Astmanproxy is just a proxy. It it just taking the load off asterisk
for multiplexing multiple Asterisk manager connections, but it does
not change the protocol (except to add a couple of features) unless
you select one of the non "standard" plugins.
Regards,
Steve
On 12/19/06, Daniel Gradecak <
Check your DNS config. That is almost always the answer to a 10 second delay :)
Steve
On 12/29/06, ast guy <[EMAIL PROTECTED]> wrote:
Hi,
I'm running Asterisk 1.2.10 on gentoo linux and facing strange kind of issue.
1. chan_sip.so takes about 10 secs to load up when asterisk starts.
2. When I
Hi,
On 1/9/07, J. Oquendo <[EMAIL PROTECTED]> wrote:
Has anyone got this annoying sidecar to illuminate when users are on the
phone?
Yup, works fine.
I've tried Context: Line, still no dice. In extensions.conf I have:
exten => 4000,hint,SIP/4000,name
Make sure that the "hint" is not the f
How about:
Action: originate
Channel: Local/[EMAIL PROTECTED]
etc
Then in extensions.conf
[indirect]
exten => _X.,1,NoOp("Click to Call")
exten => _X.,n,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten => _X.,n,Goto(from-internal,${EXTEN},1)
Get the idea? Does that help?
Cheers,
Steve
On 1/12
I think that the support for the SNOM button (which uses a SIP message
to request the recording) is part of the bristuff patch, otherwise
only *1 will work.
http://www.junghanns.net/downloads/
Cheers,
Steve
On 1/12/07, Ron McCarthy <[EMAIL PROTECTED]> wrote:
Hi,
This does nothing for me at al
I agree with C F - We just upgraded to our first non-internal 1.2.x
system last Friday. Mostly I am glad we waited. I imagine we may
upgrade to 1.4 in about a year :)
Really it depends on your customer. If it is a commercial operation I
would be cautious of 1.4 still, and at the very least test i
Hi,
I have not checked this, but I thought the intention was that 'show'
was a human readable formatted output, and 'list' was meant to be the
same data but more easily machine readable.
Of course I could be completely wrong.
Steve
On 1/13/07, Jerry Geis <[EMAIL PROTECTED]> wrote:
I thought I
I have not confirmed this independently, but I believe this is fixed
if you disable the "Show message light when a call is missed" feature
in the phone config. Alternatively, try pressing "X" to clear the
missed call indication before pressing "Retrieve"
Might work... Might not :)
Steve
On 1/17/
On 1/25/07, Peter Mitchell <[EMAIL PROTECTED]> wrote:
Has anyone seen this issue with asterisk running like a dog when the
internet is down ? Internal calls, incoming ISDN calls etc all seem to be
affected. There is a local DNS server that is always available so I'm not
sure why asterisk is so
On 1/26/07, Jon Schøpzinsky <[EMAIL PROTECTED]> wrote:
Hello List
I am having a rather big problem with a sangoma A104 card, I just installed
to replace a Digium TE410 card, that was acting up.
But now we have a problem with the sangoma card. It runs great after being
started, and calls procee
Which asterisk versions etc etc?
On 1/26/07, Jon Schøpzinsky <[EMAIL PROTECTED]> wrote:
I am running the newest version, from the sangoma wiki.
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: 26. januar 2007 10:56
To: As
I would be interested to know whether this
http://bugs.digium.com/view.php?id=8376
patch makes any difference. The problem is almost certainly not caused
by Centos (which is widely used with Asterisk) or EPIA (which I use
lots).
Regards,
Steve
On 1/29/07, Erick Perez <[EMAIL PROTECTED]> wrote
On 1/29/07, Dominik Zalewski <[EMAIL PROTECTED]> wrote:
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
applic
On 1/29/07, Dominik Zalewski <[EMAIL PROTECTED]> wrote:
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
applic
s says that in SVN 51223 it was implemented, im running
51363. However I may be wrong. I will apply that patch and let you
know.
Thanks for the pointer.
should I leave asterisk as -march=i586? or 386?
On 1/29/07, Steve Davies <[EMAIL PROTECTED]> wrote:
> I would be interested to kn
*ping*
I am interested in this too if anyone has any clues? I am looking to
do this on a Cisco 7941/7961.
