Are there any patches that would fix the agent staying in 'busy' state problem as well? I couldn't find any so far..Dinesh Nair <[EMAIL PROTECTED]> wrote: On 06/12/06 20:21 Matt said the following:> AHHH! We use the Xfer button on our Aastra 9133is to do transfers> for some reason (see ano
I think it is. Agent's device is actually a softphone, registered as one of the clients defined in sip.conf. Would it help if I posted the configs?BJ Weschke <[EMAIL PROTECTED]> wrote: On 6/12/06, aston martin <[EMAIL PROTECTED]>wrote:>> 1. Call comes in the queue (
ingyour attended transfer? Step-by-step.On 6/12/06, aston martin <[EMAIL PROTECTED]>wrote:>> It seems that Asterisk does not free up agent after attended transfer. The> agent
stays in 'busy' state for as long as the conversation between the> caller and person, to w
It seems that Asterisk does not free up agent after attended transfer. The agent stays in 'busy' state for as long as the conversation between the caller and person, to which call was transfered, is active. Does anyone know any workarounds for this problem? _
On Jun 11, 2006, at 8:15 AM, James Harper wrote:
Additionally, just to satisfy myself that I wasn't going mad I changed
the port from 5060 to 5070 and now things are working, so something is
definitely playing up on port 5060.
If you are behind a NAT perhaps two SIP devices are both trying to
On Jun 11, 2006, at 2:32 AM, John Joseph wrote:
Hi
Was able to communicate clearly with e60 and E61
with asterisk with new access point , even though the
access point security setting was of “opennetworks” ,
the previous one was of “WEP” , I feel this was a
major hurdle in communication ,
On Jun 9, 2006, at 9:47 PM, John Joseph wrote:
--- Nick Burch <[EMAIL PROTECTED]> wrote:
Recent posts indicate people have been having luck
with the nokia E60/E7x
phones and asterisk.
I was able to register E60 ,but a SIP calls made
from the other phone to E60 has problem , the PBX sip
On Jun 8, 2006, at 10:46 AM, [EMAIL PROTECTED] wrote:
Hi all,
Sorry, I've been late on this thread.
Well i own an E60. Its been almost 2 weeks i've been trying to
register myself on to my asterisk server but i have been failing to
register. can anyone provide some links or sample configs on t
On Jun 8, 2006, at 3:13 AM, Peter J Dean wrote:
I have an issue with DTMF. DTMF is being partly recognised by some
external IVR systems (banks, billing, etc), other IVR systems have
intermittent issues. Call our VSP directly and using their IVR system
without issue, and our internal IVR works
Doug,
DTMF in a complex environment like this can break for many reasons.
Please don't generalize your problem onto other people. He didn't say
he was using the Sipura 3000 so we don't know for sure what his issue
is.
I am only saying this due to my own limited experience and how many
way
On Jun 8, 2006, at 11:52 PM, Noc Phibee wrote:
Martin Joseph a écrit :
On Jun 8, 2006, at 10:26 PM, BJ Weschke wrote:
On 6/9/06, Noc Phibee <[EMAIL PROTECTED]> wrote:
anyone have a answer at this question ?
Noc Phibee a écrit :
> Hi,
>
> Is it possible de t
On Jun 8, 2006, at 7:00 PM, Doug Crompton wrote:
I think he clearly states at the end of his message that he is using
the
SPA-3000.
My bad. Should learn to read more carefully (and type too) .
My apologies.
Marty
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On Jun 8, 2006, at 10:26 PM, BJ Weschke wrote:
On 6/9/06, Noc Phibee <[EMAIL PROTECTED]> wrote:
anyone have a answer at this question ?
Noc Phibee a écrit :
> Hi,
>
> Is it possible de tell asterisk to increase the volume?
>
> When we place or recieve a call the volume is very low, us
On Jun 8, 2006, at 11:04 AM, Douglas Garstang wrote:
Well, this kinda sux.
What? You repeating yourself ad nauseam? I agree.
