Did you look at logger.conf?
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of bilal ghayyad [EMAIL
PROTECTED]
Sent: Sunday, September 16, 2007 5:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] stop log/debug messages into
This is an interesting idea, did you come up with anything?
Are your users logging into an AD domain? A script to interact with the
Asterisk server could be run after login which adds an extension mapping the
user to the phone. One set of extensions for the users (which is published) and
I would be willing to mirror it also….
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, March 14, 2007 9:39 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] What happend to
If it's using RBS then 56k is the right number.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Saturday, January 27, 2007 12:55 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] max tnt pri voice
Why don't you just give the secretary the boss' REAL extension and give a
different extension to the world that just rings the secretary?
-jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho
Sent: Friday,
A demonstration:
exten = _X.,1,Set(GROUP()=${CALLERID(num))
exten = _X.,n,Set(CDR(AccountCode)=${CALLERID(num))
exten = _X.,n,GotoIf($[${GROUP_COUNT(${CALLERID(num))} 2]?103)
exten = _X.,n,Macro(trunk,${EXTEN},residential)
exten = _X.,n,Hangup
exten =
There is an index in the configuration file which I believe it will
obey. I'll try and find it later if you haven't found it by the time I
get to the office.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday,
I'm not sure if there is a more official method but Google has always
been my friend when searching the lists.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Greene
Sent: Wednesday, December 27, 2006 12:05 PM
To:
I don't use many of the features of astmanproxy but it does work. I use
it to capture events from several servers. Some of these are running the
1.4 beta releases.
-Jonahtan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tzafrir
${Number:-10:3} if I recall correctly would give you 3 characters
starting at the 10th from the end.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John French
Sent: Tuesday, December 19, 2006 10:35 AM
To:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bruce Ferrell
Sent: Tuesday, December 19, 2006 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Parsing Area Code from CallerID
I may be making this easier than it is but something like this should
work:
A:
DIAL(IAX2/${ASTERISKB}/[EMAIL PROTECTED])
B:
[context]
exten = EXTEN,1,DIAL(Zap/${EXTEN})
I have this scenario also except we have numerous A servers connecting
via the PRI lines on B servers.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Thursday, December 14, 2006 4:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hardware TDM Switching
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Thursday, December 14, 2006 4:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hardware TDM Switching
CLIPPED
I would have some kind of user 1010 (the actual extension and username
too)
Let's say that in manager.conf i would have again some user 1010 but i
would like that this user can only see the events associated to the
extension 1010 ...
CLIPPED
I am pretty sure that using the proxy,
? I am asking that because my system guys are not available
until
friday ...
Jonathan k. Creasy wrote:
CLIPPED
I would have some kind of user 1010 (the actual extension and
username
too)
Let's say that in manager.conf i would have again some user 1010
but i
would like that this user can
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Monday, December 11, 2006 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] CLI History
-Original
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Carla Schroder
Sent: Monday, December 11, 2006 2:17 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CLI History
On Monday 11 December 2006 9:31 am, Douglas
Title: RE: [asterisk-users] Polycom 501 config questions
Dumb question here: Why the
need to dial 9 for an outside line? If your extensions are less than 7 digits
long then you know anything "XXX." is an outside call
Maybe this isn't true everywhere, just
curious.
-Jonathan
Ive only used a Quintum a few
times,sorry.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abdul
Sent: Friday, August 25, 2006 6:49
AM
To:
Asterisk-Users@lists.digium.com
Subject: RE: [asterisk-users]
quintum Calling Card
Hello Jonathan,
I tried in
Abdul, it doesnt sound like you
need to do anything to the Quintum. I would recommend making your dial plan execute
the AGI script of your choice no matter what number is dialed from the context
where the quantum users land.
-Jonathan
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I am trying to track down a problem which is occurring on
about 1% of the phone calls through a customers system.
Layout looks like this:
PSTN PRI Asterisk A IAX Trunk over point to point T1
Asterisk B SIP over LAN Polycom
IP501
1) The user on
the Polycom IP501 phone dials
Title: Re: [asterisk-users] Quick One - PHP Script to restart Asterisk
?php
execute "asterisk -rx
'restart when convienent";
?
Not the exact syntax but should be enough
to get you going.
From: [EMAIL PROTECTED] on
behalf of Paul HalesSent: Fri 8/11/2006 3:51 AMTo:
Asterisk Users
2) Phone activation at check-in/phone de-activation and billing at
check-out. Are there GUI tools for this, or should I write my own
back/front end?
The integration with the hotel systems for the activation/deactivation
and billing can be tricky. Check the archives for some discussions
Very happy with the 501 and 601. So far, like the 430 as well.
The 301 is good for what it is but the display and lack of speakerphone
are annoying to me.
