On Sun, 2006-01-22 at 15:23 +0800, Ronald Wiplinger wrote:
1. At the end of compiling asterisk I got a lot of warnings. How can I
solve that?
I used:
# svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
# svn checkout http://svn.digium.com/svn/libpri/trunk libpri
# svn checkout
Is there any way to have more specific disposition
codes in the CDR?
Currently there are only 3 values: ANSWER, NO ANSWER,
BUSY.
In this way, when i call a cell phone that is switched
off i get NO ANSWER, while i would like to be able to log that
the call is not answered because The customer you
Hi,
I'm using IAX2 but frequently get the jutterbuffer error message and it will pause my talk for about 3~5 seconds,
Jan 22 17:11:24 WARNING[19418]: chan_iax2.c:691 jb_warning_output: Resyncing the jb. last_delay 5, this delay 15876, threshold 1068, new offset -15876
Jan 22 17:11:24
I thing, that configuring nat device/firewall at consumer site isn't
always possible, thus simplest (but not optimal) way is to configure
phone in sip.conf as nat=yes canreinvite=no, this should work in most
cases even if multiple phones are behind same nat, like adsl router.
disadvatage is,
Hi all,
I'm experiencing weird cutoffs on TE110P. All cut offs are pre-seen with
an indication 5 coming from the PRI. I've talked to the telco, and they
indicated that they don't see any issues.
I've also modified the sync source to be the telco, and that didn't
solve the problem either.
Hi,
I am trying to install asterisk on Suse 9. I downloaded asterisk-1.2.1.tar and untar this package. I am following the README and the installation instruction to run make ans make install. But I can not find any make or make install in the directory
asterisk-1.2.1. Can any one please help me
On 15:30, Sun 22 Jan 06, Atif Nadeem wrote:
Hi,
I am trying to install asterisk on Suse 9. I downloaded
asterisk-1.2.1.tarand untar this package. I am following the README
and the installation
instruction to run make ans make install. But I can not find any make
or make install in the
HI All,
I have three a-to-z gateway from different
terminators, I want to add in extensions some timeout
condition.
for the example my timeout=2 seconds
if first gateway will not response in 2 second
automatically it should dial using second gateway,
respectively
I will be appreciate if any
This unit looks nice:
http://www.xorcom.com/ts-1/features.html
It's not very powerful but I'm sure it's more than enough as a clever
SOHO end point, say to manage SIP phones properly. You probably still
have to do voicemail IP-centrex style though.
Any ideas about pricing?
Cheers,
Why not get a proper GSM Gateway .
We have some for sell for £60 each ..
Contact me on sam AT cyber-telecom DOT net
Or visit cyber-telecom DOT net
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Rojas
Sent: Sunday, January 22, 2006 2:03
AM
To:
Leo Ann Boon wrote on Saturday, 21 January 2006 6:21 PM:
Trevor G. Hammonds wrote:
How about when you have four or five SIP devices at a single
location? Do you manually assign each phone a separate port and add
firewall/router rules? I am looking for an inexpensive device or
method that
I have the same problem too.
I install the G.729 (IPP) to asterisk 1.0.x, and it works well.
When I change asterisk from 1.0.x to 1.2.x, and G.729 seems work fine.
I can use show translation and find it too. But when I make a call
using G.729.
The asterisk (1.2.1) crashed. If i mark the line
On Sun, 2006-01-22 at 16:00 +0400, Jean-Michel Hiver wrote:
This unit looks nice:
http://www.xorcom.com/ts-1/features.html
It's not very powerful but I'm sure it's more than enough as a clever
SOHO end point, say to manage SIP phones properly. You probably still
have to do voicemail
I don't think there is any laws on GSM Gateway itself.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Augustyn
Sent: Friday, January 06, 2006 11:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Why not get a asterisk or Normal VoiP Gateway and then connect those
together .
Sure that will still cost less than 300 USD
and then you can run sip or iax or h323 on it.
Email me on sam AT cyber-telecom DOT net for more info
Sam
-Original Message-
From: [EMAIL PROTECTED]
[EMAIL PROTECTED] is believed to have said:
We are considering it yes, but i don't know how hard or easy it would be.
I guess we will first try to make the other versions like we want them
to be and then start looking at other os'es.
Zoa
I have no clue, at all, too
So your plan makes
Hi,
I tried a lot on these before to get in running on my [EMAIL PROTECTED] 2.2but without success.