Marty
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On Jun 8, 2006, at 6:49 AM, Jim Lynch wrote:
I need another fxo line. Has anyone had any experience with
connecting the gsm488 into asterisk?
If you mean the Handytone 488 from grandstream, yes.
It has some pretty major issues IMO. First of all, when the incoming
calls through the FXO ar
On Jun 7, 2006, at 1:51 PM, Thomas Kenyon wrote:
Mike Fedyk wrote:
First of all, I'm not knocking Sipura/Linksys. I have heard very good
things about their products.
I'm just wondering if they are the only quality shop on the market. I
know about the zoom 5801 where you can't dial out the F
On Jun 7, 2006, at 6:55 PM, M.Hockings wrote:
I have a small asterisk setup here with one POTS line, one VOIP SIP
connection an FXS connection to the house phones and a bunch of
softphones. Local calls are routed out through the POTS line and long
distance through the VOIP line. This works
On Jun 7, 2006, at 10:22 AM, Cory Andrews wrote:
Have a customer running a 3rd party PBX implementation based on Asterisk, not utilizing SIP inbound and outbound calls I believe are coming through a Digium TDM2402B. They are utilizing Polycom phones. They are experiencing frequent static on the
On Jun 7, 2006, at 7:35 AM, Jon Schøpzinsky wrote:
Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work.That means, that you cant just walk down to your local airport, and use the I
On Jun 7, 2006, at 6:42 AM, Doug Crompton wrote:
I get the following * Notice ocassionally and I was curious what it
means
and if it can safetly be ignored or corrected.
Jun 7 05:40:47 NOTICE[32153]: res_musiconhold.c:511 monmp3thread:
Request
to schedule in the past?!?!
I see that on m
On Jun 6, 2006, at 4:15 PM, Doug Crompton wrote:
Ok well I am not crazy! This seems like such an important issue I am
not
sure why it has lasted for so long. DTMF is the backbone of everything
we
do here. Without it we would not have calls!! At least get the DTMF
stuff
right. I feel a little
On Jun 5, 2006, at 11:04 AM, Andrew Kohlsmith wrote:
JUST DO NOT MAKE A FUCKING PARALLEL PORT DONGLE. USB that is either
proprietary all the way, or USB which is USB->serial internally. I'd
also
NOT recommend using a little USB PIC or Atmel part and writing your own
license code... there
What part of NON-COMMERCIAL do you not understand?
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Girls, girls, you're both pretty...
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On Jun 3, 2006, at 11:13 PM, Vahan Yerkanian wrote:
Doug Crompton wrote:
I am using an SPA-3000 3.1.10d
When I have transfer enabled - 'T' in the dial string I cannot
reliably
send DTMF keys to a bank, voicemail, or other service requiring
tones. If
I disable (remove transfer option) from t
On Jun 3, 2006, at 2:04 PM, Dakota Burns wrote:
What I was attempting to visualize is the following case:
10 people in an organization pick-up their phones to make an outbound
call. Before integrating Asterisk, all calls route through their
current non-VoIP based phone provider. After inte
On Jun 3, 2006, at 12:53 PM, Dakota Burns wrote:
Have either of you any experience integrating Asterisk-related devices
into existing phone equipment (trunks/pots lines, etc. (I'm somewhat
new to legacy & voip based telephony)) to ensure specific or random
outbound calls route through Asteris
Hi Yall,
I would love to put a very compact phone on my wife's desk at work...
Ideally this would be a very small IAX phone with 2 RJ-45's so I could
drop it in without much notice and only have to beg for 1 port from the
sysadmin.
I have looked around and I don't see such an item in existen
On Jun 2, 2006, at 2:59 AM, Thomas Kenyon wrote:
Mike Hammett wrote:
I'm looking for an ATA\Voice Gateway that runs IAX and has several
ports (8 would be nice). I am looking to avoid devices that use the
same firmware as the ATCOM devices as I found them to be buggy (and a
PITA to find the pr
On Jun 1, 2006, at 2:18 AM, Benjamin Stocker wrote:
Hi!
