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 hangs up. h is
executed when
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
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
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 to
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 =
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 order
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?
I interpret
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 meant was shall 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
Hi,
I have a couple of questions about Quad-BRI solutions for Asterisk,
and was hoping that I might get some feedback based on other people's
experience.
We currently use the Junghanns card, which is a pure Zaptel solution,
which is fantastic, but they have no hardware EC solution, and their
Is it related to Dial() options:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial
h: Allow the callee to hang up by dialing *
H: Allow the caller to hang up by dialing *
Perhaps it is some other system inline with yours that has this
feature enabled. I certainly found this to be
2008/12/5 dubravko caric [EMAIL PROTECTED]:
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
2008/12/18 Freddi Hansen f...@danovation.dk:
You might want to use the version at:
http://github.com/davies147/astmanproxy/tree/master
it's updated and an error that can cause segfault when client
disconnects has been fixed.
Freddi.
Given that I've seen at least 4 people using the code
I did this a long time ago, and just based it on a PAP2T XML
configuration, with 8 lines instead of 2, and it worked fine. Sorry I
don't have any useful examples to hand anymore. Are you sure it is not
just a missing slash or angle-bracket in your source XML? Try opening
it in a browser to see if
2009/1/27 Udo Schacht-Wiegand aster...@wiegand.name:
Grygoriy,
[...] A practice that was once described in the code comments as
being nasty.
thanks for your input. My knowledge of 'hard core' programming is limited,
so I cannot judge on what is written on freeswitch.org. Though it sounds
2009/2/11 OCG Technical Support supp...@ocg.ca:
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)
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 -
2009/3/3 Giorgio Incantalupo gincantal...@fgasoftware.com:
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
2009/3/12 Julian Lyndon-Smith aster...@dotr.com:
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
2009/3/16 David Ruggles da...@safedatausa.com:
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
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
2009/3/17 David Backeberg dbackeb...@gmail.com:
On Tue, Mar 17, 2009 at 10:34 AM, Steve Davies davies...@gmail.com 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
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 jphe...@mjlm.com:
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
2009/3/23 Kevin P. Fleming kpflem...@digium.com:
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
2009/3/31 Steven J. Douglas stev...@moij.biz:
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
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
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
answering a queue
QUEUE_MEMBER_LIST: Returns a list of interfaces on a queue
QUEUE_WAITING_COUNT: Returns the number of callers currently waiting in a
queue
Just my two eurocents,
l.
2009/3/31 Steve Davies davies...@gmail.com
Hi,
It there any way of getting queue data from within a dialplan
I have an ITSP we are trying to work with that has an Unusual way of
working, but that said my understanding of their behaviour is that it
is fully RFC compliant. Can someone suggest how I might be able to
interoperate under these circumstances:
We register fine with them, and send the default
...@host[:port][/extension]
; If no extension is given, the 's' extension is used.
There you have it ... Contact: sip:s
set the extension and you should be fine
Martin
On Mon, Apr 6, 2009 at 7:45 AM, Steve Davies davies...@gmail.com wrote:
I have an ITSP we are trying to work
2009/4/7 Olle E. Johansson o...@edvina.net:
[snip]
The REGISTER request in the RFC was really written for a device.
The way providers use it for trunks with multiple DIDs is outside of the
RFC and is discussed in relation to the SIPconnect specification in
the SIP forum.
Some providers
2009/4/6 Ed W li...@wildgooses.com:
Hi, got a Sangoma A200 with a bunch of extension cards and having real
problems getting it to deal with a normal single BT line
The A200 is a great card, and we use it quite a lot in the UK. Mostly
we use the A200D for the echo cancellation.
Symptoms are
Bria is the commercial extension of Xlite which adds corporate features.
2009/4/9 ContactTel Business li...@contacttel.com:
Xlite etc, counterpath.com have AA features, not sure about central phone
book.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Sorry - this is a bit off topic, but there is almost certainly someone
here who will know the answer... Perhaps even a snom employee :)
In recent snom firmware releases, the following sequence always causes
a call to be sent from line 'n'
Receive call on Line 'n' (where n 1)
Press Hold
the same issue.
2009/4/23 Steve Davies davies...@gmail.com:
I think I have a site where this is happening, but all I see is a
series of outbound calls, which look perfectly normal, but at some
random point, ISDN channels stop being available, until they run
out. It can go anywhere from weeks down
2009/4/23 Steve Davies davies...@gmail.com:
My comment, (forwarded from Bristuff list) - A few people are seeing a
Cause 34 (congestion) from ISDN installs, where there clearly is an
available channel. This was originally related to Bristuff as it
happens to ISDN2 users, but there is at least
Hi,
Is there some more thorough documentation of this change that has
happened in 1.6? The upgrade.txt and changes.txt files mention it, but
I have already seen details of this change that do not appear to be
documented except in conversations on the mailing list...