Recently i bought 2 channels from diguim and installed it. It works well.
Just wanna know if i can install the non-commercial version to get more channels. Will it create problems for my
I downloaded and
installed the none commercial g729 codec very often now
I only disable HT on my
systems I think * doesnt like this
One of the guys @ digium advised
me to turn it of, since they havent written * to be multi treading any
way
The codec I download is
the
I have a T3 coming from my carrier. From there I want to use an Adtran
mx2800 T1 Mux to break the T3 into 28 T1/PRI which feed into seven quad
T1/PRI equipped servers.
Everything seems very straight forward with the exception of the D
channels for the T1/PRI.
I am not very familiar with large
No solution, but more data. We have an asterisk server at a branch office
(where it will reside at the end of the evaluation) and the NTP server at the
main office. The NTP server runs under freeBSD. DHCP info at the main office is
offered up by a Windows 2000 server and at the branch office by
Successfully got the adapters to allow the BT phones to ring
on lines coming out of a TDM.. but now my latest
problem is echo.
Suggestions / Experiences in UK appreciated
Most of our clients with BT lines tend to have ISDN BRIs, but we do have one
in Northampton running 3 analogue
Use Krone cable and a genuine Krone tool It isnt the
cheapest, but it is the best.
I concur with you on the genuine Krone tool, but I'm no fan of their patch
panels.
I find STP patch panels are much nicer to work with (even where STP isn't a
requirement).
On cable, we tend to use Belden or
Hi Everybody,
My MoH is distorted when a call comes in from PSTN line or Cell Phone. I
have SIP phone line and SIP to SIP is good. I don't have any digium
hardware installed, instead I am using ztdummy and apparently it is
working without any problem. What could be causing this distorted sound
On 1/22/06, Steve Totaro [EMAIL PROTECTED] wrote:
I have a T3 coming from my carrier. From there I want to use an Adtran
mx2800 T1 Mux to break the T3 into 28 T1/PRI which feed into seven quad
T1/PRI equipped servers.
Everything seems very straight forward with the exception of the D
Mark,
Thanks a lot for the feedback. It's reassuring to say the least
Mike
Message: 18
Date: Sat, 21 Jan 2006 15:36:18 -0500
From: Mark Phillips [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP and NAT - best practices?
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Sun, January 22, 2006 13:02, Charles Wang said:
I have the same problem too.
I install the G.729 (IPP) to asterisk 1.0.x, and it works well.
When I change asterisk from 1.0.x to 1.2.x, and G.729 seems work fine.
I can use show translation and find it too. But when I make a call
using
Your mux will split the DS3 into 28 DS1 (T1) circuits. They are
numbered 1 - 28. You tell your carrier how and where to assign the D
channels. The mux does not have anything to do with D channels or
signaling; the telco's ISDN switch does.
A simple setup would have a D channel on every DS1
I want to take this opportunity to thank you all for your emails of concern.
Reports of my death have been greatly exagerated.
This said, what a great singer, how many times have I been typing away
at the Midnight Hour, listening tot hat song in my head.
wp
Hi,
Has anyone a working /etc/asterisk/capi.conf example for Germany or
Switzerland using the AVM C4 - ISDN Card.
I try to connect asterisk to 3 wires BRI-ISDN (Swisscom).
I appreciate your help and it would save me a lot of time, figuring
it out by myself.
We are using 4 BRI
[EMAIL PROTECTED] wrote:
Is there any way to have more specific disposition codes in the CDR?
Currently there are only 3 values: ANSWER, NO ANSWER, BUSY.
In this way, when i call a cell phone that is switched off i get NO
ANSWER, while i would like to be able to log that the call is not
I'm trying to write some dialplan patterns to allow my users to control
call forwarding from their handsets. Right now, I have this in
extensions.conf:
[forwarding]
exten = _*21*X.X*,1,Macro(set-cfim,${CALLERIDNUM},${EXTEN:4})
I was hoping that this would match any string of the form *21*nnn*,
I've got the same issue than you. Have you solved your problem ?
I enabled Enhanced Real Time Clock Support in the kernel config,
recompiled the kernel, then recompiled Zaptel.
I found the rtc: lost some interrupts at 1024Hz messages seem to be
related to rebuilding arrays on my RAID5
Thanks for some answers, that is what I thought.
Asterisk is NFAS capable so I am looking at seven D channels on the T3 I
guess. I don't want to put a D channel on each T1 or I will lose
several channels that could be used for calls.