Im looking for a very basic example for the following simple problem.
I've been searching voip-info.org and looked in the ORA book without a
clue. I have a SIP account at sip.provider.com and my own asterisk
server. What I want is the
On Jun 1, 2006, at 1:36 AM, Akpome Akpoguma wrote:
I have just finished building a prototype IVR server on a pc for
demonstration purpose.
My goal is to build a IVR server with the 4G memory, dual xeon
processor and a 4 x E1 card. The server would strictly receive
incoming calls and serve W
On May 31, 2006, at 10:32 PM, Crazy Boy wrote:
Hi Friends,
I have successfully implemented Intercom, Voicemail and International dialing using Asterisk. Now I want to connect my PSTN Lines to Asterisk server. I have 3 PSTN number (lines) to connect to Asterisk. For this, I want to use Sipura SPA
To answer your question on how I do the hook flash transfer here it is :in the globals section of extensions.conf put all your cell phone number like this :[globals]MartCell=5141234567Then add this macro in your extensions.conf :[macro-cell_user]exten => s,1,Playback(Call_Transfer)exten => s,2,Flas
aving the phone number
of my company displayed on my cell?
Thanks
Martin
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I haven't received any messages after 3/27/06 and I have tried to resub
twice without any success?
I miss the flood of messages, and I have other stupid questions to ask
;~)
Marty
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Asterisk
I too had a server room fry and need to replace h/w.
So what specific Dell servers did/do you deploy?
Where is the link w/Digium/s Dell caveats?
I'm using the Digium TDM400 card w/*
Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT)
From: Aaron Daniel <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users]
Hi,
just to complete this thread if someone faces a similiar problem:
The missing DID is caused by our telco company. It only happens when
having two different ptp lines (with different numberblocks) and calling
from one of these to the other. Calls from any other line in the world
come in wit
Hi,
using [EMAIL PROTECTED], with quadBri from junghanns.net I am facing a
strange problem:
I have set incoming routes for some extension / DID:
[ext-did]
include => ext-did-custom
exten => 23,1,SetVar(FROM_DID=23)
exten => 23,2,Goto(ext-local,23,1)
exten => 57,1,SetVar(FROM_DID=57)
exten => 57
On Mar 25, 2006, at 6:26 PM, Erick Perez wrote:
Martin, i guess im in dumb mode today because i don't get what you say, may also be because this will be my first welltech to configure.
what im trying to do is:
Sorry, I don't know if the 3804 actually uses a similar config setup to my
On Mar 25, 2006, at 4:07 PM, Martin Joseph wrote:
On Mar 25, 2006, at 2:59 PM, Derek Whitten wrote:
Has anyone ever gotten * to work on commercial unixes such as HP-UX,
Solaris, AIX?
What about other architectures than x86?
Yes, i am suing OSX on PowerPC. Works great, but no Zaptel
On Mar 25, 2006, at 2:59 PM, Derek Whitten wrote:
Has anyone ever gotten * to work on commercial unixes such as HP-UX,
Solaris, AIX?
What about other architectures than x86?
Yes, i am suing OSX on PowerPC. Works great, but no Zaptel support
due to differences in PCI technology (ie open fir
On Mar 25, 2006, at 11:18 AM, Douglas Garstang wrote:
Well, actually I did... sort of... I picked a random post and hit reply as I normally do. I forgot to clear the old text, but it's obvious that in no way it detracted from the other person's post, and the use of the word 'hijack' is highly dub
On Mar 25, 2006, at 2:01 PM, Steve Totaro wrote:
Just a few things Doug and they are just constructive criticism so don’t take them the wrong way. 1 You hijacked some else’s thread about a SIP trunk problem. Very frowned upon and will decrease people willing to help..
This does not appear to be
On Mar 24, 2006, at 9:30 PM, George Vagenas wrote:
Marty,
But with the same 128 bit upstream circuit, directly connecting the
SJPhone the Stun server and using ulaw, everything is perfect. The
problem comes when i am putting Asterisk in the picture.