1) It appears that it is no
2009/4/29 Tilghman Lesher tilgh...@mail.jeffandtilghman.com:
Let's also be clear about what Gosub is replacing. Gosub replaces Macro for
AEL2. The side effects of this are relatively unfelt, unless you're doing
something unusual like defining subroutines in AEL and calling them from
Hi,
This may be completely wrong, but I have a feeling it may be related.
Have you enabled overlapdialling in zapata.conf for the channels
that are on the channelbank? If not, the 1st digit will be sent in,
not match the dialplan, and be hungup.
*7xxx is probably working because that matches a
Oh, and have you enabled Sangoma's DTMF detection in their config
file? That is probably also necessary.
Cheers,
Steve
2009/5/8 Steve Davies davies...@gmail.com:
Hi,
This may be completely wrong, but I have a feeling it may be related.
Have you enabled overlapdialling in zapata.conf
2009/7/14 gergis.rasmy gergis.ra...@gmail.com:
could anyone help explaining what does this error mean?
i get this error when make a video/ audio call from X-lite to Bria prof.
phone
rtp.c:1739 ast_rtp_read: Unknown RTP codec 126 received from '10.0.0.26'
Gres
To quote Counterpath, 126 is
For the curious, can anyone tell me how this flag fixes the issue? - I
have seen the error before, but always assumed it was related to hung
channels.
Thanks,
Steve
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Problema solved!
Just put resetinterval=never inside zapata.conf
On 9/12/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Davies wrote:
For the curious, can anyone tell me how this flag fixes the issue? - I
have seen the error before, but always assumed it was related to hung
channels.
Thanks,
Steve
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED
On 9/12/06, John Marvin [EMAIL PROTECTED] wrote:
shadowym wrote:
[snip]
Asterisk not padding files to
even 20ms increments when playing them. So, although that may be a bug
in Asterisk, I thought I would see if that was the problem by writing a
quick C program to pad all my ulaw files to
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
I get many of these warnings inside Asterisk log:
WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel
0/1 already in use on span 1. Hanging up owner.
What does they mean??
Can I assume then that
On 9/12/06, Steve Davies [EMAIL PROTECTED] wrote:
I don't suppose you know what the silence padding bytes would be for ALAW?
Found it... It is 0x55.
Thanks for the program :)
Steve
___
--Bandwidth and Colocation provided by Easynews.com
On 9/12/06, Steve Davies [EMAIL PROTECTED] wrote:
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
I get many of these warnings inside Asterisk log:
WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel
0/1 already in use on span 1. Hanging up owner.
What
On 9/12/06, Olivier [EMAIL PROTECTED] wrote:
Hi,
What would you suggest to implement directed call pickup on bristuffed
Asterisk 1.2 ?
I'm after tle ability to pick a specific ringing call (without caring about
which call arrived first, for example).
Something like : *8 + local extension
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
I'm interested to understand why I many messages like:
WARNING[21314] chan_zap.c: Ring requested on channel 0/1 already in use
on span 1. Hanging up owner
How can a channel be already in use??? That means the channel is
busy...if it is
On 9/13/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi Steve,
I agree with you..telco knows better!
If telco sends a ring on channel X and asterisk has already used it,
couldn't asterisk shift that call on another channel Y or it is
obliged to answer on channel X?
The telco is in
On 9/13/06, Michael Welter [EMAIL PROTECTED] wrote:
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
To start the ball rolling:
Software: Zaptel 1.0.9, Asterisk 1.0.9, BRIstuff-0.2.0, wanpipe-2.3.2
PRI interface: Sangoma A101U (UK E1)
Phones on sites
On 9/13/06, Steve Underwood [EMAIL PROTECTED] wrote:
Artifex Maximus wrote:
If you look in http://www.soft-switch.org/download/snapshots/snapdsp,
the latest snapshot of spandsp and the app_rxfax and app_txfax
applications there provide ECM. It is less well tested than the
spandsp-0.0.2 code,
On 9/14/06, Steve Underwood [EMAIL PROTECTED] wrote:
Steve Davies wrote:
[snip]
This looks pretty good I have to say - The ECM seems as if it may be a
little intolerant... On a fax machine where I got 100% success in the
past with 0.0.2, I am now getting result (60) Disconnected after
Hi Steve,
On 9/14/06, Steve Davies [EMAIL PROTECTED] wrote:
On 9/14/06, Steve Underwood [EMAIL PROTECTED] wrote:
Steve Davies wrote:
[snip]
This looks pretty good I have to say - The ECM seems as if it may be a
little intolerant... On a fax machine where I got 100% success in the
past
On 9/25/06, Klaus Darilion [EMAIL PROTECTED] wrote:
Hi!