I wonder if there is any way that Asterisk can do NFAS
I am not sure that it is a problem more so an annoyance. If
someone dials my extension number or external DDI while I am
already in a call rather than skipping to the next priority
in the dial plan for example voicemail the line continues to
ring and while in a call I can hear the phone
This is a problem because many of my users are using GXP-2000s with
Early Dial enabled: I need Asterisk to go on rejecting the number with
484 address incomplete until it sees the final * digit.
Can anybody give me a clue how to accomplish this?
If the phone is even entry quality, it should
pain to configure) have 4 ring types. I am guessing that I would
need to figure out how to tell this particular phone to use a
different ring tone unless there is a way to send a
stutter type ring to the phones.
Has anyone found a solution to this?
I did a similar thing for a
Can * detect distinctive ringing on a SIP line? The reason I ask is I
have broadvoice with an add on line. It does not send any type of info that I
know of for the two separate lines so I can not determine which number is
ringing. Broadvoice can however send distinctive ring tones so if I
I second your opinion that Polycom needs to change their policy on this.
Strengthening the Reseller Channels is one of the more nonsensical
justifications for not publicly providing updates for their own product.
Especially in light of the fact that you can easily and legally get
the source
On 1/20/06, Steven [EMAIL PROTECTED] wrote:
I have seen a lot of references to SER.
Where might SER help?
Why are people using it with Asterisk?
I simply bring it down to this:
Am I service provider? If yes, then I probably want to use SER for my
registrations. If not (standalone PBX --
I'd love to see this as well.
TIA,
Max
On 1/21/06, Thomas Johnson [EMAIL PROTECTED] wrote:
Thanks!I'd love to see your extensions.conf file.I appreciate it.TomOn Jan 20, 2006, at 8:31 PM, Alexander Lopez wrote:
Contact me off list, I have a sample extensions.conf file that I can share. It has
Everytime you come home, removing SIM out of cell phone and into GSM
Gateway can be a hassle. (the reverse when you go out of home.)
On 1/22/06, Sam Tam [EMAIL PROTECTED] wrote:
Why not get a proper GSM Gateway .
We have some for sell for £60 each ..
Contact me on sam AT cyber-telecom
I am sure that are many more that would be interested (including me).
Why not just post it on the list after sanitizing private information?
Thanks.
On 1/22/06, Max Clark [EMAIL PROTECTED] wrote:
I'd love to see this as well.
TIA,
Max
On 1/21/06, Thomas Johnson [EMAIL PROTECTED] wrote:
Please stop plugging the book. Its annoying. We know
its out there.
Dovid
--- Leif Madsen [EMAIL PROTECTED]
wrote:
On 1/16/06, John Falk [EMAIL PROTECTED] wrote:
Can someone show me how to set up DUNDi, I will be
using it to connect
14 asterisk servers internally. I don't want to
use it
Hello List,
I have more of a generic question. A lot of times when
links to books, little bits of codes, diffrent
programs etc. are posted I do a wget to my server so I
can have it for future yes. Every now and then I reply
to questions with links to these kinds of things. I
have never posted the
Dont think one exists. You may want to get an ATA that
has 24 FXS ports on it.
Regards,
Dovid
--- Giordano Grandis [EMAIL PROTECTED] wrote:
Hi all,
I'm looking for a PCI card which i could install on
asterisk box, with
purpose to use 15-20 cordless dect phone in a very
dect cell.
Is
RANT
Funny your concerned about copyrights and moral issues regarding the
work of others.
One question you may want to ask YOURSELF is:
Why would I use as my email a copyrighted work followed by the
name of the Company that owns the copyright???
[EMAIL PROTECTED], Come on!! Who are
Matt,
Wouldn't they have to actually enter a forwarded number for the forward to
activate? I've hit the forward button myself many times after a call ends, and
the phone asks you for a new number to forward to.
Douglas.
-Original Message-
From: Matt Darnell
I don't think you can beat the Polycom's for design, features, configuration
options and functionality tho. :)
-Original Message-
From: Wilson Pickett [mailto:[EMAIL PROTECTED]
Sent: Sun 1/22/2006 10:32 AM
To: Asterisk Users Mailing List -
Or on the wiki
http://www.voip-info.org/wiki-Polycom+Phones
-Original Message-
From: Nilesh Londhe [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 22, 2006 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Need a good
Hi,
Douglas Garstang wrote:
Hang on there's a non commercial G729 codec that will work with Asterisk?
