[EMAIL PROTECTED] wrote:
Time Warner provides an emta not an ATA and the technology is
different. You do not even need internet connection for that and runs
over their own private network through DOCSIS.
Who manufacturers that unit? Have you found a way to interface it to a PBX?
--
Chris
There are three possibilities to set the CallingPartyNumber (own number for
outgoing):
1) Set(CALLERID(number)=12345)
before Dial()
2) Dial(CAPI/contr1/12345:${EXTEN}/)
3) Dial(CAPI/contr1/${EXTEN}/d,...) and 'defaultcid=12345' in capi.conf
with this defaultcid you can set a number
Hello,
I have 2 channels in iax.conf
[iaxfwd]
type=user
callerid= Free World Dialup
inkeys=freeworlddialup
auth=rsa
context=incoming
qualify=yes
[iaxfwd-outbound]
type=peer
host=iax2.fwdnet.net
username=xx
secret=***
auth=md5
The problem is:
When I tell FWD to call me I have this
Thanks greatly for this. I will give it a go with these cards. I was trying
to use Diva ones before. ISDN was by far my preferred choice, if I could get
it to work...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tim Robinson
Sent: 23 February
Paul -
Let me know when you have the cards and if you need any help. Main
thing is to ensure that you have each card on a seperate IRQ. this is
ESSENTIAL! Unless the bios is able to assign specific IRQs to specific
cards it might be a bit of a fiddle. For £15 you can't go far wrong
Anton Krall wrote:
Maybe this is a stupid question but how to you enable debubg or logging on
spandsp? I see you can do that for RXFAX but when people tell you to enable
debug on spandsp, do they mean this with rxfax or how do you do it with
spandsp?
You can do it writing:
exten =
Solved the problem downgrading zaptel 1.2.4 to 1.2.3.
Mantaining the same configurations now everything works fine.
Regards,
_fangi_
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Francesco Angi
Inviato: martedì 21 febbraio 2006 14.35
A: Asterisk
Do you have any success receiving the caller id with your TDM400 FXO?
I have the same problem when I connect the GSM gateway to a SPA3000 FXO
line and thought this a Sipura's problem. On a phone connected to the GSM
gateway I can see the callerid, but not on the Sipura's PSTN line ...
this
I have a strange problem when calling some numbers with asterisk, I get
an hangup for busy condition even if the phone at the other end isn't busy.
I can route the calls via SIP to another carrier and then I have a SIP
code 486
or I can terminate them on digium cards (E1) and I have an Hangup
On Thu, Feb 23, 2006 at 02:45:25PM +0100, Armin Schindler wrote:
Hello Armin, hello List
I'm trying to get chan_capi working with asterisk from debian stable
(asterisk 1.0.7, the debian version number is 1:1.0.7.dfsg.1-2).
I managed to get it compiled by providing my own version of
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Garth van Sittert
Sent: 24 February 2006 07:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk hints
Hi Mike
I have build 18 on
Can the asterisk support a
coaching
function for the Supervisor to tap onto a call and coach the agent as
she speaks to the customer without the customer hearing it.?
Customer database
management softward
(or CRM) is this being included?
Best regards
Stephen
I checked the permitions and updated the ones with the wrong permissions.
No it is reading the number of messages correct, but as soon as I press 1 to
listen it stops again. So again, I checked the permissions on the
messagefolder but it seemed ok. I see now that another person on this lista
On Fri, 24 Feb 2006, Ralf Schlatterbeck wrote:
On Thu, Feb 23, 2006 at 02:45:25PM +0100, Armin Schindler wrote:
Hello Armin, hello List
I'm trying to get chan_capi working with asterisk from debian stable
(asterisk 1.0.7, the debian version number is 1:1.0.7.dfsg.1-2).
I managed to
Hello list,
Is there any way to use a2billing without the IVR for the sip/iax users. (authentication is done by the user id and pass as user registers with asterisk).
I want to dial the destination number to the asterisk. for example:
user dials,
exten =_011.,1,DeadAGI(a2billing)
system will
Well - a Sangoma Card won't bring you your money back. At least not
immidiately. ;-) And a expensive highend echo cancelling card is not a
good replacement for a relatively cheap TE405. So let's try to bring
your existing investion to work.
I presume you checked that your machine is working again
Hi,
I configured Buddy Watch function on my Polycom IP 601. It works well, until I
make a reload of Asterisk. After reload, if I give the show hints command in
Asterisk's CLI, it says that there are no watcher for the extensions that I
configured.