Is the softphone also next to the aste
On Mar 24, 2006, at 1:19 PM, George Vagenas wrote:
Hi all,
I have the following problem, working with a SIP provider, if i setup
my SJPhone to register directly to their STUN server and working over
a 384/128 ADSL i have a really good quality, but then if i configure
Asterisk to register t
On Mar 24, 2006, at 4:25 PM, Eric "ManxPower" Wieling wrote:
"show applications" in the Asterisk CLI will list the applications
availble. "show application whatever" will give detailed docs on that
application. Also look in the docs directory of your Asterisk source
code tree for much more
On Mar 23, 2006, at 3:48 PM, Mike Dent wrote:
Hi,
which OSX softphone do you use that supports IAX2 protocol with
Asterisk?
There is a new one called JackenIAX that is working stunningly well for
me. It's still beta, but it's way better then Iaxcomm.
Marty
__
On Mar 23, 2006, at 6:58 PM, BJ Weschke wrote:
We run into situations like this often as well, and it's truly
unfortunate, because it gives our industry and the technology driving
it a bad name, and like you, some customers want to go back to TDM and
have nothing to do with VoIP at all because
On Mar 23, 2006, at 11:05 AM, Ronald Lewis wrote:
After months of BroadVoice ignoring my trouble tickets for dropped
calls, delayed termination, etc., I'm throwing in the towel. While
they have credited $19.95 to my account, they refuse to credit
anything more, despite ALL of the problems I'v
On Mar 22, 2006, at 10:24 PM, Erick Perez wrote:
Hi, does anybody have a working config or tips to connect the welltech
wellgate 3804 (4fxo) unit to asterisk via SIP ?
I think I register it via SIP with my * box, but when sending calls
from * to the wellgate the unit does not pass the call to
On Mar 23, 2006, at 10:00 AM, Charles Marcus wrote:
On 3/22/2006 Avi Miller ([EMAIL PROTECTED]) wrote:
A smarthost is another SMTP server (e.g. your corporate email server,
which should already be capable of sending outbound email) that your
Asterisk box is configured to send all outgoing mai
On Mar 23, 2006, at 8:00 AM, Michael Welter wrote:
Using a SIP connection with a CLEC, the downstream (received) audio is
perfect when the mute button is activated on the phone. However, when
there is upstream audio (i.e., talking or even breathing into the
microphone), the downstream audio
On Mar 22, 2006, at 8:25 PM, Nathan Alberti wrote:
I hope this isn't considered cross posting, i sent the following email
to Digium support but figured someone on the list may also have better
insight into my questions.
I have purchased 2 g729 licenses from Digium for testing and have the
On Mar 22, 2006, at 2:49 PM, Avi Miller wrote:
Will Glass-Husain wrote:
My local phone is a Grandstream GXP-200
dtmfmode=info
For the GXP2000's, you want to change this to:
dtmfmode=rfc2833
They don't really handle INFO mode well, in my experience.
That's what I was thinking also. In a
On Mar 22, 2006, at 5:31 AM, Bjorn O wrote:
Hello all! For several months now we’ve been experiencing a really strange problem with sound which best can be explained as choppy/stuttery, and with a touch of echo on top. Basically, parts of a conversation might be choppy, but often combined with so
On Mar 22, 2006, at 10:32 AM, Andrew Latham wrote:
I think you would need to alter the firmware to set the kewlstart to
FXS instead of FXO. This is just a thought, I have not done such. I
decided that if it worked then such a device would have been marketed
already.
D-Link has a 4 port FXO dev
On Mar 21, 2006, at 1:25 PM, Douglas Garstang wrote:
Good grief. Considering all the libraries and requirements, it'd
almost be easier to find some windows software to do this.
I prefer the Mac for audio stuff.
Did you find the slick website that does conversions?
http://www.asteriskguru.
On Mar 21, 2006, at 4:01 PM, Douglas Garstang wrote:
Oooo I think I am gonna poo my pants.
