Doesn't seem to help. I did it early yesterday morning and have
another 'stuck' call this morning
Does anyone have any other ideas on what I can do to correct this?
thanks
Shawn
CLI core show channels
Channel Location State Application(Data)
DAHDI/8-1
Right, this is how I expected it to operate. My prior question though was
regarding the 'T1 over Ethernet' scheme someone mentioned which ran full
throughput all the time.
That is true. If you're doing a clear-channel or pseudo-wire T1 over
ethernet you will always be using 1.54 Mbps
don't, the
channel appears to be
in-use forever.
Thanks
Shawn
The setup is fairly straight-forward
Extensions
[in-phone2]
exten = s,1,Answer()
exten = s,n,Noop(CALLERID(name))
exten = s,n,Noop(CALLERID(num))
exten = s,n,Dial(SIP/cordless2,25,tTo)
exten = s,n,Hangup
[out-phone2]
exten = _[*#0-9]!,1
I'm setting up an asterisk server to extend several extensions from a mitel pbx.
I'd like to display the caller id that I receive from t he mitel pbx
on the sip phone. The mitel
PBX person has setup the PBX to send be callerid, but I don't see it.
I've set chan_dahdi up with
usecallerid=yes
Yes, I'm talking about mid-call.
I do have rtptimeout and qualify set, both to 30 seconds, which should be
plenty of time.
I set them both because if a phone moves out of range, and never comes back,
asterisk was keeping the channel open way to long.
On Wed, May 4, 2011 at 7:50 PM, Matt Riddell
I have a situation where we have an asterisk box that is extending several
Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310). Everything works great, except
when the cordless
phone walks out of range of one access point and into range of another
(cisco 1100 series APs).
I've been
I have a situation where we have an asterisk box that is extending several
Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310). Everything works great, except
when the cordless
phone walks out of range of one access point and into range of another
(cisco 1100 series APs).
I have
I have 2 separate Asterisk servers that are both exibiting this problem. 1
has a 4 port
FXO digium card, the other an 8 port.
For some reason when the machine reboots, the dahdi drivers are not properly
loaded. Then asterisk
ends up starting without dahdi support. I've tried everything that I
You have a powerful name, yourself.
sk
On Wed, Jan 20, 2010 at 9:38 AM, Sean Bright sean.bri...@gmail.com wrote:
On 1/17/2010 3:25 PM, shawn bright wrote:
Hey all,
i love your name, btw.
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Hey all,
We have been using a TDM400 card at work to provide our IVR.
We we have upgraded our server and now require the same capability, but on a
card that goes into a PCI Express.
Any suggestions would be greatly appreciated.
oh, and it has to work with the zaptel drivers for linux.
thanks
asterisk directly on that appliance. You would probably save power
consumption versus a new server or even the old server currently in
use.
On Sun, Jan 17, 2010 at 3:25 PM, shawn bright sh...@skrite.net wrote:
Hey all,
We have been using a TDM400 card at work to provide our IVR.
We we have
Softphones out of the question?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Beckman
Sent: Tuesday, March 10, 2009 10:24 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] 1.4.23 +
will go out? I'd like
to have
speed-dial buttons that will go out on line2 instead of line 1. Anyone know
if this
is possible?
Thanks
Shawn
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Depending on how many faxes you have coming in a simple fxs/fxo card
will do the trick .. either Sagnoma or Digium or any others you could
also use any decent ATA.. Asterisk only needs to know its a fax and what
dialed number it came on to route it to the correct fax machine.
Asterisk would
anyone know if this is possible, and if so
how to do it?
Thanks in advance
Shawn
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to initiate
the second ?
Do they dissapear ?
thanks
shawn
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to svn instead of CVS.
thanks much.
shawn
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on the Asterisk
server. No attempted subscribes/logins/registers.
Thanks in advance!!
--Shawn
[EMAIL PROTECTED]
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then don't answer the phone professionally since we think that it
is our employee calling us.
Thanks!
--Shawn
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if the phone is disconnected.
This is for an IVR application where we phone out to our customers if the
status of one of their machines changes
thanks for any tips,
shawn
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We have a PRI line setup on an asterisk box.
I want to provide a couple dial-in lines for internet access from our remote
offices.
The remote offices only have analog lines.
I thought I could do this with the ZapRAS or PPPD functions in Asterisk, but
the more I read this is only for ISDN
messages.
Does anyone know if this is possible with the Asterisk.NET interface?
Or does anyone know of another way to accomplish my needs?