Before the reload in the CLI appears:
-=
This probably has nothing to do with your problem, but I had a problem
with similar symptoms, except asterisk was actually crashing whenever
I tried to access voicemail. It would sometimes say some digits, but
never got far (never got as far as the actual message). Problem
turned out, crazily
On Fri, Feb 24, 2006 at 10:43:31AM +0100, Armin Schindler wrote:
Interesting is, that I receive an INFO_IND *before* the CONNECT_IND.
This looks like an interesting variation of Austrian ISDN to me.
Maybe it is a variation of the ISDN line, but the driver should fix that.
Sending
Asterisk Sales wrote:
mailto:asterisk-users@lists.digium.com
Hello list,
Is there any way to use a2billing without the IVR for the sip/iax users.
(authentication is done by the user id and pass as user registers with
asterisk).
I want to dial the destination number to the asterisk. for
On Fri, 2006-02-24 at 10:54 +1100, David Ankers wrote:
Are you sure those switch figures are right? 16ms delay in the switch path
sounds a bit long. Cisco's mid-range switches like the 2950 have switching
times measured in micro seconds. Then again a 2626 procurve is only around
$700.
I meant
It must be microseconds that is being quoted, as even the 2626 that you
mention lists a less than 13.3 microsecond latency.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Ankers
Sent: Thursday, February 23, 2006 6:54 PM
To: 'Asterisk Users
On Fri, 2006-02-24 at 09:44 +0800, Ronald Wiplinger wrote:
My database machine is broken and I have to use another one.
I made somewhere mistake(s) and get now in the debug file:
[Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query: SELECT * FROM
sip_buddies WHERE name = '886'
[Feb 24
Hi,
I'm trying to use the spandsp fax-back facility and despite I can send and
receive faxes, this is not working fine. I would like to get a debug of what
is going on. I am using the flag debug, but I don't know if txfax is
generating any log information or where it is saving it. I just don't
Good morning everybody,
Can someone explain to me the interconnection
between
thesefour things: indications.conf,
SetLanguage(), zaptel.conf
and ring-back ? If
there is any !! :- )
I am having this case where some users cannot hear
ring back
from a DeadAGI script and it seems to be
Benchev is believed to have said:
Thanks Aldo,
No I do not have a manual and I don't believe such a thing
exist. Actually, that GSM gateway is a Dock-N-Talk kind of thing
with the exception that the handset is imbedded, so pretty much
no need of a manual.
Is your Grandstream a HT-488? If so you
I am using IP10s also
It was working fine, but you needed to go into telnet mode,
to activate the busy lamp, with the hint option ...
moreover, if you wanted to pick up the phone call,
then you needed also to add another telnet command to handle this pickup !
I know that swissvoice has now
Victor Alvarez wrote:
I'm trying to use the spandsp fax-back facility and despite I can send and
receive faxes, this is not working fine. I would like to get a debug of what
is going on. I am using the flag debug, but I don't know if txfax is
generating any log information or where it is saving
Victor Alvarez wrote:
is going on. I am using the flag debug, but I don't know if txfax is
generating any log information or where it is saving it. I just don't find
anything.
My line in extensions.conf is:
exten = ,1,txfax(/home/victor/testfax.tif|debug)
Asterisk's debug facilities
app_milliwatt is a nice tool for a quick check of the
line quality.
Anyway, hearing to that tone for more than a minute is
painful.
Does anyone know the opposite application, i.e. an
application, that hears and listens for a 1000 Hz
tone and displays the quality in any unit?
If not,
Here is the SIP transaction log. Caller called 7176 (Cisco 7960) from outside PSTN line, 7185(polycom 601, ip: 192.168.2.104) is the phone which monitors 7176.Reliably Transmitting (no NAT) to 192.168.2.104:5060:NOTIFY sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP
Hello,I'm new to Asterisk. I want to do the folloing job.I want to run Asterisk as a voip gateway to forward h323 calls to another gateway. my-gateway - Asterisk -- your-gateway h323 h323Is it possible to do this? If so, can anyone give me an idea
I was using IAX2 with ILBC and no trunking. I also set the
resyncthreshold=-1 to turn it off. Still had major jitter problems.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, February 23, 2006 6:44 PM
To: Asterisk Users
Rich Adamson wrote:
app_milliwatt is a nice tool for a quick check of the
line quality.
