Edwin Lam wrote:
Carla Schroder wrote:
On Monday 23 October 2006 17:38, Edwin Lam wrote:
Re: [asterisk-users] Polycom SP4000 ftp problem
From: Edwin Lam [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Carla Schroder wrote:
Sooo...stick with tftp? :) Seriously, that's what it's for.
Joseph wrote:
Though what option am I suppose to pass it.
The process seems to me correct, when I get-in to disa-access I have
access to voicemail extension 1000 (otherwise it wouldn't let me dial
ext. 1000; when I dial it it asking me for mailbox number and password,
except that password is not
Thomas Kenyon wrote:
Remco Barendse wrote:
It is not, asterisk is correctly started after networking services,
however it seems that when the box is booting the dns is replying just
a split second too late for the taste of asterisk and it seems that
asterisk then marks the provider as
mail-lists wrote:
Hello,
I've posted this at the trixbox and freepbx forums and haven't been able
to get an answer. I thought perhaps the guru's here might be able to
help me out :)
I'm having some issues with setting caller IDs. There are 2 problems
that I would like to solve.
1. I have
David Edwards wrote:
Thanks..
Did I misread the posts? These look like the VWIC-1MFT-T1 is connecting to
the PSTN and not connecting to a PRI card on an Asterisk box..
We are looking to do the following..
Asterisk PRI card - VWIC-1MFT-T1 - SIP -
Why not have Asterisk connect directly to
Joseph wrote:
On Mon, 2006-10-23 at 07:46 -0500, Eric ManxPower Wieling wrote:
Joseph wrote:
Though what option am I suppose to pass it.
The process seems to me correct, when I get-in to disa-access I have
access to voicemail extension 1000 (otherwise it wouldn't let me dial
ext. 1000; when I
Martin Joseph wrote:
On 2006-10-22 20:58:46 -0700, Avi Miller [EMAIL PROTECTED] said:
On 23/10/2006, at 10:13 AM, Joseph wrote:
I'm trying to log-in externally (from PSTN line) to check my
voice-mail so I created context to authenticate log-in
Just create an inbound route to
Avi Miller wrote:
On 23/10/2006, at 2:24 PM, Martin Joseph wrote:
It doesn't work. pressing * during my outgoing message does nothing.
Works for me. 1.2.12.1 with FreePBX. When I press *, I get a password
prompt. Entering my password gets me into the main voicemail menu.
FreePBX is NOT
Joseph wrote:
What does the | do
like this:
exten = s,8,Voicemail(11|)
From the CLI show application voicemailmain
Description]
VoiceMailMain([EMAIL PROTECTED]|options]): This application allows
the
calling party to check voicemail messages. A specific mailbox, and
optional
corresponding
Lacy Moore - Aspendora wrote:
Have you looked at
http://www.didww.com/support/index.php?_m=knowledgebase_a=viewarticlekbarticleid=3nav=0,1
yet?
I can't really trust a company that does an allow=all and a
dtmfmode=inband in their recommended setup.
This combo will FAIL to pass DTMF if you
[EMAIL PROTECTED] wrote:
Thanks to all that replayed, I made like Mr Watkins told me, and my problem is
apparently solved, although, because of the usage of the syntax
VoiceMail(${EXTEN}|u), now, two more sound files are played: vm-theperson and
vm-isunavail, while before were only played
Ricardo Carvalho wrote:
I'm running Asterisk version 1.2.10. I also tried with version 1.2.4 and
got same problem.
I use SIP and in my extensions.conf I have the following code:
exten = _[a-z].,1,Answer
exten = _[a-z].,2,Wait(1)
exten = _[a-z].,3,VoiceMail(${EXTEN})
exten = _[a-z].,4,Hangup
Todd- Asterisk wrote:
Thanks for the help Jerry - I'm getting closer, but still no luck...
Now, I hear the lady say S. I think what is happening is that the
GoTo command is setting the extension to 's' when it transfers control
to the context defined in the IAX.conf -where I have the trunk
Todd- Asterisk wrote:
Hi Eric- It wasn't typo, it was truncated for posting :) Below are
the complete relevant files. I'm getting 'S' when I want to hear the
DID number.. This machine was a trixbox about a two weeks ago, but
I've since tossed away the GUI and do everything by hand now.