I have problems when bridging from SIP to PRI. As soon as the setup
message is sent, Asterisk replies with 183 to the sender.
Although there is nor PROGRESS message received, the 183 is sent as the
SIP channel received a voice frame and
On 9/25/06, Colin Anderson [EMAIL PROTECTED] wrote:
It's excellent home phone. I wouldn't use it in a business environment.
No
hold, no one-touch voicemail. However, it works great!
aw crap, that's a biggie but I think I can work around it, teach the user to
dial *98 for voicemail, *700 for
On 9/27/06, Mike Hammett [EMAIL PROTECTED] wrote:
Has anyone actually gotten ASTTAPI to work? I can't seem to get it to work,
yet I have other TAPI setups (SNAP and xtelsio) working fine. I have
noticed that SNAP and Xtelsio act differently. Etelescript is the
application that will be
On 10/17/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote:
Hello,
i have call time irregularites in my asterisk CDR. I a currently using a
mysqly backent to save CDR records and use this to generate bills at the end
of each month. However, my users are complaining that they gety charged for
even
Hi,
For a significant time now (since about 0.2.0-rc8n) the qozap driver
has become very verbose if an ISDN line is not connected... I get the
messages below every couple of seconds in the asterisk logs.
The flaw in the messages is the Alarm cleared message - The alarm
cannot possibly be
On 10/20/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Oct 19, 2006 at 01:42:01PM +0100, Steve Davies wrote:
Hi,
For a significant time now (since about 0.2.0-rc8n) the qozap driver
has become very verbose if an ISDN line is not connected... I get the
messages below every couple
On 10/20/06, Steve Underwood [EMAIL PROTECTED] wrote:
M. Shokuie Nia wrote:
Dear folk,
My problem solved after two day research and try and error method ;). It was
related to rxgain of the board im using. I've set the rxgain to 12 and it
seems made some problem. As far as I got the spandsp is
On 10/20/06, R.R. Libera [EMAIL PROTECTED] wrote:
Wow, this is a completely neutral and very valuable review. Thanks a lot
Zoa.
I´m an * newbie; my little box will only needs 20 extensions to give
termination to remote users and I´m about to buy a PRI interface; I
decide to get Sangoma
On 10/20/06, Mohammad Shokuie [EMAIL PROTECTED] wrote:
Anyways, let me take the most benefit as im sure you'd read this post, i
have problem with the size of received page which is shrinked, can u give me
a hint about this problem too :)
This is probably the problem of the application that
On 10/23/06, Giedrius Augys [EMAIL PROTECTED] wrote:
Hi,
I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1. I get error:
configure: error: Can't build without libtiff . But I have installed tiff
from port tiff-3.8.2. I understand that the problem is about libtiff, and
spandsp can't
On 7/12/05, Patrick Friedel [EMAIL PROTECTED] wrote:
OK, last showstopper that I just can't puzzle my way through - parking
calls with the snom phones. I get the two phones connected, I hit
transfer on one, the other phone goes to MOH and the first phone gives
me DT, so I dial 700 and hit the
Hi,
I have searched and searched, but cannot identify what is happening here...
I have several snom190 phones, and all of them have the 5th function
key set to call asterisk by using the destination option. This
automatically causes the phone to SUBSCRIBE for NOTIFY messages for
the asterisk
Hi,
I am using a number of snom190 phones, and an asterisk gateway
server, and recently started experimenting with call transfers. The
snom phones provide support for attended and un-attended call
transfer, so I would rather use that than call-parking.
I have found that un-attended transfer
On 6/13/05, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
I am using a number of snom190 phones, and an asterisk gateway
server, and recently started experimenting with call transfers. The
snom phones provide support for attended and un-attended call
transfer, so I would rather use that than
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 and
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
:
Is it onerous to backport or is it a case of fiddling around with 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
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!) - It should
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
normally clean line in about 10 seconds
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 get a
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 SetCallerID, but
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 console:
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
that status changes are not notified
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. So
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 cancelling is
] 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: martedì 28 marzo 2006 16.43
A: Asterisk Users Mailing List - Non
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 wrong? The
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
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 would
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
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 haven't
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: Avoid
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 Caller
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 idea, I
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 changing the reboot on ethernet
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
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)
-
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 install
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 here
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,
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