Can someone point me to where I can find it?
Check out
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing
The binaries from
http://kvin.lv/pub/Linux/Asterisk/
work for me (* 1.2.2
Wilson Pickett wrote:
Further, Polycom SIP phones have the longest boot time of any phone
I've ever seen (something like 5 min, compared to a Sipure, less than
Give a SIP based Cisco 79XX phone a try, just about as long in boot time.
Doug
--
Ben Franklin quote:
Those who would give up
On Sun, January 22, 2006 19:40, Douglas Garstang said:
Hang on there's a non commercial G729 codec that will work with
Asterisk? Can someone point me to where I can find it?
Thanks,
Doug.
Intel provides a sample for non-commercial/testing.
http://www.voip-info.org/wiki-ITU+G.729
and
On Sun, 2006-01-22 at 11:40 -0700, Douglas Garstang wrote:
Hang on there's a non commercial G729 codec that will work with Asterisk?
Can someone point me to where I can find it?
non-commercial is a misnomer, the patent may still apply for your usage,
then again it may not. The
I am trying to construct a macro for long distance dialling. I have two
internet feeds, I have all routes including Teliax on Internet A and a
static route to Voxee on Internet B. I thought I could use the dialplan
entry below which uses the ChanIsAvail() command to check the
connection, but
If it worked with 1.0.7 then it was a bug, it should *not* work. The g
will only work if one statys on the phone, from show application dial:
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
If the origianl channel hangsup as well,
Anyone know if it is possible to control how aggressively the
Aggressive mode behaves.
Meaning, is it possible to dial back the aggressive mode to have a happy
medium between
Regular and the Aggressive defaults.
I have a situation where Normal echo cancellation is not quite enough,
however when
non-commercial is a misnomer, the patent may still apply for
your usage, then again it may not. The libraries that are
used are intels IPP which are free for non-commercial
non-distribution purposes, if you want to distribute you have
to pay intel money, but that gives you the core from
Hi,
I've successfully used the 'd' flag in Dial() so that when
I dial into my phone system from out there in the PSTN
network I can press the 2 key while the phone is ringing
to listen to my voicemail.
It seems that one issue is that the public providers
do not deliver DTMF, or anything, until
Title: Snom 320 and message retrieve key
Hi,
I found some issues with Snom 320 message retrieve key. This button works only when the MWI does not blink! If MWI
blinks and I do press retrieve button I get Unknown on display and busy tone. From the sip debug it looks like that Snom
does not
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Guillermo Salas M wrote:
I've the same problem with sip1.sipdiscount.com. The calls are not
connecting but are billed.
SIPDiscount seem to have been having intermittent problems since Friday
morning. It seems to be working now however.
- --
On Sun, 2006-01-22 at 19:43 +, Chris Bagnall wrote:
non-commercial is a misnomer, the patent may still apply for
your usage, then again it may not. The libraries that are
used are intels IPP which are free for non-commercial
non-distribution purposes, if you want to distribute you
Has anybody managed to get asterisk working reliably with the
SIPDiscount inbound number?
I have got as far as having an extension = myusername in the inbound
context. As a test I have configured this to answer, play the
tt-weasels message than hangup. Watching the asterisk console, when I
On 16:19, Sun 22 Jan 06, David Hajek wrote:
Hi,
I found some issues with Snom 320 message retrieve key. This button
works only when the MWI does not blink! If MWI
blinks and I do press retrieve button I get Unknown on display and
busy tone. From the sip debug it looks like that Snom
does
Hello,
To detect an answering machine I have found following two commands,
BackgroundDetect (comes with asterisk)
MachineDetect (asterisk add-ons)
First question, does BackgroundDetect works well with g729?
I havn't try MachineDetect yet, what is the benefit of MachineDetect over
On Sun, January 22, 2006 22:32, Ron Wellsted said:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Guillermo Salas M wrote:
I've the same problem with sip1.sipdiscount.com. The calls are not
connecting but are billed.
SIPDiscount seem to have been having intermittent problems since Friday
I also didnt comment on whether or not anyone can prove that
you do have licenses, even if they know you use the codecs.
Because to rely on that would be dubious at best, shut you
down at worst.
Out of curiosity, I wonder what one's legal position would be if one bought
the appropriate
Hello Giovanni and everybody,
Thanks a lot for your
suggestion.