Todd- Asterisk wrote:
When dialing from internal extension, it gives me the number I dialed
( in my case..). When dialing from AIX or SIP from didww.com
nothing comes through on the SayAlpha thanks for the thought
though... What I want is the number that the user dials to get my
Robert La Ferla wrote:
I have been experiencing a problem where after someone calls me from an
analog line, the phone call is terminated after a period of time
(anywhere from 15 seconds to 15 minutes) The phone that I use to answer
the call is an Aastra 9133i SIP phone. There are several
Matt wrote:
In the case of you example the IAX2 registration came in from the source
port on the far device of 1207.
Connections don't just move between ports.
I understand all this. However, here is my question.
MY on 4569 OTHER SIDE 1027.
Is both the incoming and outgoing traffic on
Matt wrote:
On 10/16/06, Time Bandit [EMAIL PROTECTED] wrote:
Why is it running on port 1207?
because Asterisk is listening on port 4569 and when a connection comes
in, it as handed to another port so it can continue listening on port
4569. Otherwise you would only be handling 1 connection at
Time Bandit wrote:
Thanks for the answer, but I don't buy it. There are currently 0
calls up on that bridge, while another connection which has calls up
on it is on Port 4569.. please try again. IAX2 is suppose to run on
ONLY one port.. this is why it is so nice for use in firewall
situations.
Benjamin Jacob wrote:
Hello ppl,
This post is to do with the variables 'nat' or 'canreinvite' for sip
entities.
Idealy users, wont be static, they could be roaming all over the globe.
So, setting someone as behind NAT, and disabling canreinvite, etc.,
restricts the roaming capabilities of a
Richard wrote:
I would have to second the Sangoma buy. Their tech support is second to
none and more then helpful.
I've never had any problems with their products that wasn't my own fault.
You do have to be careful with Sangoma. We recently had a Digium 4-Port
T-1 card blow up.
Alex Robar wrote:
If the phone is not registered, how will you make outgoing emergency calls?
The ONLY thing registration does is notify the server what this device's
current IP address is. This permits the server to send calls to the
device. It has NOTHING to do with calls from the device
Brian Candler wrote:
On Fri, Oct 13, 2006 at 07:00:54PM -0500, Eric ManxPower Wieling wrote:
* Phones = stations, regardless of where they are
Asterisk = SIP Server, Phone = SIP Client
* Trunks = trunks to other SIP servers, bilateral
Asterisk and the other server is peer to peer
Stephen Bosch wrote:
What's the trick for getting the Polycom IP 501 message light to go on
when there is voicemail waiting?
There is no trick that is specific to Polycom.
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Remi Quezada wrote:
When I reload the asterisk I get the following warnings:
Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring
switchtype
Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring
signalling
Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874
Matt wrote:
On the asterisk console I see:
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
--
* Phones = stations, regardless of where they are
Asterisk = SIP Server, Phone = SIP Client
* Trunks = trunks to other SIP servers, bilateral
Asterisk and the other server is peer to peer
* Services = services you register for, like BroadVoice, Voop or FWD.
(where asterisk acts as a
[EMAIL PROTECTED] wrote:
I am an asterisk newbie. I have successfully installed asterisk on Freebsd.
The problem I am having is when I try to route based upon incoming DID.
CALLERID(dnid) nor CDR(dst) have a number in them. Please help.
Digium analog cards do not support DID service.
Alex Robar wrote:
Analog routes (ie. copper telco lines) do not have DID information on them.
Only digital lines (PRI, often VoIP DID) have this information sent
alongside the call.
Analog lines in the USA can support DID, but only using things like EM
Wink which the Digium cards do not
As far as I know any signaling protocol you can get on a Channelized
Voice T-1 you can also get on analog lines. In fact these signaling
protocols originated on analog lines and are simply emulated on voice
T-1s. Notice I said Channelized Voice T-1, not ISDN PRI.
Granted, analog lines with
Matt wrote:
Avi Miller wrote:
On 04/10/2006, at 1:55 AM, Matt wrote:
How can I make * aware of the other ext on the remote box so the DID
caller can access them like he can with the local box?