Any advice is greatly appreciated.
--Shawn
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this using FastAGI or via the Manager API? I
have yet to fully understand these two methods of interface along with the
differences of each.
Could someone please give me some advice?
Thanks in advance!
--Shawn
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I had the same problem.
Checking voicemail via the phone was perfectly normal but the email
attachments were so quiet we had to turn the computer volume all the way up
along with the speakers amps just to make the attachment understandable.
Then just wait until someone forgets to turn the volume
Beware of the SPA-3000, we had a nightmare trying to get rid of echo issues
with it on the PSTN connection. We still haven't got it quite right even
after trying all kinds of settings and firmwares.
-Original Message-
From: Jay R. Ashworth [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October
Stan,
I agree with the comment below, we switched from analog lines to a PRI and
it's not always as reliable as some people think. We are in a somewhat rural
location and we have outages regularly. 1-4 hour outages every few months
are not uncommon for us. Outages of 60 seconds or so are even more
the call to one of our remote employees (home or cell).
Loss of revenue was a good idea, but don't think it's the cause here.
--Shawn
-Original Message-
From: James [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 28, 2006 5:34 PM
To: Asterisk Users Mailing List - Non-Commercial
with a PRI or do you have to have some
other type of PSTN connection such as SS7?
Thanks!!
--Shawn
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Im told by Adam below that I can use a Speed Dial to accomplish this.However, I dont know how to map a speed dial to the key.I know the syntax for mapping a function to it ( IP_500 key.IP_500.31.function.prim=BuddyStatus/ )However, I dont know how to do a speed dial.Any one out there
appreciated.
Thanks!
--Shawn
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here ?
some of the code is from an example app that i found on the internet.
There is something in this code ( i suppose ) that limits me from getting
anything but an integer as a response from here
if you have read this far, thanks for your time.
shawn
at 10:31:30AM -0400, shawn bright wrote: lo there, i am running a python agi script that gets a DTMF number from the user and passes it back to the script.Any reason you're not using Read for this?
It works fine with numbers, but if they enter a star (*), it doesn't want to play.Could you please
i couldn't find any docs on it either, found some scripts that just use pythonto communicate with asterisk. Thats kinda why i have been eying ruby. However, i did get some stuff to work from some examples on the net, if you like i can send them to you.
-shawnOn 8/14/06, Douglas Garstang [EMAIL
Hey there all,
i have an app that calls our customers when the status of their
machines change. I am using a Zap channel to dial out on an analog line.
I do this by putting a drop.call file in /var/spool/asterisk/outgoing
and connecting it to an extension which fires up a python script.
the
well, i dont know about how to get a digital line out of the office,
but that would be cool. Clean up some other stuff too. Ah-well, guess i
will settle for door number 1. The press 1 to continue loop.
thanks, guys, let you know how it turns out.
shawnOn 8/10/06, Eric ManxPower Wieling [EMAIL
here here !
skOn 8/10/06, Jerry Geis [EMAIL PROTECTED] wrote:
It would be great if some swift asterisk coder would write a little applicationthat could monitor the channel for call progress.so on cases of busy, ring (with timeout) and TALK detection an appropriate response
could be taken for these
[EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],shawn bright [EMAIL PROTECTED]
wrote: i have a python script that drops a call.call file into the /var/spool/asterisk/outgoing directory the extension it points to reads like this [outboundmsg1]
exten = s,1,Answer exten = s,2,Wait(3) exten = s
closer now.
-skOn 8/8/06, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],shawn bright [EMAIL PROTECTED]
wrote: ok, here is the text written by my script in a file called drop.call in /var/spool/asterisk/tmp. After the file is created, it is moved to /var/spool/asterisk
um, i dont understand, could you elaborate on that ?thanksOn 8/8/06, Tzafrir Cohen [EMAIL PROTECTED]
wrote:On Wed, Aug 02, 2006 at 05:00:29PM -0500, shawn bright wrote: Lo there,
i have an app that needs to initiate a phone call on a zap channel. i have been able to test it out ok with the method
Hey there,The main reason i got into ruby was for RAGI ( the asterisk api ). From the tutorials, there is a lot of emphesis on using it with Rails. I was wondering if
it is possible to use Ragi without having to set up rails ? We have a small IVR server, and that is the computers sole
nothing to do, you only need a programming language, notnecessarily a framework.RegardsOn 8/7/06, shawn bright
[EMAIL PROTECTED] wrote: Hey there, The main reason i got into ruby was for RAGI ( the asterisk api ). Fromthe tutorials,there is a lot of emphesis on using it with Rails. I was wondering
Hey there,i have a python script that drops a call.call file into the /var/spool/asterisk/outgoing directorythe extension it points to reads like this[outboundmsg1]exten = s,1,Answerexten = s,2,Wait(3)
exten = s,3,AGI(server_dialout.py)exten = s,4,Hangupthe problem is that when the script is run,
is this?