Anyway, hearing to that tone for more than a minute is
painful.
Does anyone know the opposite application, i.e. an
application, that hears and listens for a 1000 Hz
tone and displays the quality in any unit?
John Kelly wrote:
I'm looking to handle 3 PSTN lines with my Asterisk server. I have a
Digium TDM22B and Sipura 3000. The Sipura works great, but the TDM22B
seems to have terrible problems with my board---virtually all
peripherals need to be disabled in BIOS, and then there is terrible
Hi Jean-Marc
I tried removing the call-limit setting. It still doesn't work. I am
using a SNOM 360 to monitor the line status'.
Do I still need to activate the busy lamp on the IP10S' or is this only
if you want the IP10S' to monitor the hints?
Garth
Jean-Marc Salsa wrote:
I am using
Hello Asterisk community.
We have a small User-group in Melbourne Australia.
Recently I brought up the issue of STANDARDS for dialing Applications on
a PBX.
This generated some interest but also the fact little has been done on
this topic.
Below is a rundown of our THREAD. (start from
Is there any particular reason for the native file format stuff to be in
asterisk-addons as opposed to that code being merged into trunk?
It isn't. You are mis-interpreting the information in this thread (it's
been unclearly stated anyway).
The only portion that is in asterisk-addons is
Aha, micro seconds in networking terms is normally written usecs or us
(actually it's the greek letter mu as in ulaw) rather than ms which are
milliseconds seconds - what had me puzzled was that it was stated that this
could harm the voice path!
The difference can also cause unnecessary delays
I would like to capture the lat/lon coordinates from a GPS-enabled cell
phone or PDA. Is this possible? Must I subscribe to this information
from the cellphone network provider, or can I capture it without charge?
What devices will broadcast the coordinates? Is there a device that
hello all, do i must must see Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface when in install TDM2424E card i think so but i can not see this in lspci is there is software tools to make sure that my motherboard have pci 2.2 and see TDM2424E see it right b4install zaptel driver and did this
On 2/24/06, Marco Maiolini [EMAIL PROTECTED] wrote:
Hi,
I configured Buddy Watch function on my Polycom IP 601. It works well, until
I make a reload of Asterisk. After reload, if I give the show hints command
in Asterisk's CLI, it says that there are no watcher for the extensions that
I
On Friday 24 February 2006 07:56, Paul wrote:
Maybe the first approach should be to setup a test extension for
recording the tone. The idea is to get best resolution possible in real
time. Then process it as much as needed to get the info you want. Such
an approach would give you more
Ive been testing how to receive faxes using TDM400P cards and so far, after
playing with gains, echocancell and echotraining on zapata.conf.. Ive ha dno
luck, faxes come in as garbage or broken or with blank lines.
Anybody has successfully done this? Any tips.. Also I have some ideas:
1.
Hi All,
Ihave installed a Digium TE110P card on an
Asterisk 1.2.1 system. Its connected directly to the PSTN. But I am unable to
make outbound calls on the zap channels. The light on the card is green.
Asterisk CLI shows all 24 channels when I give the command 'zap show channels'.
I also
On Thursday 23 Feb 2006 20:34, Colin Anderson wrote:
It's stupid. Don't ever connect 2 different building with copper.
Just wait until you get some kind of lightening hit or electrical
fault, but make sure you are no where near it. Use fibre.
Thanks for the reply. Unfortunately, the conduit
Hi There,
this is very much dependent from your provider, your PDA/cell phone and the
network. For GSM networks in Europe e.g. the providers have different types
of information available through the CB channels of their base stations.
This data can always be read and stored in your
I have seen some very expensive switches fail. Nice thing about lower
cost devices is that you can afford to have spares. If you stick to a
standard way of labeling and connecting wires you can use good open
source monitoring software to detect switch failure. If you allow people
to randomly
Andrew Kohlsmith wrote:
On Friday 24 February 2006 07:56, Paul wrote:
Maybe the first approach should be to setup a test extension for
recording the tone. The idea is to get best resolution possible in real
time. Then process it as much as needed to get the info you want. Such
an approach
Hi all, this is another post about this problem.
I installed from scratch a new Suse Linux 10.0, with latest stable
asterisk.
Moreover I add the lines to /etc/udev/rules.d/50-udev.rules, in order to
let the driver create the /dev/zap...