On each box, define the other range:
Box A:
exten = _9XX,1,Dial(IAX2/BoxB/${EXTEN})
Box B:
Dave Cotton wrote:
On Thu, 2006-10-12 at 12:00 -0400, Jay R. Ashworth wrote:
On Thu, Oct 12, 2006 at 04:53:51PM +0200, Dave Cotton wrote:
It's more likely directly linked with how asterisk deals with
registrations to external SIP/IAX servers it appears to sit there for
ever trying to do the
Dave Cotton wrote:
On Thu, 2006-10-12 at 11:56 -0500, Eric ManxPower Wieling wrote:
This has been talked about quite a bit on this mailing list. Search the
archives.
Why? I don't have a problem I've solved it in my case. But my solution
will be of no use whatsoever for most others. I don't
Matt Florell wrote:
If you downgrade, let us know if it fixes things for you.
It's strange that there were so many changes in the 1.2 SVN branch
after 1.2.7.1 that seem to be complete changes in how some things
operate(like the transcoding optimization mess for Asterisk 1.2.11 and
1.2.12 that
Zeeshan Zakaria wrote:
I set up facilityenable=yes in zapata.conf, but it still didn't work for
caller name.
Searched google and found out that when Asterisk is configured as
switchtype
National with signalling pri_net, it does not send the Display information
in the Facility message.
Mark Quitoriano wrote:
Hi list,
i noticed from the cli my asterisk box is accepting unauthenticated calls
how can i prevent this?
CLI:
-- Accepting UNAUTHENTICATED call from 192.168.0.2:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host
Douglas Garstang wrote:
Crikey. I can't get this to work!
Allegedly, you can press 0 while in the voicemail greeting and be dropped to
the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't
clear about what context the o extension should be in. The voip wiki says
the
Douglas Garstang wrote:
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 10, 2006 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Press '0'
Douglas Garstang wrote:
Crikey. I
What I don't understand is why people MUST use the 2.0.x firmware.
Jessee J Holmes wrote:
A few of our technical support staff here at Atacomm are currently
working on this issue with Polycom, Digium and one of our customer's
(who posted in here earlier this week).
There are some major
Douglas Garstang wrote:
How can I check a number is within a specified range in the dialplan?
What's the greater than operator? How would I use a combination of
greater than and less than in conjection with GotoIf()?
README.variables should give you what you need. It's in
Sean Kennedy wrote:
Hey list,
Short version:
I have a need to disable the hardware can on the tdm24xxp I have. I
figure it's something in zconfig.h in the zaptel directory, but I'll be
damned if I can figure it out.
Long version:
I have a tdm2403e card which is experiencing an odd problem;
I had a similar problem. Turned out I wired it backwards. The Tellabs
only does EC in one direction.
Doug Lytle wrote:
Another question,
Is anybody using the Tellabs 2572 EC with a PRI. When we moved from our
analog lines to a PRI, I thought it would be simple stuff moving the EC
to the
Technically DTMF should be a signaling thing, but I believe Asterisk
must stay in the media stream if you want to use t/T/w/W. This may have
changed in 1.4. canreinvite=no in sip.conf would keep Asterisk in the
media stream.
Steve Glaus wrote:
Eric ManxPower Wieling wrote:
I don't know
Naija Man wrote:
Hello all,
Asterisk 1.2.8
zaptel 1.2.6
Hardware: digium TDM2422P
I have a fully configured asterisk system with POTS line for PSTN access. I
am not receiving the callerid for incoming calls from the PSTN. I get the
following error message.
-- Starting simple switch on
I don't know if this is even possible. I might be totally wrong but once
this call is on the cell network, how are you gonna communicate with
asterisk?? From what I understand, while the voice (RTP) traffic still
travels through asterisk, You have no access to any kind of signalling.
Please
yusuf wrote:
Eugeniy Khvastunov wrote:
Hello!
Why Asterisk tell: Unknown signalling method 'pri_cpe'
Why the asterisk does not know such signaling method?