Zaptel, SIP, IAX, FXO, PRI??
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
shawn bright
Sent: Monday, August 07, 2006 9:49
PM
To: asterisk mailing list
Subject: [asterisk-users] agi
script runs even if no answer
Hey there,
i have a python script
this project.
So i may, for consistency sake in my larger scale app, learn twisted anyway.thanks.shawnOn 8/6/06, Shidan
[EMAIL PROTECTED] wrote:np, but in general its well worth the learn tho if you like python ;)
On 8/3/06, shawn bright [EMAIL PROTECTED] wrote: Thanks for the reply Shidan, i have looked
ok, this seems like a workable solution. i will give it a shot tomorrow at work.thanksskOn 8/3/06, Benjamin Stocker
[EMAIL PROTECTED] wrote:2006/8/2, Andy Kuo
[EMAIL PROTECTED]: Hi, Can you give a quick example on how to query an EXTERNAL database?Create a AGI Script. It may take actual
a python script.i need to pass asterisk the phone number and then a couple of files to play.if anyone can tell me how to pull this off, or could post a link to some good doc or how-to,
i would greatly appreciate it.thanks- shawn
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[EMAIL PROTECTED] wrote:
Check out http://starpy.sourceforge.net/ if you have any questions let me know.---ShidanOn 8/2/06, shawn bright [EMAIL PROTECTED]
wrote: Lo there, i have an app that needs to initiate a phone call on a zap channel. i have been able to test it out ok with the method of dropping
Hello there all, i am using an agi python script. It is kinda from an example in the ATOF book ( O'Reilly)it simply answers the phone, receives 5 DTMF digits, and writes those digits to a text file.
however, it isn't working.The script is in python, and i have stderr writing out some debug
, not remotely connected Asterisk sessions(asterisk -r)MATT---On 7/29/06, shawn bright
[EMAIL PROTECTED] wrote: Hello there all, i am using an agi python script. It is kinda from an example in the ATOF book ( O'Reilly) it simply answers the phone, receives 5 DTMF digits,
and writes those digits
Hello there,i am a newbie here, but i have managed to get asterisk up and running with one zap channel, and various tutorials with dial plans.Cool. What i need to do is a little more complex though. We monitor field equipment for farmers, right now we store their info in a mysql database, and
it somehow.thanksskOn 7/28/06, shawn bright
[EMAIL PROTECTED] wrote:Hello there,i am a newbie here, but i have managed to get asterisk up and running with one zap channel, and various tutorials with dial plans.
Cool. What i need to do is a little more complex though. We monitor field equipment
can do to make the colorized version
work ok on Console 8 or 9(which ever one its at)?
Thanks,
--Shawn
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Hi all,
We are having an issue with Random Disconnects wit our PRI
connection.
We are going into a T100P card from a Cisco IAD.
Below is a copy of our PRI Debug.
We will be talking on the phone and all the sudden the line
will go dead. It acts as if the remote party hungup. However,
I have been unable to find any documents on what exactly the PRIEXCLUSIVE
setting is used for in zapata.conf
Does anyone know what it does?
Thanks,
--Shawn
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Does the CALLERID(all) also set the ANI Information, or
does the Set CALLERID(ani) also have to be called?
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Hi,
I am fairly new at working with Asterisk.
I am having a call quality issue that I really need to get
ironed out before we go to rollout the system in a week.
Any help would be greatly appreciated!!! Even if it is just
pointing me in the right direction.
My current setup:
I have
We are getting ready to deploy Asterisk on a Dell PowerEdge
1600SC Server.
We have a TE110P Digium card. I noticed on Digiums website
that there are some compatibility issues with this card on this machine series.
Does anyone know what these issues are?
Thanks,
--Shawn
Never try upgrades half-asleep and 1/4-knowledgable!
Got a link from a friend about the FLITE TTS that was rewritten to work
really well with Asterisk. So I downloaded and installed it on my 1.0.9
server - oops. So, I downloaded Asterisk 1.2.7.1 did the proper install
process, got all
, 2006-05-11 at 14:30, Shawn Porter wrote:
Never try upgrades half-asleep and 1/4-knowledgable!