They are all very stable and compatible though. The provisioning on
these phones is excellent as well.
-Jonathan
-Original
Haven't read this whole thread (got way behind in this list :) )
Polycom has a softphone with video support also. Not sure if it is good
or not, just downloaded the trial version to test it out.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
I'll second that. I really like the provisioning features. My customers
prefer the 501 because they like the layout and speaker phone
functionality.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
West
Sent: Friday, June 23, 2006 10:20
I have a strange problem. I have a central server with my PRI on it.
There are three peripheral servers connected via IAX.
I have a 64bit system for my central server and the backup system is a
32bit system. If I have forwarding (sip redirect) turned on and
forwarding to an outside number (i.e.
I am not by any means recommending this to anyone but I wanted to
publish this for reference.
I have an Asterisk system connected to a provider via IAX trunks. There
are 32 phones on our network and we have about 400 calls per day to/from
our system. The hardware running this is a Pentium Pro
Just appending the area code variable is not always going to be correct.
You will need to lookup (google local calling guide) the proper NPA for
the NXX you are dialing. For example, in Louisville, Ky if you dial
948-1592 you will actually reach 812-948-1592 instead of 502-948-1592
even though
Below is a snipped debug on our PRI. We are getting number
only for the CallerID but the telco says they are sending us Name and Number.
We are getting the Name in a second frame but Asterisk is not passing it to the
device it rings. The message below says Presenation allowed of network
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PRI caller ID
Pleaase read the archives or the wiki - you will shortly find you
need a wait in your dialplan
On Apr 19, 2006, at 8:10 AM, Jonathan k. Creasy wrote:
Below is a snipped debug on our PRI. We are getting
I could be wrong but off the top of my head I think that it is in the
features section of the config file.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Tuesday, April 18, 2006 4:47 PM
To: 'Asterisk Users Mailing List -
I can dial other extensions internally, and can get to voicemail, but
when I try an outside number, I hear dial tone, the digits dialed, yet
nothing happens when I press Send.
Nothing appears on the Asterisk CLI screen.
Did the upgrade modify the dialplan setting on your phone? This
Does anyone know the format for the TOS element in the Polycom
config?
-Jonathan
Jonathan Creasy
Network Engineer
BluegrassNet Development
www.bgnd.com www.bluegrass.net
o. 502-589-4638
c. 502-889-5567
h. 502-541-0566
___
Anyone know what has happened to the local calling guide?
http://members.dandy.net/~czg/search.html
-Jonathan
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I apologize if this information is posted elsewhere. Unfortunately I
haven't found it yet if it is. I'm not familiar with the channel
counting features could you please explain? Also, how are you tagging
the phones to account codes?
You can limit calls using the set/check group commands.
Title: Re: [Asterisk-Users] Blocked channels,according to our telco... leading
to CONGESTION status
Id say try it out and see what the
CPU load is. Its not that hard to drop it in your dialplan and give it a
try. Its much easier than figuring out all the possible variables in
your setup
I have had this working but not reliably. It seemed to work like this:
Phone A watched B and C.
Phone B watched A and C
and Phone C watched A and B.
I could see on Phone A (601) when phone B (501) was on the phone. Phone
C never saw the status of either and Phone B would show the status of C.
You have to use H323 the last time I did
anything with their equipment. It has been almost a year but I think it went
fairly smoothly. Do you have a specific question?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen Arulraj
Sent: Friday, March 31, 2006
I agree we have this working also.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Boris Bakchiev
Sent: Friday, March 31, 2006 8:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
This is not a dumb question.
Most of the other replies I have read mentioned various ways to connect
to the pstn. I wanted to mention why it makes sense to do that. Many of
the companies I have installed asterisk for didn't even have their
system on a network with a gateway. They have dedicated
I think there is a bug related to this. I haven't been able to track it
down or really recreate it with any certainty yet. When I do I'll post
something to Mantis. If you have any info to share with me about your
situation when this occurs let me know.
I have noticed that I can get it to occur
I have found this to be true also.
[whatever] has to match username=
It appears that it ignores the username field for IAX users.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tomas Komarek
Sent: Monday, March 27,
I haven't read every message in this thread so I apologize if this is a
repeat. Have you considered using the cfg files and an ftp server to
configure the phones? I have found it to be very convenient as a way to
manage many phones spread out across several locations as well as
maintaining one or
Do you want to dial an outgoing line as well as the SIP line?
Dial(SIP/${OUTGOING}/${EXTEN}) ?
I can't say obviously without more info but it sounds to me like you are
looking for the wrong solution
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
It's a toll free number. You can call it from anywhere and the costs of the
call go on the callee not the caller.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Wednesday, March 22, 2006 7:50 AM
To: Asterisk Users Mailing
We are doing this with the latest spandsp, iaxmodem and hylafax.