[chan_zap.so] = (Zapata Telephony)
Oct 3 13:04:02 ERROR[5823]: chan_zap.c:10601 setup_zap: Unknown
signalling method 'pri_cpe'
Oct
Naija Man wrote:
As a habit, I do not force users to dial 9 or any other prefix of any
kind
to access external lines. You can just check the dialled number and prefix
with appropriate digits appropriately. See below. NOTE: THIS IS
US-CENTRIC!!
but can be easily made to work for any country.
Jay R. Ashworth wrote:
On Sun, Oct 01, 2006 at 09:42:32AM -0500, Eric ManxPower Wieling wrote:
My problem with this type of dialplan is that users must wait for
DigitTimeout before the call is processed.
I was going to ask about that.
What's the common value for that number, and secondarily
Not having a [globals] section (even if it is empty) has caused Asterisk
to screw things up in the past. I think it causes contexts to not be found.
Brian Candler wrote:
On Thu, Sep 28, 2006 at 09:44:07AM -0500, Eric ManxPower Wieling wrote:
You need the [general] and [global] sections
stan ford wrote:
if you have to setup an office of 100 users now. would you rather setup a sip
trunk,a t1-pri, or even a t1? and why?
Always a PRI. PRIs have fast call setup, are reliable and work well.
___
--Bandwidth and Colocation provided by
Make sure you have a /etc/asterisk/indications.conf
Shidan wrote:
For some reason I'm having problems with DISA. This is what I have:
exten = s,1,Answer()
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(10)
exten = s,4,DISA(no-password|from-internal)
I can generate tones with no
Try eBay
Forrest Beck wrote:
To rich for my blood. Googled it. Looks like it is about $12000, I
hope to stay in the $1500 range. We are but mearly a private school.
On 9/29/06, James [EMAIL PROTECTED] wrote:
I use the Lucent MAX TNT.
They are cheap, will do up to 24 T1's, have 12 fans and
Brian Candler wrote:
Hello,
I have an anomoly that I am unable to explain.
My entire extensions.conf is attached. You can see that the [from-sip] and
[internal] dial plans are identical, each including 4 other contexts in the
same order:
[internal]
include = extensions
include = outbound
Matthew Crocker wrote:
Does anyone have a working sip.conf for a SIP trunk to a Tekelec T6000
switch. I can get everything to work except the DTMF. The t6000
requires RFC1833 and I have that in the sip.conf but it still doesn't
seem to work.
RFC2833 not RFC1833
Naija Man wrote:
You can try VoipJet (http://www.voipjet.com)
A simple configuration in you extensions.conf as below will solve your
problem.
exten = _X.,1,SetCIDNum(1341212)
exten = _X.,n,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
1 is not valid as the first digit in a NANPA phone number.
Blind Transfer normally preserves CLID, no special hacks required.
Kristian Kielhofner wrote:
Aaron Daniel wrote:
Thanks... I did some research and found that it's actually not what I was
wanting (unless I missed something lol). I'm actually looking for a
way to
forward caller id
What is wrong with using the WaitForRing app?
Rich Adamson wrote:
Nick Ellson wrote:
I am in the process of learning my A1200P, and i would like an elegant
way to prevent it from answering the phone, but still make outbound
calls. I tried zap destroy channel 1 (which worked, but pissed off
Jay R. Ashworth wrote:
Assuming that you don't need to have a T-1 card in their for your
*trunks*. Since I'm told that you can only have, say, one Digium card
per chassis, this can be an issue.
You were told wrong. I have had up to FOUR Digium cards in a chassis.
3xTDM400P and 1xTE110P. I
Nick Ellson wrote:
Erm.. nothing that I know of, other than I do not yet know what that
means? :)
pbx-1*CLI show application waitforring
pbx-1*CLI
-= Info about application 'WaitForRing' =-
[Synopsis]
Wait for Ring Application
[Description]
WaitForRing(timeout)
Returns 0 after waiting at
Barry D. Hassler wrote:
We seem to be getting unexpected hangups on our * system, very
consistent when calling particular numbers that we can associate with a
clients phone system. These hangups generally occur when our call is
transferred within their system (to voicemail usually).
