is
Philipp Kempgen wrote:
SendImage() in 1.4:
---cut---
SendImage(filename): Sends an image on a channel.
If the channel supports image transport but the image send
fails, the channel will be hung up. Otherwise, the dialplan
continues execution.
The option string may contain the following
Carlos Chavez wrote:
I have a new customer that wants to upgrade their Asterisk installation
from 1.2.27 to 1.4.22. They use FreePBX for administration. Since
there are many syntax and command changes from those versions of
Asterisk, is there an easy way to convert the FreePBX
Alex Balashov wrote:
The solution for the problem of an IAX client is a SIP client.
That's not a particularly good solution if you have a NAT between your
client and Asterisk. IAX is still *much* easier to get working through
a firewall.
___
--
Russell Bryant wrote:
On Nov 8, 2008, at 1:30 PM, Atis Lezdins wrote:
Asterisk offers very much the same flexibility. You can disable
specific log levels (for example warnings) in logger.conf or you can
log everything to syslog, where filter out this specific message.
Of course,
Tzafrir Cohen wrote:
On Sat, Nov 08, 2008 at 02:33:18PM +1100, Rob Hillis wrote:
Maybe it's me, but I think that warning should be regarding a problem
I can fix. Malformed network content does not neceserily fall under that
definition. notice?
Absolutely it does. Warnings
Louis-David Mitterrand wrote:
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Your monitoring app is not sending valid IAX2 packets to the server. If
it was sending a true IAX2 POKE, it would be a valid packet and wouldn't
generate this warning.
Could asterisk at
Tzafrir Cohen wrote:
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
I'd take this warning seriously. It means that your monitoring app isn't
monitoring what you think it is.
I always want to know when I get malformed protocol packets in. It is
always bad news, mostly either
Steve Anness wrote:
Good Day,
I have been tasked with fixing the time on our asterisk server. I am
having a hard time finding documentation to tell my what asterisk uses
to get its time information to push to phones (or a better question,
where does the SPA-962 get its time information)?
Hales wrote:
It should ignore the keywords, but you will get lots of errors in the CLI.
My guess is that if you put it all in a DB (and use realtime) you can
probably do whatever you want.
PaulH
Rob Hillis wrote:
Hi guys,
I'm about to embark on a small (undoubtedly to get much
Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rob Hillis wrote:
Unfortunately RealTime isn't going to be an option - it's another level
of configuration I want to avoid, but more importantly since I'm
planning on being able to run these scripts on an Astlinux
Hi guys,
I'm about to embark on a small (undoubtedly to get much larger) project
to write a set of scripts to handle provisioning of phones - Snom to
begin with, possibly with others (most likely Polycom and Linksys) to
follow later. Since I want this script to handle *all* aspects of phone
Emmanuel Pascal Bruno wrote:
I have turned off firewall on the linux box, I have turned off
firewall on the router I still have the same problem :-(
Disabling firewalls is almost certainly going to ensure the problem
persists. You need to ensure that all SIP and RTP ports are
port-forwarded
Emmanuel Pascal Bruno wrote:
I have a DID from IPKall.com which is forwarded to my asterisk box.
Then this extension should call my ip phone using Dial application.
Everything works fine, except when I pickup the phone, I can talk, the
other party can hear me, but I cannot hear anything the
Peter Galiovsky wrote:
Does anyone have any idea what should I try next?
Either contact Digium support directly or the people you bought the G729
license from. You're more likely to get the assistance you need in a
shorter period from these people than this list.
Kev Szaszvari wrote:
Hi there
Our company is using the Linksys SPA-942 Phones, and they are pretty
useless.
They dont have any central management or provisioning, as well as a
pretty bad interface.
This is completely incorrect. Linksys SPA-942s *do* have the ability
for central
Tim Panton wrote:
Does anybody know any free WebCall solution to let our customer call
us directly via our web site?
Any clue will be welcomed.
Yep, take a look at our offering on www.phonefromhere.com
A per-minute charge does not constitute a free solution. Please read
requests
Eric Chamberlain wrote:
I should have clarified, we're only making outbound calls, not
inbound, so there is no registration.
Is there a particular reason you /can't/ register? It would seem that
registration would provide the functionality you require, even if you're
only making
Eric Chamberlain wrote:
Is there a particular reason you /can't/ register? It would seem that
registration would provide the functionality you require, even if
you're
only making outbound calls.