Seems to work very well for us so far.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Gaudette
Sent: Tuesday, March 21, 2006 3:34 PM
To: 'Asterisk
I am having this problem also. I have 2 systems running 1.2.5. I had the
problem and one system was running 1.2.4 and the other was running a CVS
HEAD from October so I upgraded them both to 1.2.5 with no success.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
Well for one thing, on a PRI it is usually still transmitted with a bit
set that tells the system to hide it.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Thursday, March 16, 2006 9:49 PM
To: Asterisk Users Mailing
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Thursday, March 16, 2006 10:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT: Unblocking bloced CID
On 3/16/06, Jonathan k. Creasy [EMAIL PROTECTED] wrote:
Well for one
I have a problem where my Asterisk server stops answering new TCP
requests and begins to use 99.9% of the CPU on my box. The server is a
64bit Xeon with 2GB of ram.
I haven't been able to recreate the problem but it occurs sometimes when
there is a call coming from my provider (via IAX) to a
BOFH told me he uses it to listen to his co-workers
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Thursday, February 09, 2006 12:27 PM
To: asterisk-users@lists.digium.com
Subject: SOLVED: Re: [Asterisk-Users]
This feature also works on the IP301 phones. The obvious caveat is that
it is one-way only. Still nice for an all-page though.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Thursday, February 09,
It's something like exten = 15,1,Dial(Console/DSP)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Azzopardi
Sent: Saturday, February 04, 2006 2:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How
Do people not use the Grandstream ATA's because they are cheap or
because there is actually a problem with them?
They have a 2 line version for around $50 that I have used in various
locations. I have about 8 or so. They seem to do an excellent job.
-Jonathan
-Original Message-
From:
I have been planning to do the same thing but never got around to it, I
actually did write a nice class to wrap the interface to the manager but
it isn't complete.
Would you be willing to share your work?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
The Grandstream ATA (480 I think...) does this and usually costs less
than the Sipura. It has 1 FXS and 1 FXO.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Thursday, February 02, 2006 8:28 AM
To:
I am getting the following message when trying to lookup up a number via
Dundi:
Feb 1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key
'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for
'00:a0:c9:55:91:89'!
I have created keys on each box with astgenkey -n
Anyone in Winnipeg Canada?
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I am looking to be notified via email when a host fails it's qualify (is
unreachable). I found this patch
(http://bugs.digium.com/view.php?id=5372) but I wasn't sure if I could
get that from it.
Anyone else tried this?
-Jonathan
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I have 13 Polycom IP301's where the clock keeps resetting to a +5
offset. I can change the config file to show -5, change it to -5 on the
phone and after an hour or so the phone will update itself back to +5.
Anyone have any ideas? The other 70+ phones are not exhibiting this
behavior.
This is an authentication problem. Check the username, password, number
and context being sent across to see if they are correct.
Post your iax debug info for the call if you can.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
I am going to step out on a limb and guess that you need to hangup the call
when the digits are received which would move you on to the next priority where
you could then enter your loop. This is an autoattendant I made at some point
when I was playing around with how to do it.
Maybe it will
What causes this?
Dec 13 15:16:06 NOTICE[2660]:
chan_iax2.c:1561 iax2_destroy: Avoiding IAX destroy deadlock
Something occurs and I get a flood of
these then the box quits taking calls and asterisk wont die.
-Jonathan
The contents of this email message and any attachments are
I chose this method and have been happy with the results.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Anyone using any H.263+ video phones and want to relay their
experiences?
-Jonathan
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I don't think you even require SER in that case.
That will be $100.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac
Sent: Thursday, November 24, 2005 7:11 PM
To: users@openser.org;
I've thought about doing that as I have a few spare also. I would use
the raq4 I think.
Let me know if you have any trouble with it.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bram
kortleven
Sent: Monday, November 14, 2005 5:14 PM
To:
I'm just throwing this out here, not dissing anyone. Someone asking
these types of questions may want to seek some professional consultancy
with regards to the network before building a mission critical
deployment.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
For which equipment?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of xcel
Sent: Friday, November 18, 2005
11:53 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Provisioning server
Can any one help me in setting up Provisioning sever
??
What context are your phones in? (context= in sip or iax config)
If your phones are in the local-users context, they will be able to dial
numbers found in local-users, extensions and local.
If your phones are in the long-users context, they will be able to dial
numbers in long-users, local,
initially for Sipura SPA 1001.
Regards,
*** REPLY SEPARATOR ***
On 11/18/2005 at 12:44 PM Jonathan k. Creasy
wrote:
For which equipment?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of xcel
Sent: Friday, November
FXO ports are an interface between your system and a phone carrier. FXS
ports are an interface between your system and a phone station (or
handset).