Thanks,
Steve
On 1/26/07, Naija Man <[EMAIL PROTECTED]> wrote:
Hello,
We have an asterisk system with about 40 cisco 7940/7960 phones and a few
linksys SPA941. I recently analyzed our net
Hi,
I have a problem understanding which 'h' (hangup) extension is used in
which case - It seems to vary depending on channel type. Assuming the
following simplified dialplan:
[macro-faxhere]
exten => s,1,rxfax(file)
exten => h,1,NoOp(Hangup in macro)
[fax]
exten => _X.,1,Macro(faxhere)
exten
On 2/6/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
Steve Davies wrote:
> I have a problem understanding which 'h' (hangup) extension is used in
> which case - It seems to vary depending on channel type.
It doesn't. It depends on which side of the call hang
On 2/12/07, Rob Schall <[EMAIL PROTECTED]> wrote:
From what I read on the voip-info page, they did not define how to have
the phone muted when a paging call would go out. I already have the
paging (without mute) working using those same headers from that site.
But they don't cover the issue I'm t
On 2/12/07, Radu Padure <[EMAIL PROTECTED]> wrote:
I recommend you to use Sangoma A102D or A104D.
I agree, though if you are on a budget, the A101 + software echo
cancellation is pretty functional these days.
Cheers,
Steve.
___
--Bandwidth and Coloc
On 2/12/07, Rob Schall <[EMAIL PROTECTED]> wrote:
Steve,
I posed a similar question to Shane, but maybe you'll know as well..
I was able to get app_page to work. So when I call... **8050, it auto
answers and the callee is muted. However, what if that person wants to
"answer" the page and pickup
On 2/15/07, Dominik Zalewski <[EMAIL PROTECTED]> wrote:
Hi All,
I'm using asterisk 1.2.15 and call forwarding doesnt work for me.
from my extensions.conf:
; Unconditional Call Forward
exten => _*21*X.,1,NoCDR
exten => _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
exten => _*21*X.,3,Playb
Hi,
In older versions of asterisk I used to be able to use
"incominglimit=1" to effectively disable call waiting on a specific
SIP channel (Where broken phones do not allow this on the handset
itself)
In 1.2.x this became "call-limit=1", but this prevents the phone from
opening a 2nd line in ord
On 2/24/07, Pavel Jezek <[EMAIL PROTECTED]> wrote:
Brian Capouch wrote:
>
> But the included comments say, "The user part of a type=friend call
> will still be affected by the call limit"
>
> Those seem to be in conflict, but perhaps it's just my parser :-)
> Could someone clueful explain?
>
>
On 2/27/07, Steve Davies <[EMAIL PROTECTED]> wrote:
Thanks for all of the pointers on this - I think merging the
"limitonpeers" change from trunk into 1.2.15 is my favourite option
right now. Or should I just take chan_sip.c from trunk? Would that be
fairly safe?
Err... What I
Hi,
Could someone double-check a behaviour I am seeing in 1.2 SVN HEAD
In sip.conf, create a type=friend entry with call-limit=1
1) Place an outbound call from the device
2) Place a call in to the device
"sip show inuse" is now something like:
* User name In use Limit
Hi,
An observation on this feature, which I may have completely
misunderstood, so flame away if I am being dumb :)
Looking at the code, setting "limitonpeers=yes" causes all user and
peer calls to be ref-counted as if they are peer calls (assuming a
user and peer of the same name exist).
A side
On 5/3/06, Dr. Michael J. Chudobiak <[EMAIL PROTECTED]> wrote:
One of my users reports frequently hearing echo on her Snom 360 phone,
even while talking to other Snom phones (via Asterisk) on the same LAN
(i.e., all-digital low-latency connection). I can never reproduce it
though, and swapping t
On 5/10/06, Joseph Rothstein <[EMAIL PROTECTED]> wrote:
From what I have tested, using cisco phones and 1.2.5, the original callerID
is not kept when making a transfer.
Any other ideas?
We use SPA, snom and aastra phones, and I had assumed that this was a
limitation of the SIP protocol. I woul
I don't know which version you downloaded, but if you can get the
source from CVS on Sourceforge, and build it yourself, you may have
more luck - The CVS version has code contributed from several sources,
and is slightly better that the packaged version.