Using the microbrowser on a Polycom 601, I was able to get it to
execute a cgi script upon selection of an item. The cgi script used
Net::Telnet connect to the manager interface on another Asterisk
sy
appears that the variables sent to the dialparties.agi (methodology and
extension map) are not reset to the one extension being transferred to. Instead,
they keep the values from the original call.
Anyone run into this
and is there a fix.
Thanks
Martin
On Mar 20, 2006, at 4:39 PM, Gabriel Afana wrote:
I just did a little RTP debug and this is what it shows:
== Spawn extension (304, 301, 1) exited non-zero on 'SIP/304-d9f8'
-- Accepting AUTHENTICATED call from 216.152.244.81:
requested format = ulaw,
requested prefs = (),
actual format
On Mar 20, 2006, at 3:47 PM, Matt wrote:
I received an e-mail from a vendor who says:
"We have recently become aware of an issue in the chan_iax2
implementation of IAX2. This issue leads to degraded audio quality.
Due to this we are urging everyone to move to SIP."
I don't want to discount wh
On Mar 19, 2006, at 2:04 PM, Oliver Vermeulen wrote:
Hi All, Anybody knows how to terminated calls using Grandstream Ht488 and the FXO port ?I can ring the FXO port fine , rings 1once then give me dial tone.
I had:
exten => _NXX,1,Dial(SIP/@2003,60,D(w$EXTEN}))
exten => _NXX,2,Hangup
On Mar 18, 2006, at 11:31 PM, Steve Murphy wrote:
Hello--
In the interest of Symmetry, does anyone else in the world see any need
for a device like the IAXy (or the SIP ones from other manufacturers,
like the ATA186), but one that presents an FXO interface instead, so it
can be connected not t
On Mar 18, 2006, at 2:05 PM, <[EMAIL PROTECTED]> wrote:
I was also thinking a list for newbies...
As a newb I think that is a bad idea. First of all, the heavy hitters
will all want to avoid it( a newb list). Secondly I have learned a LOT
just by reading other peoples (non newbs) problems
On Mar 16, 2006, at 12:36 PM, Martin Joseph wrote:
So, I am answering my own post (bad form I know)...
I am trying to get my DTMF to use RFC 2833 rather then inband, so that
I can utilize lower bandwidth codecs through my FXO.
Ok, I have given up on this. There seems to be some kind of
On Mar 17, 2006, at 2:21 PM, George Pajari wrote:
I wrote earlier:
> Please tell me the obvious mistake I'm making here
The problem was a lack of sleep. Sorry to have troubled the list.
It's always nice to share the answers with us, just in case we are too
sleepy some night also...
On Mar 17, 2006, at 1:23 PM, Jerry Rasmussen wrote:
Exchange 12 will support "OVA" Outlook Voice Access. It will do this using VOIP. I was wondering if anyone has given any thought as to how Asterisk might interface with Exchange 12. It will do this using a VOIP gateway.
If this could happen
On Mar 17, 2006, at 11:37 AM, Joe Hood wrote:
Is there such a thing? Or is the Digium IAXy device the closest one
can come?
How about the PA168 based phones? Like the AT-320...
I have the ATA that is related to that phone and it has IAX firmware.
Additionally, any idea how to get the mes
On Mar 14, 2006, at 3:53 AM, artifex maximus wrote:
My asterisk system seems to have problems detecting hangups. I am
getting a LOT of voicemails with dialtone or silence.
I am using an external gateway (wellgate 3701a) and don't have zaptel
at all.
I think your 3701a don't "understand" hang
Hi again,
I am trying to get my DTMF to use RFC 2833 rather then inband, so that
I can utilize lower bandwidth codecs through my FXO.
After much tinkering I was able to get my gateway (wellgate 3701A)
configured to a point where I have some success, but no real joy.
I have configured the R
On Mar 16, 2006, at 3:24 AM, Aisling wrote:
Hi everyone, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but t
On Mar 15, 2006, at 6:36 AM, Stojan Sljivic - GDS wrote:
Hi,
I have downloaded an IAX softphone and tested the connection locally.
The sound is perfect.