Got a link from a friend about the FLITE TTS that was rewritten to work
really well with Asterisk. So I downloaded and installed it on my 1.0.9
server - oops. So, I downloaded Asterisk 1.2.7.1
Ravi,
Take a look here. http://www.voip-info.org/wiki-Asterisk+auto-dial+out
I would think that for what you are doing use a cron job and a shell script.
Shawn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ravi
Shankar
Sent: Friday, December 23, 2005
Would
someone be so kind as to point out what stupid little mistake I have made.
I thought I did everything according to the setup page but I fail to
register.
HOSTS
file contains
147.135.8.128 sip.broadvoice.com
SIP.CONF
[general]context=iaxclients; Default context for incoming
Thanks Steven
Works great.
They should put a little more detail in the setup page as to where you get
that password!!
very difficult to figure to that out in the wee hours of the morning.
Shawn
-Original Message-
From: Steven Job [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 20, 2005
I have been wondering the same thing. I would like to be able to link 2
channels inside an AGI script.
Also, a way to send variables back-and-forth.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Senyo
Sent: Tuesday, December 20, 2005 2:33 PM
Robert,
This configuration is working fine for me (In ontario with Bell Canada)
dring1 is the 2nd ring pattern on our line, it is a double-ring
dring3 is the regular ring, which I wanted to ignore but since you cant do
that I just send it to a wait loop
ZAPATA.CONF
[channels]
usercallerid=yes
mistakes/assumptions.
Shawn
P.S
Contarra Envox I know, Asterisk I am learning.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Serge
SchumacherSent: Tuesday, December 20, 2005 4:23 PMTo:
Asterisk Users Mailing List - Non-Commercial
my own
criticism. I just talked with a friend about erlang tables.
completely blows away all the stuff I just wrote below...
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Shawn
PorterSent: Tuesday, December 20, 2005 4:47 PMTo
hear echo at the first moment a call is placed, but it
should completely disappear in a few seconds.
--
Shawn
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Altus,
Just looking over the voip-info wiki
http://www.voip-info.org/wiki/Asterisk+dimensioning it seems you have hit
the limit of h323.
about 1/3 way down won't be able to run more than 20-25 decent quality
calls
Shawn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
oops, typo!
http://www.voip-info.org/wiki/view/Asterisk+dimensioning
-Original Message-
From: Shawn Porter [mailto:[EMAIL PROTECTED]
Sent: Friday, October 21, 2005 10:08 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] how many oh323
Altus
Sorry to bug
everyone with such a silly thing, but I am not having the best of mental days
today.
For some
reason I am unable to make calls from my Diax to my * box (same LAN)
as you can
see by the CLI output below I am registering and authenticating but unable to
call in. Yet, I can
and going to a
wildcard.
I appreciate any feedback, as it will end up being my job to install and
configure the server and I am not looking forward to it.
Shawn
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I have just installed spandsp-0.0.2 onto my Asterisk 1.0.9
I am using an old Intel 536EP (actually found drivers that work)
BUT...all my faxes are coming in landscape mode
Has anyone come across this?
any fixes?
Shawn
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in portrait style (the map image should be
8.5 wide and 11 long)
I can't take the old fax machine offline until I get this resolved. If
anyone has any ideas I am open to suggestion.
Shawn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Shawn Porter
Sent: Tuesday
-info.org/wiki-Asterisk+tips+ivr+menu
Shawn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Angus Comber
Sent: Friday, September 30, 2005 1:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Best way to create IVR/voicemail
Does anyone know of a provider that
a) allows/works using Asterisk
b) provides local DIDs to the Newmarket/Aurora area?
thanks
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I don't have your problem with not hanging up. And I do get a dial
tone. It even stutters when there is voicemail. I have an older one,
BT-100 I think.
I do have a few other beefs with it though:
- The display backlight times out too fast normally. I wish it would
stay lit for a few minutes
.
we are experiencing about a 6-8second delay.
I read about it possibly being timestamps on the packets, I did an ethereal
dump but really have little idea what I am looking at.
Any thoughts/ideas/suggestions would be appreciated.
Shawn
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On 9/25/05, Anders Svensson [EMAIL PROTECTED] wrote:
There is also www.talkycallshops.com
That looks interesting. Do they offer iax service or sip only? Do
you have any .conf example?