Cheers,
Steve
On 5/12/06, Tomislav Vojvod
On 5/12/06, stoffell <[EMAIL PROTECTED]> wrote:
On 5/11/06, Tim Robinson <[EMAIL PROTECTED]> wrote:
> There is a lot of junk in your zapata.conf that you do not need, as it
> relates to analogue lines. This might be causing confusion?
I have tried a similary config to yours, doesn't helps. I ha
On 5/12/06, Jerry Jones <[EMAIL PROTECTED]> wrote:
I believe the hint priority must be in the same context as the phones
extension number, in this [local]
Additionally, it may not be the first 'exten =>' line, at least in
some versions, so best to put them at the end of the context.
PLUS: Avo
On 5/16/06, Avi Miller <[EMAIL PROTECTED]> wrote:
Michael J. Tubby B.Sc (Hons) G8TIC wrote:
> call then transfers it on to another extension transferee (recipeient)
> sees the Caller*ID
This behaviour changed in Asterisk 1.2 -- add "o" to your Dial options
and Asterisk will retain the original C
On 5/19/06, Remco Barende <[EMAIL PROTECTED]> wrote:
Most people seem quite positive about Snom phones, I cannot share this
opinion.
The displays are dying quite often, and firmware is buggy. I have tried
every firmware from 4.5 up to 5.x and 6.04 but keep having problems with
phones locking up
On 5/19/06, Remco Barende <[EMAIL PROTECTED]> wrote:
> I find that the snom phones can be over-sensetive to network glitches,
> which with the default configuration can cause a reboot (usually
> caused by cheap switches). Try changing the reboot on ethernet unplug
> setting to "ignore".
Good ide
On 5/22/06, Remco Barende <[EMAIL PROTECTED]> wrote:
On Fri, 19 May 2006, Steve Davies wrote:
> I find that the snom phones can be over-sensetive to network glitches,
> which with the default configuration can cause a reboot (usually
> caused by cheap switches). Try changin
On 6/7/06, Marnus van Niekerk <[EMAIL PROTECTED]> wrote:
I have a 1.2.4 * box with two HFC modems using chan_modem_i4l and several
SIP phones and ATA's.
We have a terrible delay on calls between the PSTN (isdn BRI) and the SIP
phones. All internal calls are fine. My first thought was that t
On 6/8/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
I have to dial prefix 9 for non local numbers however
when i missed calls i Can't redial this number
because of "9" is not append .
I use polycom phones .
What Can i do ?
My preferred answer to this question is to not use a '9' prefix.
On 6/8/06, Brian Swan <[EMAIL PROTECTED]> wrote:
[snip]
I've followed the numerous suggestions in the mailing list archives
which is what has enabled me to get this far. After trying all the
echo cancelers, and all the settings on each I settled on:
- KB1 (with AGGRESSIVE_SUPRESSOR)
- echocanc
On 6/9/06, Dr. Michael J. Chudobiak <[EMAIL PROTECTED]> wrote:
Consider getting a Sangoma A200D
(http://www.sangoma.com/datasheets/p_a200-specs) with the optional
hardware echo canceller module. It "just works" for echo cancellation;
no tweaks required. It takes a while to figure out how to inst
On 6/9/06, Brian Swan <[EMAIL PROTECTED]> wrote:
Actually, that's what I started out with, and outboud calls were the
same as now, inbound calls had a huge amount of echo (until I turned
on Aggressive). In my testing I actually didn't notice any
difference between KB1 actually worked better then
On 6/9/06, Doug Crompton <[EMAIL PROTECTED]> wrote:
Does it matter if you use upper or lowercase rules - I.E. - "x" vs. "X" or
mix them? Not that I would do that as a rule but sometimes you make
mistakes!
I have no idea, but I bet you could try it, and find out faster than
you'll get an answer
On 6/12/06, Doug Crompton <[EMAIL PROTECTED]> wrote:
It seems that any firmware is usable on any hardware as my hardware is
2.x. I wonder if 3102 firmware could be used on the 3000. Is the size the
same? I guess you would have to be willing to make a brick to find out!