How should I troubleshoot this IAX connection between these two
Asterisk
servers?
Is there some tool that can help in determining the caus
On Mar 15, 2006, at 4:20 AM, nik600 wrote:
hi
in my callcenter i start asterisk on server with asterisk_safe
command, after 4 days i can see that it is crashed 12 times, reporting
segmentation fault error...each time asterisk is correctly restarted
without loss of services but, is it normal?
On Mar 14, 2006, at 9:11 PM, Douglas Garstang wrote:
I discussed the native sounds with my boss the other day. We decided
not to use them because there's only 197 sound files out of a total of
>1200 installed on the system from asterisk and the asterisk-sounds
packages. We wanted to have a co
On Mar 13, 2006, at 8:57 AM, Thczv F. Thczv wrote:
On 3/12/06, Martin Joseph <[EMAIL PROTECTED]> wrote:
But what the OP wanted was a sulotion that together with the SAP3000
makes for something that works even when there is a blackout, since
the SPA3000 allows for failover to the FX
On Mar 13, 2006, at 9:25 AM, Thomas Johnson wrote:
We currently operate an MKC Communications Server for our small
company. We have 4 offices across Canada and calls to our toll-free
number are answered by our VOIP server and directed by the auto
attendant in the server office to the 3 satell
Another dumb question...
My asterisk system seems to have problems detecting hangups. I am
getting a LOT of voicemails with dialtone or silence.
I see over at asteriskguru.com there is an explanation of how to
configure for polarity reversal in zapata.conf?
Does zapata.conf have any functi
On Mar 13, 2006, at 12:31 AM, Wilson Pickett wrote:
At the moment, I can't seem to get more than one IAX client
registered behind NAT... am I correct in my above assumption or have I
missed something ?
I've used multiple hardware IAX phones behind NAT without a problem.
Is the asterisk
On Mar 14, 2006, at 9:04 AM, Keith Schmidt wrote:
I have 3 POTS lines that I want to use with Asterisk, I am looking at
prices for FXO cards and the cards with echo cancellation are really
pricey... is echo cancellation really worth it for a 3 or 4 line
system? Will I notice a difference wit
On Mar 13, 2006, at 12:00 PM, Bob McDowell wrote:
It depends
http://www.callcorder.com/phone-recording-law-america.htm
Thanks for the info!
12 states require, under most circumstances, the consent of all parties
to a conversation. Those jurisdictions are California, Connecticut,
Flor
On Mar 13, 2006, at 12:12 AM, Adrian Carter wrote:
BTW.. without sparking a flame war, and I have no idea how accurate the information is, but it seems that 'single party consent' applies as long as the recorded is not to be used for illegal purposes.
This means only one party (in this case the
On Mar 12, 2006, at 11:10 PM, Tomislav Parcina wrote:
Hi Dan!
Yes, that news group follows this mailing list. They head some problem in past few days. Now it's working.
gmane.comp.telephony.pbx.asterisk.user
It's totally hosed, as far as it appears thorough my newsgroup server... YMMV I gu
On Mar 12, 2006, at 10:35 PM, Adrian Carter wrote:
I'd teach the boss to appreciate recorded calls and just ensure they
are
secure.
In the US I think this illegal? Aren't you supposed to have some sort
of notification or beeping to indicate a recorded call to the other
party?
Just a tho
On Mar 12, 2006, at 1:31 AM, Simon Dorfman wrote:
Has anyone tried getting Speakeasy VOIP to work with Asterisk?
I just got Speakeasy DSL and am thinking of trying out their VOIP [1]
with
the hope that the quality/stability will be better than broadvoice.
I searched in the usual places (voi
On Mar 11, 2006, at 6:52 PM, C F wrote:
But what the OP wanted was a sulotion that together with the SAP3000
makes for something that works even when there is a blackout, since
the SPA3000 allows for failover to the FXS port from the FXO port
if/when there is no power to the unit. Which makes i
On Mar 11, 2006, at 6:25 AM, Whisker, Peter wrote:
The G.726 codec is the current Asterisk 1.2 version (revision 7221).