When I plug into usb port my TigerJet adapter, I see
On Sat, 2006-02-25 at 00:21 +1100, David Ankers wrote:
Aha, micro seconds in networking terms is normally written usecs or us
(actually it's the greek letter mu as in ulaw) rather than ms which are
milliseconds seconds - what had me puzzled was that it was stated that this
could harm the voice
On Thursday 23 February 2006 13:57, Bob Goddard wrote:
It's stupid. Don't ever connect 2 different building with copper.
Just wait until you get some kind of lightening hit or electrical
fault, but make sure you are no where near it. Use fibre.
That's a great rule of thumb, but the reality
Hi!
I am using tdm400 cards for receiving faxes. It worked quite out of the box. I
installed spandsp for the rxfax application only.
I use it as phone/fax switch:
All incoming calls are answered automatically to listen whether its a fax or
not. If it is a fax, the call is forwarded to the
udev drove me absolutely bat-shit in this regard; udev is a horror in many
respects. Here's how I solved the problem, reliably:
I run this script at boot-time:
#!/bin/bash
mkdir -p /dev/zap
rm -f /dev/zap/ctl
rm -f /dev/zap/channel
rm -f /dev/zap/pseudo
I currently use asterisk version 1.0.10 with AMP 1.0.010,
our setup is APX 8000 - Interaction SIP Proxy 3.0.013 - asterisk
server. When I use Asterisk version 10.0.10 everything works
perfectly, however when I use 1.2.4 I lose the ability to receive calls from the
PSTN. All I get is the
Nitin Joshi wrote:
Hi All,
I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its
connected directly to the PSTN. But I am unable to make outbound calls
on the zap channels. The light on the card is green. Asterisk CLI
shows all 24 channels when I give the command 'zap show
On 2/23/06, Geoff Manning [EMAIL PROTECTED] wrote:
How would you categorize a Yellow Alarm sensed by the Asterisk side in a Legacy PBX integration?We have a Mitel SX200 connected to an Asterisk(1.2.4) with a TE110P.Twice today (first time in over a month) we received a Yellow Alarm on the TE110P.
Chris Modesitt wrote:
I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is
APX 8000 - Interaction SIP Proxy 3.0.013 - asterisk server. When I
use Asterisk version 10.0.10 everything works perfectly, however when I
use 1.2.4 I lose the ability to receive calls from the PSTN.
On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote:
Its my understanding the cell phone coordinates are sent to the cell phone
provider and their equipment reads (and holds) that data. Its not part
of any data available to you in any form unless you talk to the cell
provider and
On Wed, 22 Feb 2006 18:02:27 -0800, mustardman29 wrote:
Just the person I have been looking for. If you don't mind, would it be
possible to get your opinion on feature for feature comparisons between the
501 and 480i CT(not including cordless phone).
Things like programmable buttons, display,
Chris Modesitt wrote:
I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is
APX 8000 - Interaction SIP Proxy 3.0.013 - asterisk server. When I
use Asterisk version 10.0.10 everything works perfectly, however when I
use 1.2.4 I lose the ability to receive calls from the PSTN.
I wrestled with this for a long time, as have many others and it just
doesn't work with spandsp and asterisk alone.
Use iaxmodem and hylafax in conjunction with asterisk... it works like a
champ. I have a single POTS line coming in so I get voice fax with a
single number using fax detect.
In the US, Sprint's CDMA network will do the fancy GPS+AFLT business,
but like someone else mentioned, it only sends the location data back
to Sprint's network. There is an API that you can use to access this
data for your handsets, but you have to pay some amount of money for
each location fix.
Hi Olle,
Will u be there for the speech of Jan Janak?
If yes, you will find a guy, 1m83, with a bear and a red suit, it's me.
You also can call me on my mobile to fix the voip beer (0032495283361).
We will try to have Jan and other guys
Olivier
___
no chance, also with your scipt
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
1 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
Done..
They don't show much but they do show some problems with lost lines or
something
Thx Bartosz
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Bartosz Piec
|Sent: Friday, February 24, 2006 2:54 AM
|To: Asterisk Users Mailing List -
maybe you didn't want suggestions, but too bad :).