I have not tried this,
On 6/14/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
Sangoma is NOT releasing a single T1 with echo cancellation.
But AFAIK they ARE releasing (have released?) a dual T1/E1 card with
hardware EC, which may be sufficient compromise.
Cheers,
Steve
___
On 6/15/06, Idris AVCI <[EMAIL PROTECTED]> wrote:
Hello,
There are 3 PRI's connected to the card each from different operators.
Especially echo occured on span 3 is really annoying. Configuration files
are as follows. Is there something wrong in conf ?
Have you verified that the provider on
On 6/15/06, Mike Fedyk <[EMAIL PROTECTED]> wrote:
Steve Davies wrote:
> We have even experienced problems within Europe where providers route
> national calls via international routes to save money. This adds
> significant latency and makes any echo so heavily delayed that
&g
>> More than 128ms?
>
> 128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms, but
> definitely more than 16ms.
No, 128ms = 1024 taps
Like what sangoma offers.
Ding, Ding, Ding, Ding!
Okay, to be complete in my answers:
No I do not get more than 128ms delay caused by European routin
On 6/19/06, Idris AVCI <[EMAIL PROTECTED]> wrote:
Hi Steve,
Thank you for your answers. First of all span 3 is not a satellite link
and no echo occurs when I connect this line to another pbx with HW EC
feature. I use TE411P with hardware EC and asterisk version 1.2.5. Do I
have to do something
On 6/19/06, Remco Barendse <[EMAIL PROTECTED]> wrote:
found it, in bristuff-0.3.0-PRE-1q/zaphfc/Makefile
again it is required to change KSRC=/usr/src/linux/ to
KSRC=/usr/src/linux-2.6/
I wonder why neither florz nor kapejod fixes these problems (several
modules do not compile).
This is a dist
On 6/20/06, Warren <[EMAIL PROTECTED]> wrote:
If anyone out there using VoIP WiFi phones? If so, which ones and what
do you think about it?
As others have said, they are all horrible.
If you /must/ have one, the Hitachi WIP3000 or WIP5000 both do the
job. AFAIK these are the only phones with
On 6/26/06, Peter J Dean <[EMAIL PROTECTED]> wrote:
I have a issue trying to understand why Asterisk-PBX, when a SNOM
(320 or 360) successfully redirects/diverts a call when it is a local
extension, but fails when you enter external number.
Both the local extension dial and external extension dia
On 6/29/06, Mark Ackroyd <[EMAIL PROTECTED]> wrote:
Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10614 setup_zap: Unknown
signalling method 'pri_cpe'
Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10239 setup_zap: Signalling
must be specified before any channels are.
Am I right in thinking that's it's so
Hi,
Does anybody know whether it is possible to completely disable call
waiting on an Aastra 9112i?
The latest 1.4.x firmware allows the call waiting /tone/ to be
disabled, but I have users who really do not want a second call to
ring them at-all if they are busy, yet the Aastra seems to insits
alplan from *
Just starting to play with the Aastra so not much knowledge on them yet.
On Jul 3, 2006, at 6:41 AM, Steve Davies wrote:
> Hi,
>
> Does anybody know whether it is possible to completely disable call
> waiting on an Aastra 9112i?
>
> The latest 1.4.x firmware allow
On 7/6/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
Been testing a new spa941 with the latest firmware (sip to sip). I
noticed that unlike 7960's, if a user of a 7960 hangs up at the end of a
conversation, the 941 does not automatically hangup. Rather, the 941
sits there for about 5 seconds, then
Hi,
I was wondering whether anyone has any input into the reliability of
faxing (over a PRI) using spandsp and rxfax.
99% of times this is a reliable combination - we use it almost
exclusively, but there seem to be certain fax devices which have
problems talking to us. Most notably fax modems, a
On 7/10/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
>
> Any pointers on how to diagnose or improve this would be appreciated.
>
Install HylaFAX and iaxmodem on your Asterisk box.
Thanks, I will do.
I assume that iaxmodem talks to the PRI, and then HylaFax talks IAX to
asterisk? Any downsides/go
Feedback on my testing here.