I am using G.711a (alaw) between a Sipura ATA and Asterisk at each end
of the link and am testing alternative codecs on an IAX link (not in
trunk mode) between the two Aste
On Mar 9, 2006, at 9:30 PM, Marc Archer wrote:
Hi All, This is probably a stupid question, but I’m trying to figure out if I Asterisk is in the middle of the media stream or not… Is there a command or something that indicates weather of not the two endpoints are talking directly? I am seeing mess
On Mar 7, 2006, at 7:02 AM, artifex maximus wrote:
Hi all!
I have the following setup:
Phone lines -> traditional PBX -> Welltech 3802
-> VPN ->
Asterisk -> Linksys PAP2/Welltech ATA-151 -> phone
There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2)
PBX extensions. Asterisk is
On Mar 7, 2006, at 9:51 AM, Giridhar Bandi wrote:
Hey thanks for the prompt response ( that's what i liked about this
list )
i was not able to start recording
i have pap2 box as clients and the dial plan of pap2 is as bellow
(*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.)
ca
On Mar 7, 2006, at 2:38 AM, Giridhar Bandi wrote:
ya i found it it *1 to start recording from the caller end
Also pushing *1 again stops recording.
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On Mar 6, 2006, at 10:32 PM, bmw suzuki wrote:
Hello all ... mY first ever post in here.
I am bit or (full) confused on what this program does.is it useful if
i have a alcatel pabx system.And i can bill my guests for their call
charges etc..
You will need to decide that.
can i use it on ca
On Mar 6, 2006, at 7:14 PM, Tom Vile wrote:
The firmware made thing much worse for me. Call waiting does not
work, I cant disable the local feature codes and DTMF transmission is
horrible as well. It has not locked up on me yet though because I
through it in the trash and plugged in my Sipura
On Mar 6, 2006, at 6:46 AM, Giorgio Incantalupo wrote:
Hi,
I have an asterisk 1.2.1 (on a debian sarge) box with a TDM400P card.
I connected the TDM400P to a grandstream 286 to use a VoIP provider.
It seems all right except for a little problem: one call every 30 is
made to a wrong number.
I
On Mar 6, 2006, at 6:46 AM, <[EMAIL PROTECTED]> wrote:
Hi friend,
I am running asterisk in production and it is being used by many
people using h323. I cannot afford to change all their configurations.
Also, the newer asterisk dosenot support inband for h323 properly.
Thats why I want two
On Mar 6, 2006, at 8:39 AM, Colin Anderson wrote:
http://www.voip-info.org/tiki-index.php?
page=Asterisk%20MacOSX%20Support
It works but it's bitchy as hell to run because of "root" issues in
OSX.
I wonder what the above "root issues" means?
I run it on my Mini. Zaptel is not supported. Y
Probably just me being dumb, but I am trying to update my asterisk to
the latest (1.2.5)
When I go to my /usr/src/asterisk I type:
make upgrade
make install
This seems to be doing it's thing, but when I type show version from
the console (after a restart) I still get:
Asterisk SVN-branc
On Mar 4, 2006, at 7:54 AM, The Asterisk Development Team wrote:
Asterisk 1.2.5 is now available for download on the ftp. See the
ChangeLog for details about what has changed.
Reading the changelog I notice the following... I suppose it should
say incorrect?
2006-02-17 01:55 + [r1030
I realized today that my call waiting isn't working properly. If I am
using the FXS attached phone and a call comes in the FXO, it just goes
directly to voicemail, with no indication (call waiting beep).
If I flash there is a second dial tone, and I can initiate a second
call.
If I am dial
On 3/3/06, Martin Joseph <[EMAIL PROTECTED]> wrote:
On Mar 3, 2006, at 11:35 AM, Tom Vile wrote:
I have had the same issue with Sipura as well, they gave me quite the
run around on 1 bad 2002 ATA and decided to just forget about it and
bought a new one to save time.
So they gave you
http://grandstream.com/BETATEST/HT488_496_386/
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