My favorite up until recently was the polycom 501 and I found it was
good quality and clear calls and priced well. but the production of te
phone is slowing down so I bought a few linksysspa941. and iVll
tell you I have a new favorite
do you have a defaultcontext=something parameter in sip.conf [general] section?? If not, the default is... em default
RegardsOn 2/23/06, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,I am getting repeated messages in my logs with the following:*Feb 23 07:56:11 NOTICE[2470] pbx.c:
Well, I have the same effect on my TDM as in the E1 using unicall... Faxes
get here as garbage :(
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Rich Adamson
|Sent: Friday, February 24, 2006 7:28 AM
|To: Asterisk Users Mailing List -
Polycom does support Asterisk, Asterisk Business Edition.
-Original Message-
From: Michael Graves [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 23, 2006 6:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] What business IP phone to use
Any modification made to zapata as far as echo and gains?
Should echocancel be on or off?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
|Sent: Friday, February 24, 2006 8:25 AM
|To: Asterisk Users Mailing List - Non-Commercial
you need to set a TRANSFER_CONTEXT, either for the transferer or
transferee channel. I dont know why, but res_features give priority to
the transferee TRANSFER_CONTEXT, if not found, then use the transferer
TRANSFER_CONTEXT. That context is used to match the extension to dial.
So you can set this
In the UK this is common; several websites enable you to track a cell phone
online:
http://www.traceamobile.co.uk/
and another:
http://www.followus.co.uk/
Works the same way that Steve stated... The police here in Australia have
been using this since the late 90s.
Interesting article:
Thank you Lee, Dave, Rich, Joel and of course also Kevin. Between your
various messages I finally understand what's happening and how it works,
and have actually converted everything to alaw, ulaw, slin and gsm and
am not actually using the mp3 side of things at all anymore. The
difference is
I have talked with of a couple people(don't remember their names) who
had this developed on a contract basis for the 1.0 Asterisk code tree,
they did not want to release it to GPL because of how much it cost
them and the fact that their code supposedly won't run on 1.2, but it
is technically
On 2/24/06, Doug Lytle [EMAIL PROTECTED] wrote:
T1s require a D (Data) channel, unless connecting to a channel bank, Itshould be 23 voice 1 data.Also, I would strongly suggest moving to
1.2.4
Not totally true. A PRI is 23b 1d. A DS1 (US) is a 24 channel circuit.
Nitin- When you stop/start
Mike,
Were you able to get this working?
Even after with a entry in the dialplan.xml does not work for me.
Thanks,
Ken
On 6/20/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED] wrote:
Andrew,I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide?When you say mapped, dou
Andrew Kohlsmith wrote:
What is being discussed here is basically what I was planning on doing for an
automatic VOIP quality check. Using miliwatt and analyzing it for
pop/jitter/etc as well as sending other known waveforms and comparing what
was received to what was expected and coming up
Date: Fri, 24 Feb 2006 14:56:54 +
From: Steve Kennedy [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] GPS-enabled cell phone/PDA
On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote:
Its my understanding the cell phone coordinates are sent to the cell phone
provider and
Hi list,
I got a question:
When I try to ChanSpy a SIP channel I only listen one
channel, for example,
I call from 302 extension and I have two active channels:
SIP/r1-voip-1b7b
(None)
Up Bridged Call(SIP/302-f1f1)
SIP/302-f1f1
[EMAIL PROTECTED] Up Dial(SIP/[EMAIL PROTECTED]|4
Anton Krall wrote:
Well, I have the same effect on my TDM as in the E1 using unicall... Faxes
get here as garbage :(
I really would like to see sometime some audio recordings made by
IAXmodem for people that had problems with TDMs and faxing with rxfax/txfax.
Not that I have some hope of
Anton Krall wrote:
Any modification made to zapata as far as echo and gains?
As a rule, don't let anything manipulate the audio at all... even echo
cancellation. That said, I have seen situations where gain had to be
increased.
Should echocancel be on or off?
Off, most definitely
Am Friday 24 February 2006 16:48 schrieb Anton Krall:
Any modification made to zapata as far as echo and gains?
Should echocancel be on or off?
i have echocancel switched on, faxdetect is on, rx- and txgain is not used.
(commented out).
my var/log/messages says:
Found a Wildcard TDM:
Anything under 1ms is so far below the threshold of perceivable sound
quality, echo, delay etc. that it's a mute point to discuss IMHO. Not even
in any cumulative effect it may have.