On 7/10/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
Steve Davies wrote:
> Hi,
>
> I was wondering whether anyone has any input into the reliability of
> faxing (over a PRI) using spandsp and rxfax.
>
> 99% of times this is a reliable combin
On 7/11/06, Muhammad Zaka <[EMAIL PROTECTED]> wrote:
I have following issue with receiving fax from Samsung sf2000 fax
machine. Faxes with other machine works OK;
I am running asterisk 1.0.10 with spandsp-0.0.2pre26 on zaptel 1.0.10.
The following is the log
[snip FLOW trace]
Please can you
On 7/12/06, al gav <[EMAIL PROTECTED]> wrote:
Hi all
I need a help with asterisk+fax -> fax to email
I am trying to setup fax to email with asterisk with no success.
I have asterisk 1.2.9.1 running on CentOS
i have created extension 300 which should receive faxes.
Which version of spandsp d
On 7/13/06, I T <[EMAIL PROTECTED]> wrote:
Actually Aastra phones do support disabling Call Waiting on the their
phones. Just add the following to your configuration file:
call waiting tone: 0
It's in the release notes for the 1.4 release
http://www.aastratelecom.com/downloads/RN-001024-00-08%2
On 7/13/06, unplug <[EMAIL PROTECTED]> wrote:
In dial plan, we can set the priority. However, I find that the
priority count is a global value. It will continue to increment no
matter in different context. Below is what I have tried.
This is exactly how it is designed, and is a very useful f
On 7/13/06, unplug <[EMAIL PROTECTED]> wrote:
Thanks. Could you tell me how useful it is? As I think if the
priority is reset in every context. I can design something like this.
My default context is myflow and your default context is yrflow. So
I can easy to delete a context in the context
On 7/17/06, unplug <[EMAIL PROTECTED]> wrote:
According to your setting, below is meaningless. Am I right?
[apps]
include => emergency
include => cfwd
include => mailbox
[emergency]
exten => 911,1,do stuff here
[cfwd]
exten => *31,1, enable cfwd
exten => *32,1, disable cfwd
exten => 911,1, do
Someone here suggest to use
macro to implement my design. As I want to use ARA in my design. If
I use macro to here, ARA will be meaningless.
Yes, I suggested macros. Sorry, what is ARA?
Steve
___
--Bandwidth and Colocation provided by Easynews.com
On 7/19/06, Dan Brummer <[EMAIL PROTECTED]> wrote:
Hello,
Well I was having transfer issues in 1.2.9.1 so I downgraded to 1.2.7.1.
For testing I installed 1.2.10 on a test server and setup two Polycom SIP
phones. Tried the transfer on this configuration and had the same issues.
Here is a log fro
On 7/20/06, Dan Brummer <[EMAIL PROTECTED]> wrote:
Setting canreinvite=no on all the sip peers and gateway made the warm
transfer work. I'm still noticing SIP entries, is this an
issue?
== Spawn extension (ANC, 1691, 2) exited non-zero on
'SIP/1691-09766938'
I think the ZOMBIE entries are
On 7/26/06, Paul Hales <[EMAIL PROTECTED]> wrote:
Watch your extensions don't conflict with local numbers - in Australia
1XXX numbers are valid!
And similarly "emergency services" 3-digit numbers, 112, 999, 911 etc.
In fact I would avoid numbers that are even similr to this. 1112 could
easily
On 7/27/06, Gareth Blades <[EMAIL PROTECTED]> wrote:
Has anyone else had problems with the Nokia E61 and E70 phones not
always answering voip calls?
We have them connected via a local access point (so no router/NAT) and
sometimes the phones dont ring when called. They are registered ok and
if you
On 7/27/06, Jon Schøpzinsky <[EMAIL PROTECTED]> wrote:
Hello
Just use Snom or grandstream phones. They can be provisioned very easily via
HTTP.
You just setup a config URL on the phones, and they get their configurations
from there.
If you want to get more advanced, they can send along their M
On 7/28/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote:
On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote:
> Does anyone know how to set up QoS on the SNOM 360 ? Thanks.