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On 9/20/05, Nathan Pralle [EMAIL PROTECTED] wrote:
database on an incoming call? Much head smacketh ensued, and as I made
Thou hast confused the present tense with the present participle.
Thou couldest have written smacketh head smartly but perchance it
is better to write there was much
right to call an internal extension.
Im sure
both of these are quite simple, I have probably missed some little thing in my
frustrated state.
Thanks for
any help.
Shawn
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, of course, my zaptel drivers do not work and therefore my asterisk
does not work.
any help would be greatly appreciated..
Shawn
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Brad,
I posted a similar question on voipuser, no response yet, but I ended up
making a separate extension
Its not perfect, but it does technically ignore the call.
[Home]
exten = s,1,Wait(30)
exten = s, 2, Hangup
Shawn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Is there a way to tell asterisk to ignore an incoming call?
I am using distinctinveringdetection and I am only interested in answering
calls
on the 2nd number. Any call to the main line should just be ignored.
right now I have a context set for dring2 cadence 0,0,0
exten = s, 1, wait(30
exten =
So has anybody got one of these?
http://www.amazon.com/exec/obidos/ASIN/B0007LQQUK/qid%3D1106972010/sr%3D11-1/ref%3Dsr%5F11%5F1/102-1529886-6420131
I'm thinking that it should be possible to connect it directly to an
Asterisk box and not use their software, as long as there was Linux
support
is possible? Please note as well that only a portion
of the phones would be near a computer.
If not, perhaps some other method of contending for the lines is
possible, maybe with an auto-retry and dial-back once a line becomes
available?
--
Shawn Iverson
Technology Associate
MCSA, Linux+, Network
Prospective user question
What is the simplest/inexpensive board to use in order to be able to
receive faxes in Asterisk.
I have a couple of cards I bought off E-bay think they were TX-1000 (or
supposed to be anyways)
but I assume I need some form of fax card though.
Thanks
caller id (example below), it does not see that it is a national
call and sets the wrong type of number.
From: Shawn Lawrence
sip:[EMAIL PROTECTED]
Is there anyone out there who has come across the
same problem? If so, how did you take care of it? Is there a way I
can make asterisk add
this file that would be willing to email it to me?
Any help will be very appreciated.
Best,
Shawn
[EMAIL PROTECTED]
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have
them. The sound quality is bad and the line fail over reverses the polarity
and makes the Nortel phones unhappy.
If someone has a suggestion for a place to buy such a device or wants to
build a prototype for me to test I would appreciate it.
Thanks,
Shawn
a feature where the card would tie the first trunk to the
first extension when the software was not controlling it or if the power
failed.
Thanks,
Shawn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Monday, January 10, 2005 2:58 PM
get a 404 error on the phone (
Call failed). Nothing shows in the Asterisk console.
Thanks in advance,
Shawn Dillon
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I cannot get the Do Not disturb function to work with my
Asterisk box. If I dial *78 , or *79 on my phone ( Sayson 480i SIP IP Phone)
and I get a call failed message.
Is the *78 , *79 function installed by default? Does it use
a certain module in Asterisk?
Any advice/comments?
Shawn
to
channel=1-8 I get errors that it cannot init channel #5.
I must be missing something simple.
TIA
Shawn
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We are in the final stage of a rollout of Asterisk in our
company. We had some Polycom IP 600 , a Snom 220 , a Grandstream 102 and
recently a Sayson 480i phone. I am interested in anyones opinions in the phone
they suggest to implement. I must admit I am a little partial to the Sayson 480i
,
= ,Shawn Wilson,[EMAIL PROTECTED]
and zapata.conf looks like this:
[channels]
context=incoming
signalling=fxs_ks
language=en
echocancel=yes
echocancelwhenbridge=yes
relaxdtmf=yes
rxgain=1.5
txgain=1.5
channel = 1
context=local-access
signalling=fxo_ks
language=en
echocancel=yes
echocancelwhenbridge
into the support queue manually?
As an aside , this community has
been very helpful in getting my Asterisk box up and running. Thanks to all.
Shawn Dillon
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?
TIA
Shawn
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receiving support calls.
Thanks
Shawn Dillon
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to standardize on IP phones
( Polycom 600) all around.
Also has anyone had experience with the Draytek VOIP
Wireless routers?
And finally, if we need to use a Sipura 3000 in the remote
offices is there any benefit with going with a Sayson analog phone versus any
other?
Thanks
Shawn Dillon
to have it
dial the exchange first . Something like 413,,5554441212.
Thanks for the help
Shawn Dillon
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