Unfortunately, that does not help. With
READ_SIZE=16, I got vibrating voice on the speaker phone. With READ_SIZE=80, the
voice came back to normal and the echo is more reduced but still noticeable. I
finally changed it back
High availability. If you have multiple Asterisk systems, SER can really make failover a lot less painful.
On 1/22/06, Leif Madsen [EMAIL PROTECTED] wrote:
I simply bring it down to this:Am I service provider? If yes, then I probably want to use SER for my
registrations. If not (standalone PBX --
In article [EMAIL PROTECTED],
Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Dan Austin [EMAIL PROTECTED] wrote:
Tony wrote:
I should tidy it up and submit it, but haven't got round to it :-(
Let us know if you can. I'm already maintaining a grocery list
of
Dear all,
I know, you get what you pay for. I bought an IP
SIP Phone/2.0.6 from safe.com (£55) and the basic functionality is
fine.
The problem is when it tries to re-register it
hangs for a minute or so and you can not dial nor receive any calls. It
also
has a registration button which
Title: Snom 320 and message retrieve key
use
exten =
asterisk,1,VoicemailMain(${CALLERIDNUM})
or universal
exten =
default,1,VoicemailMain(${CALLERIDNUM})exten =
asterisk,1,VoicemailMain(${CALLERIDNUM})exten =
unknown,1,VoicemailMain(${CALLERIDNUM})exten =
Innocent Evil wrote:
To detect an answering machine I have found following two commands,
BackgroundDetect (comes with asterisk)
MachineDetect (asterisk add-ons)
Check out http://bugs.digium.com/view.php?id=5959
app_AMD blows everything else out of the water. I haven't run it in
production
I had:
exten = 695,2,SayDigits(${CALLERIDNUM}) ; Says your phone number
but it does not work anymore after upgrade. How should it be now?
___
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Trevor G. Hammonds wrote:
While I have not used siproxd, I have read a bit about it. From my
understanding of the docs, the local SIP agents register to siproxd, but
siproxd does not register to Asterisk. So the calls will traverse the NAT
properly, but features like MWI will not work in this
Last time I checked, Broadvoice sent the Alert-Info header in the
INVITE message. The main line does not have this header, an add-on
line does.
On 1/22/06, Robert Mann [EMAIL PROTECTED] wrote:
Can * detect distinctive ringing on a SIP line? The reason I ask is I have
broadvoice with an add on
Leo Ann Boon wrote on Sunday, 22 January 2006 4:32 PM:
Trevor G. Hammonds wrote:
While I have not used siproxd, I have read a bit about it. From my
understanding of the docs, the local SIP agents register to siproxd,
but siproxd does not register to Asterisk. So the calls will
traverse
check /etc/asterisk/logger.conf
regards,
cameron
-- Forwarded message --
From: Angelito Manansala [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sun, 22 Jan 2006 06:57:05 +0800
Subject: [Asterisk-Users] How
Hi,
Where can I find objective reviews of VoIP phones? Somebody out there must
have done a comparaison of those phones, unfortunately all I can find at
reviews of one phone (without comparing them to others) or obviously biased
ones.
Also, I'm looking for a good value business phone (for me,
I cannot see it
make[1]: Leaving directory `/usr/local/src/svn-versions/asterisk/pbx'
/bin/sh: curl-config: command not found
make[1]: Entering directory `/usr/local/src/svn-versions/asterisk/apps'
Makefile:103: *** missing separator. Stop.
make[1]: Leaving directory
Ronald Wiplinger wrote:
I cannot see it
Found it!!! Tab and spaces are hard to see,
make[1]: Leaving directory `/usr/local/src/svn-versions/asterisk/pbx'
/bin/sh: curl-config: command not found
make[1]: Entering directory `/usr/local/src/svn-versions/asterisk/apps'
Makefile:103:
How should be the macro rewritten?
[macro-faxreceive]
exten = s,1,Set(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,Set([EMAIL PROTECTED])
exten = s,104,Goto(3)
...
[Jan 23 10:43:38]
This is a UK site, but not sure if it's available elsewhere, or even if
it's what you want!
http://www.voiptalk.org/products/COM-ON-AIR+Desktop+Set+Clearance
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid
Bender
Sent: 22 January 2006 18:25
To:
You could also achieve the same result with phones that support some type of
failover, such as a simple list of systems to try in order, or DNS SRV lookups.