What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a
Snom 360 that will manage things for you
On 7/28/06, Dovid Bender <[EMAIL PROTECTED]> wrote:
Also SNOM says by Vlan to set the vlan and then the value for the qos. When
you set Vlan to 0 it is supposed to be no Vlan. However once I set it the
vlan on the SNOM to 0 and I reboot the phone is no long accessable from the
network and I have
On 7/31/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote:
On Friday, July 28, 2006 3:08 PM Dovid Bender wrote:
> I am trying to have thier PC run thru the port on the phone and the
> phone give prioroty to itself and the rest to the PC. When my client
> does a big download the phone call gets r
Sangoma are about to release a 2-port card I believe, but I have not
heard of a 1-port unit. You would need to buy an external device,
which will probably raise to cost so close to the 2-port solution that
you may as well use that instead.
Regards,
Steve
On 3/9/06, Avi Miller <[EMAIL PROTECTED]>
On 3/15/06, Robert P. McKenzie <[EMAIL PROTECTED]> wrote:
> A user of mine has discovered that when you call into asterisk and get the
> IVR menu with options 1-5 available, if you
> dial 1 then immediatly dial 2 it will connect you to 2 and not 1. I expect
> this is due to the digit timeouts an
In case this is useful to someone...
Initially running * 1.0.7 and the default canceller, about 1 in 20 E1
PRI calls still had echo, sometimes quite bad.
Updated to * 1.0.9, and backported KB1 from 1.2 HEAD to this build -
No noticable change over the previous version, but we ran with it
anyway a
the
> makefile? Care to post a backported tar?
>
> -Original Message-----
> From: Steve Davies [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, March 15, 2006 2:47 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Echo canceller data-points
>
>
>
oops. attachments are blocked :) I'll email it directly to anyone who
provides an email address.
Regards,
Steve
On 3/16/06, Steve Davies <[EMAIL PROTECTED]> wrote:
> Here is the patch file which I use (I manually removed some other
> parts of the patch, so I hope it is okay
On 3/16/06, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
> On Wednesday 15 March 2006 16:46, Steve Davies wrote:
> > I thought I would try the 1.2 trunk/HEAD version of MG2 with the extra
> > knobs and whistles, but found 2 problems. This version trains even a
> > norm
On 3/6/06, Colin Anderson <[EMAIL PROTECTED]> wrote:
> I was always puzzled by posts to the list about people having problems
> getting hints to work on a Snom, since I always seem to have no problem
> making it work. That is, until today when I tried to get a sidecar to work.
> All I could do was
On 3/21/06, Nabeel Jafferali <[EMAIL PROTECTED]> wrote:
> > try SetCallerId or set callerid=name <(xxx)xxx-> in sip.conf or
> > iax.conf (depending on what you are using)
>
> I am not using SIP or IAX2 clients. As mentioned in the original email, this
> is from PRI to PRI.
>
> I could use SetCa
On 3/21/06, Mimmus <[EMAIL PROTECTED]> wrote:
> Hi,
> at my Asterisk box, I have a few of IAX2 phones (configured with
> alaw/ulaw/gsm codecs, in this order) and a PRI E1 line.
> In iax.conf I hav:
> disallow=all
> allow=alaw
> allow=ulaw
> allow=gsm
>
> During some incoming call, I read at con
On 3/22/06, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
> On Wednesday 22 March 2006 05:26, Steve Davies wrote:
> > Another "hint" for getting hints working, although this only relates
> > to older 1.0.x versions of Asterisk (It is already fixed in 1.2.x) is
On 3/23/06, Jared Davison <[EMAIL PROTECTED]> wrote:
> I was having trouble getting hints to work with my GXP-2000 (with the beta
> firmware). I am running Asterisk 1.2.5. I had hyphens in the SIP channel
> names and it wasn't working. I have changed them to underscores and it has
> worked in 1.2.5
On 3/28/06, Giordano Grandis <[EMAIL PROTECTED]> wrote:
>
> Hi all,
> I'm using bristuff 0.2.0 RC8o with a HFC pci card and on several calls I saw
> that the echo cancellation is on OFF
>
> Echo Cancellation: 0 taps, currently OFF (the result of "zap show channel
> 1-1" for example)
>
Echo cancel
AIL PROTECTED]> wrote:
> Ok, but is there a way to check if echo cancellation is active on a call in
> progress ?