In the case of a server like Asterisk, wouldn't sending a register
disrupt the flow of
Tilghman Lesher wrote:
Can someone suggest the best way to deal with this without resoring to a
highly repetitive/iterative dialplan?
Leif and I discussed something like this at Astricon 2008, and we came up with
this patch:
http://bugs.digium.com/view.php?id=13632
Nice! For those of
Wesley Haut wrote:
Yell at me if you will, but I hate func_realtime - it's not very
usable nor is it change-friendly (update your database and your
dialplan completely breaks).
I agree completely. As it stands, the REALTIME() function is nearly
completely useless. If Asterisk had better
Tilghman Lesher wrote:
On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote:
Wesley Haut wrote:
Yell at me if you will, but I hate func_realtime - it's not very
usable nor is it change-friendly (update your database and your
dialplan completely breaks).
I agree completely
Olivier wrote:
2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hi,
When dialing a number, I use :
exten = _123X, 1, Dial (SIP/${EXTEN})
Then, I get TRYING and RINGING SIP messages which both include
this kind of line :
To: sip [EMAIL PROTECTED]
Babcock, Michael Alex wrote:
windows smart phone v 6.0 example
htc shadow
is what i have. It has wifi abilitys.
Googling for windows mobile sip yeilds a multitude of results. I'm
sure one of them will point you in the right direction.
___
--
Josiah Bryan wrote:
The script design supports plugin formatting as it stands. E.g. I can
insert any formatting algorithm if anyone has any suggestions. Right
now, the formatter script just does:
#!/usr/bin/perl
use strict;
my $file = $ARGV[0];
print ~pp~\n;
print `cat $file`;
print
Giorgio Incantalupo wrote:
Hi,
why do not you simply delete them from zapata.conf and restart your PBX?
Because that simply doesn't acheive what he's wanting to achieve. On
PRI circuits you can dynamically enable and disable circuits at the
data-link level. Whether this can be achieved with
Joseph wrote:
I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo portage
but I think this version has a problem with RFC2833 DTMF signaling and I
don't think there
will be any newer version available anytime soon on portage.
I need stable version, I'm using Asterisk
Olivier wrote:
Hi,
I'm receiving this :
[Aug 28 02:25:16] NOTICE[5895] chan_sip.c: Received SIP subscribe for
peer without mailbox: 9163
I've read this :
http://lists.digium.com/pipermail/asterisk-users/2008-May/211701.html
I typed this:
asterisk -rx reload
asterisk -rx voicemail show
Olivier wrote:
Now that root cause is found, would you say that warnings or CLI
should have been different ?
Obviously, MWI subscriptions must come from SIP hardphones (at least
those supporting MWI feature).
So in this case, Received SIP subscribe for peer without mailbox:
9163 rather
VoIP Cyprus wrote:
Can you share with me your experiences with Asterisk 1.6? Is it stable
enough for commercial service?
No. No matter how good some people may tell you it is, 1.6 is still
beta software and software is rarely beta for no good reason. Don't
even THINK about running 1.6
Tilghman Lesher wrote:
Given that this is the case, we may want to do one of the following:
a) document that qualify=yes is incompatible with realtime, unless
rtcachefriends is turned on, b) automatically disallow qualify=yes if the
peer is realtime and caching is not turned on, or c)
Jakub Arkon Syrek wrote:
Hello, we have strange problem, till now everything was working fine,
there where no problems with dial and answer calls.
Yesterday our system crashed and we notice strange behavior.
What type of event caused the box to crash? Given the fact that you've
also
Olivier wrote:
Hi,
Though this is a bit off-topic on this list, I think this might
interest those looking to build Asterisk appliances out of mini-ITX
boards such as http://www.pcengines.ch/alix1c.htm.
If you're interested in the ALIX type boards, there is a reseller in
Australia that
Ken Williams wrote:
We're entertaining moving our intranet to Hughes satelite for our
remote locations. I'm curious if anyone with Asterisk servers has
used satellite, and if so, is the latency an issue. My understanding
is that you immediately introduce 250ms latency for travel time up
Dan Peters wrote:
We have had Asterisk up and running for a while now and it works very
well. Recently we tried to integrate a Linsys SPA962 with the
associated SPA932 console. We can get the BLF lights to blink when a
phone is ringing and we can get the BLF lights to go solid when that
If a phone is unplugged, it's not likely to have time to send
notification of this to Asterisk before it powers off. There's nothing
you can add to your dialplan to overcome this, however you *can* set the
qualify parameter within sip.conf (or it's equivalent realtime table)
to overcome this.
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