-Original Message-
From: Jon Radon [mailto:[EMAIL PROTECTED]
Sent: Sun 1/22/2006 4:27 PM
Check out www.generationd.com for their fax2mail and mail2fax scripts. It
might make life simpler
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Sunday, January 22, 2006 9:56 PM
To: Asterisk Users Mailing List - Non-Commercial
Line 103 in Makefile has multiple spaces at the beginning instead of TAB character.
On Mon, 2006-01-23 at 10:19 +0800, Ronald Wiplinger wrote:
I cannot see it
make[1]: Leaving directory `/usr/local/src/svn-versions/asterisk/pbx'
/bin/sh: curl-config: command not found
make[1]:
How to set the callerid? I had prior 1.2:
exten = _91NXXNXX,3,NoOp(SetCallerID(${username}))
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I've posted this to SNOM, but was wondering wheter anyone here has issues with
SNOM 190 phones not showing the correct DST adjusted time (using the latest
firmware).
--
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
The one you demonstrate should have *never* worked.
Pre 1.2 you do:
exten = s,1,SetCIDNUM(12345789)
Post 1.2 you do:
exten = s,1,Set(CALLERID(num)=123456789)
On 1/22/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
How to set the callerid? I had prior 1.2:
exten =
Sadly, most of the phone manufacturers do not understand Southern Hemisphere
daylight savings.
Don't know why, but they just don't.
PaulH
Rod Bacon [EMAIL PROTECTED] wrote:
I've posted this to SNOM, but was wondering wheter anyone here has
issues with
SNOM 190 phones not showing the
C F wrote:
The one you demonstrate should have *never* worked.
well, it did,
Pre 1.2 you do:
exten = s,1,SetCIDNUM(12345789)
Post 1.2 you do:
exten = s,1,Set(CALLERID(num)=123456789)
I need to get the callers phone number there!
How can I do it now?
exten =
On Tue, 15 Nov 2005 11:51:33 -0500, you wrote:
On 11/15/05, Brian Roy [EMAIL PROTECTED] wrote:
On 11/14/05, BJ Weschke [EMAIL PROTECTED] wrote:
There is a known issue right now where using mixmonitor with
chan_local is going to cause an unintentional disconnect. Are you
using Local/ with
On Sun, 22 Jan 2006, Steve Totaro wrote:
I have a T3 coming from my carrier. From there I want to use an Adtran
mx2800 T1 Mux to break the T3 into 28 T1/PRI which feed into seven quad
T1/PRI equipped servers.
Everything seems very straight forward with the exception of the D
channels for
On Sun, 22 Jan 2006, Steve Totaro wrote:
Thanks for some answers, that is what I thought.
Asterisk is NFAS capable so I am looking at seven D channels on the T3 I
guess. I don't want to put a D channel on each T1 or I will lose
several channels that could be used for calls.
I wonder
I have some users who like to forward their extensions out to cellular phones on weekends. They can currently do this using *72cell # which AMP provides. However, in the event that this forward is enabled and a call is forwared to their cell phone but they do not answer it, it will be passed back
At 12:36 PM 01/20/2006, you wrote:
So please respond if you have used either of these two phones. Give
me any information you may have on them.
I've had the Aastra 480i CT for a week or 2 and so far I've been very
happy, but I've not used it much. Aastra claims that while it will
support 4
I am unsure of * capabilities on NFAS (we do not use PCs to terminate
any PRIs), but it allows bonding of desparate PRIs to use a single
d-channel. ie, you can have 1 d-channel (optional backups) for the
entire DS3. Not sure if * can communicate across cards like that in the
same bus though.
On
Hi all,
Just wanna know if what i understand is true.
Does asterisk support G723. Well, i im given to understand that it just allows passthru. Is it true can anyone clarify.
Thanks
Dan
On 23/01/06, Chris Bagnall [EMAIL PROTECTED] wrote:
I also didnt comment on whether or not anyone can prove
Is there a way to change an agent's password without editing the password field in agents.conf and reloading asterisk?
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On Monday 23 January 2006 06:03, Ronald Wiplinger wrote:
C F wrote:
The one you demonstrate should have *never* worked.
well, it did,
Pre 1.2 you do:
exten = s,1,SetCIDNUM(12345789)
Post 1.2 you do:
exten = s,1,Set(CALLERID(num)=123456789)
I need to get the callers phone
Well, show codecs indicates 723.1 is codec 1, but show translation
indicates 723 can't be translated to anything. (Same is true with g729
unless you installed the codec.)
That would suggest its not supported unless you buy and install the codec.
Pure passthru should work.
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