>
> Thanks
>
> Giordano
>
> -Messaggio originale-
> Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Steve Davies
> Inviato: m
On 4/5/06, Jon Farmer <[EMAIL PROTECTED]> wrote:
>
> I can't for the life of me work out why this is not
> working. When in the campon contect if you hit a DTMF
> key 2 you get moved to the exten => 2 defined in the
> mainmenu context not the exten => 2 defined in the
> campon context. What is wron
2009/2/11 OCG Technical Support :
> Don't expect too much from Aastra. In our previous dealings trying to
> report serious bugs (like phone lockup/crash) to Aastra, they didn't want
> the details, or they simply gave us canned answers which did no good.
> (Superficial tech support)
>
> We've moved
Hi,
The part of pedantic=yes that you need to make '#' work is URL
encoding, unfortunately it comes with a whole load of other baggage
that breaks a lot of different things. A simple fix might be to
comment out the parts of pedantic=yes that you do not need in the
source code and re-compile - This
2009/3/3 Giorgio Incantalupo :
> Hi,
>
> I'm monitoring an Asterisk 1.2.18 box because sometimes I get two Snom
> 360 phones creating a lot of SIP channels between them and it seems they
> never die.
> How can it be?
>
> Thank you.
>
> Giorgio
I would suggest looking for network dropouts and for p
2009/3/12 Julian Lyndon-Smith :
> Has anyone in the UK got ANI to work on an inbound call ?
>
> Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30
>
> Julian
>
Have you asked the Telco to send the ANI data? AFAIK, this is disabled
by default on all BT lines. I assume they are able
2009/3/16 David Ruggles :
> Is it possible to control the light on a programmable button without the blf
> option? I'm using a programmable button to turn call recording on and off
> and I would like the light to indicate the status.
>
> Thanks,
>
9133i phones are pretty much obsolete, and are not
While we have your attention Steve (Underwood) do you have a
high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We
currently use 0.0.4 with a very high success rate. Is there any
benefit in moving up to a newer library? I looked at the Changelog in
the source, but it stopped at 0.0
2009/3/17 David Backeberg :
> On Tue, Mar 17, 2009 at 10:34 AM, Steve Davies wrote:
>> While we have your attention Steve (Underwood) do you have a
>> high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We
>> currently use 0.0.4 with a very high succe
Hi,
I have just discovered (a year after it was implemented) a possibly
undocumented incompatability between IAX in Asterisk 1.4 and any
version of Asterisk pre-March 2008.
It seems an AST_CONTROL_SRCUPDATE frame type was added (in March '08),
but no mechanism to negotiate whether it can be sent
2009/3/23 Jeffrey Phelps :
> I’m trying to get the BLF to work correctly on my Polycom phones. I have
> the buddy watch working correctly, but can’t get the BLF to change based on
> the state…
>
> Example:
>
> When an extension is ringing, I get the same ‘red light’ that I get when the
> extension
2009/3/23 Kevin P. Fleming :
> Tilghman Lesher wrote:
>
>> It will have no effect. The issue has always been that if the stream source
>> changed during a call, the sequence numbers could be reset, sometimes
>> causing audio weirdness. What has changed is that we're now able to tell
>> the other
2009/3/31 Steven J. Douglas :
> Yahya Mohammad wrote:
>> I'm running asterisk on Ubuntu 8.10. I have two 'register' lines in
>> iax.conf for registering with two remote servers. However only the
>> first one registers at system startup. I always have to issue an 'iax2
>> reload' command before * r
Hi,
It there any way of getting queue data from within a dialplan in order
to change call routing based on what is already happening? Something
like the following would be ideal:
exten => X.,n,Set(WAITING=${QUEUE(qname|waiting)})
exten => X.,n,Set(TALKING=${QUEUE(qname|talking)})
Can anyone sugg
I have found that you get good results by setting a per-device
GROUP_COUNT(), which prevents dialling if it is non-zero, and setting
call-limit to 999.
In Asterisk 1.0.x there were separate in- and out-bound call limits,
but IIRC this was pretty broken, and was removed.
See http://www.voip-info.o
201 - 300 of 435 matches
Mail list logo