You can send outbound calls on an FXO port as well as receive them.
To dial on the TDM card you would dial : DIAL(Zap/g1/${number}) or
I am trying to replace the overhead paging function of an old phone
system. There is a device with an RJ11 connection connected to two
screws on the phone system. The two screws are on a cartridge labeled as
the subject of this message.
I thought the other device was probably a station andthat I
I can't seem to compile IAXmodem.
sh build:
iaxmodem.c: In function `cleanup': iaxmodem.c iaxmodem-cfg.ttyIAX lib
README termpkg-ttydforfax.patch TODO
iaxmodem.c:90: error: too many arguments to function `iax_register'
iaxmodem.c: In function `main':
iaxmodem.c:705: error: `IAX_EVENT_CNG'
mailbox= in the sip.conf
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sixto Diaz
Sent: Tuesday, November 15, 2005
9:33 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Message
waiting notification
Hi i want to notify a user that he has an
Subject: Re: [Asterisk-Users]
Message waiting notification
i want to ring the phone user or change the
tone is this posible with mailbox= ?
- Original Message -
From: Jonathan k.
Creasy
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent
Use notepad if you must edit them on a windows box.
Nano/Pico/Joe are pretty user friendly editors for the *nix environment.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Tuesday, November 15, 2005
Anyone used a sip from from Planet Network?
VIP-153
http://www.planetnw.com/
http://www.planetstoresite.com/Merchant2/merchant.mvc?Screen=PRODStore_Code=PNIProduct_Code=VIP-152TCategory_Code=VOIP
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Is the ambient noise in the room high?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Tuesday, November 08, 2005 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Title: Extension Ring on Multiple Phones
EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE)
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow
Sent: Tuesday, November 08, 2005 1:51
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject:
I guess I should have read up further before I posted a response.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC
Sent: Tuesday, November 08, 2005 2:51 PM
To: Asterisk Users Mailing List -
I thought there was a sip image for that phone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Reynolds
Sent: Tuesday, November 08, 2005 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Cisco
Any IRQ or duplex problem with your NIC? Any collisions or errors?
I have had similar results to others here in that conferences with
50-100 users are just fine even on fairly outdated hardware.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Has anyone figured out how to make the shared line
appearance thing work with asterisk?
From wiki: http://www.voip-info.org/wiki/view/Polycom+Phones
Supports shared lines (but asterisk does
not) - Anyone having details on the specifications used for Shared Call /
Bridged Line
Can a Polycom IP601 with the addon modules be setup to work like an
attendant console showing the status of other lines?
How does that sort of thing work with Asterisk?
-Jonathan
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Anyone using the Genesys framework with an Asterisk PBX?
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Anyone know what's causing this:
-- SIP read from x.x.x.x:56800:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKba5c00f818387D67
From: Xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B
To: sip:[EMAIL PROTECTED];user=phone;tag=as57402fc2
CSeq: 1 ACK
I dont get it.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Richmond
Sent: Thursday, October 20, 2005
9:46 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] user
name
I am geting e-mail but asterisk doesn't know my user name or
I probably can't provide any better information for you, however, have
you looked through the Polycom configuration files. The button mappings
are there. I haven't spent much time with it so I can not attest to what
you can map them to do.
Hope this helps you a little.
-Jonathan
-Original
You can do it with a Polycom (and probably a Cisco) by setting an Alert
var and it will handle the call using a defined class.
Search for paging.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Pyeron
Sent: Tuesday, October 18, 2005
, 2005 9:00 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] No Audio from Console but
mpg123fromshellworksfine.
On 10/16/2005, Jonathan k. Creasy [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Sunday, October 16, 2005 2:59 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] No Audio from Console but mpg123 from
shellworksfine.
One possibility is that the
Anyone have anything on this? (I'm sure someone will complain about me
bringing it up again, chill out...)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, October 14, 2005 10:15 AM
To: Asterisk Users Mailing List - Non
I get audio from mpg123 at the command line but when I load up asterisk
and try to get audio from the console it looks like it's working, and
even pauses like it is playing the file but there is no audio coming
from the speakers.
I have searched and looked through the archives and tried to fix
I have been getting that message also. I have been using various
versions of CVS head since Feb. 2005.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
Gault
Sent: Friday, October 14, 2005 10:23 AM
To: Asterisk Users Mailing List -
I dont think the Quintum hardware
supports SIP devices (just SIP trunks).
-Jonathan
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Claudio Canseco
Sent: Friday, October 14, 2005
4:32 PM
To: Asterisk Users Mailing List -
Non-Commercial
I have changed the IP. It would only have an affect on your system if
you have a specific bind x.x.x.x in your config files. I use bind
0.0.0.0 to use all addresses on the machine so I had no problems.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
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