I can certainly see the advantages of SNMP for remote troubleshooting but
hard to justify for small offices with
Brian Roy wrote:
Not totally true. A PRI is 23b 1d. A DS1 (US) is a 24 channel circuit.
Nitin - When you stop/start asterisk does it load all 24 channels? Any
errors? How about zap show channel 1 in the CLI?
Learn something new every day.
Doug
Paul Tinsley wrote:
Does anyone know of a way to specify what extension is dialed when 0
is pressed in the voicemail system. I have a situation where there is
more than one secretary and they want the 0 to redirect to the
appropriate secretary for the two groups of people.
So an example
I'm having difficulty with an Asterisk system. The external party has
very good call quality, but the internal party hears clipping and drop outs.
The WAN comes in from the Cisco IAD and into a LAN switch (DLink
DGS-1005D w/ 802.1p) where the two public IPs are switched to different
devices.
Hi, i have several incoming sip peers (mostly ciscos) , with 1.0 i
always registered them like this:
register = @prepago-in
[prepago-in]
type=friend
host=192.168.10.120
context = from-external
dtmfmode=rfc2833
insecure=very ; required for incoming FWD calls
Now with 1.2.4 it doesnt work any
Some more recent phones have the possibility to be connected to seperate
GSM-boxes. E.g. there is a plug-in for the (older) Nokia 9210(i)/9290(i)
Communicators and most of the Symbian phones with Bluetooth support can be
connected to any Bluetooth-enabled GPS-mouse ...
I think, getting the
Are you sure you're supposed to be using EM?
On Feb 24, 2006, at 5:39 AM, Nitin Joshi wrote:
Hi All,
I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its
connected directly to the PSTN. But I am unable to make outbound calls
on the zap channels. The light on the card is
Hi,
Okay but then how do you transfer across contexts then?
Thanks
Moises Silva wrote:
you need to set a TRANSFER_CONTEXT, either for the transferer or
transferee channel. I dont know why, but res_features give priority to
the transferee TRANSFER_CONTEXT, if not found, then use the
Aha, micro seconds in networking terms is normally written usecs or us
(actually it's the greek letter mu as in ulaw) rather than ms which are
milliseconds seconds - what had me puzzled was that it was stated that this
could harm the voice path!
The difference can also cause unnecessary
it seems im not undestanding your question then. Could you provide a practical example?On 2/24/06, Chuck Bunn
[EMAIL PROTECTED] wrote:Hi,Okay but then how do you transfer across contexts then?
ThanksMoises Silva wrote: you need to set a TRANSFER_CONTEXT, either for the transferer or transferee
Hello,
I'm posting this to the list in case others run into the same issue.
I've recently been connecting * to a legacy Avaya InDEX switch over
E1 ISDN PRI here in the UK. Everything was working OK, except that DTMF
digits were not being recognised by * when sent by the Avaya switch
Hi,
We're having problems dialing out to Asterisk from
our Grandstream GXP-200 phones. About 2 of 3 times, when we dial, nothing
happens. Looking at the console in max debug mode, there are no messages
except the following:
Feb 24 10:29:20 WARNING[2475]: chan_sip.c:1208
retrans_pkt:
Hi,
On the general section of my sip.conf I had a
disallow=all.
Then I put disallow=all, allow=g729, allow=ulaw on
my users.
It didn't work until I removed the disallow=all
from the header.
I know disallow=all in the header is totally
useless in this case (since I have it for every
Hi,
I support multiple context on one asterisk server. I have a situation
where there is a spa that has seperate voicemail and extensions and a
resturant on the same campus that has different extensions and
voicemail. They both use the same asterisk server but I do need the
ability to
what zap device are you using?
IIRC disalbing the vpmdtmf on a 406 or 411 might help you. I think
it's done in wctxx4p.c
On 2/24/06, Matt King [EMAIL PROTECTED] wrote:
Hello,
I'm posting this to the list in case others run into the same issue.
I've recently been connecting * to a
I am trying to use ImportVar to get some information out of a SIP/ZAP
channel. I cannot seem to find an example of the syntax, or what
variables I can access.
Basically, I would like to output which person is being called. i.e:
SIP/25 calls SIP/21. 25 executes a macro, and the result is
Steve,
You wrote this referring to monitoring a call in Asterisk, how about from an
IP phones LCD display screen:
1. go to www.google.com
2. type asterisk monitor application
3. click on the first result
4. read and implement
5. google is your friend
I hope I made myself clear too ;-P
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