I'm
Naija Man wrote:
Hello all
I have an asterisk box running Asterisk 1.2.8 and I installed a digium
TDM2400 with 8 FXO ports. When I amke a call to the PSTN, the zap channel
answers, and teh call goes through if a PSTN is connected to the answered
port. However, if there is no dial tone in the
Both can cause random hangups. This is a well known issue. It even
says in the sample configs that these features are prone to false positives.
Alyed Tzompa wrote:
I'm curious... why will this work??
busydetect will just cut the line if there are 4 tones (les or more
Use IP addresses instead of hostnames in your Asterisk config. It
sucks, but that is the only way I know of.
Eric Bishop wrote:
When we loose Internet access (DNS) Asterisk basically halts until Internet
comes up even for internal registrations and calls. We are even running a
caching DNS
Rich Adamson wrote:
Eric ManxPower Wieling wrote:
Use IP addresses instead of hostnames in your Asterisk config. It
sucks, but that is the only way I know of.
Eric Bishop wrote:
When we loose Internet access (DNS) Asterisk basically halts until
Internet
comes up even for internal
Barry Fawthrop wrote:
Hi all
Anyone using a sonicwall firewall ?
I have been and then suddenly it drops UDP packets because SIP is no
longer on port 5060 but some random assigned port ?
Why ?
SIP is still on 5060, but the AUDIO (which is RTP) is on a dynamically
negotiated port. Now you
Asterisk does not support this, as it already has features for
multi-client configuration within a single Asterisk installation/process.
Douglas Garstang wrote:
I'd like to know if anyone has sucessfully managed to run multiple instances of
Asterisk on the same system.
- Did you run each
Best of luck getting multiple instances of Asterisk to play nice when
accessing Zap channels.
James Texter wrote:
Doug,
I actually see this as a pretty logical way to solve the problem.
Please keep us posted if you have any luck sorting out running multiple
instances, or mail me off-list
nice.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Monday, September 25, 2006 3:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk
Best
That error message is almost always because the two sides cannot agree
on a codec. HOWEVER, if you are using SIPura and G726, there is a
Makefile option for Asterisk to make it work.
Guy M Guyadeen wrote:
Our pbx is a Fedora Core 4 box running Asterisk 1.2.6. It has a public
IP and is
Dave Fullerton wrote:
Anthony Cennami wrote:
What's wrong with a channel bank? Can be a much cleaner solution with
greater room for expansion in the future.
Not to mention cheaper.
I never really considered one. I've never used one for that matter. This
system is really only a testbed. If
Craig Guy wrote:
I was afraid that may be the case - The issue I have with that approach
is how do you avoid manually mapping extensions to mac addresses in the
dialplan? Assuming I have a PRI with 100did and I want to use the last
4 digits of the DID as the internal extension, I want to use
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Perhaps you are tying to use wildcard destinations in your setup. This
does not scale.
Wildcard:
exten = 1234567,1,Dial(SIP/${EXTEN})
This does not scale.
Each extension should have it's own exten = line and
I have done looping playback and never experienced significant gaps.
Earle Clubb wrote:
John Marvin wrote:
Earle Clubb wrote:
Hello,
I'm trying to play an audio file to a phone an arbitrary number of
times. The audio is a five-second segment of a sine wave. I need
this to be played
Lacy Moore - Aspendora wrote:
On 9/20/06, Craig Guy [EMAIL PROTECTED] wrote:
[9580]
type=peer
auth=000413242fff:[EMAIL PROTECTED]
It would be
[MAC ADDRESS]
type=peer
...etc..
Or at least, that's how I interpreted what Eric said. I think that's an
excellent approach. THe phones are
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Your definition in the sip.conf would be defining devices according to their
MAC addresses. Your dial plan would call these devices based on extensions.
exten = 100,1,Dial(SIP/MAC) ; where MAC is the MAC address
Michel Zenone wrote:
Hi!
Is this possible to make asterisk follow the dial plan after executing
VoicemailMain?
Happens by default, unless the caller hangs up of course.
; Give voicemail at extension 3509
exten = 3509,1,SetVar(LOOP=1)
exten = 3509,2,Answer
exten = 3509,3,Wait(.5)
exten =
Earle Clubb wrote:
Hello,
I'm trying to play an audio file to a phone an arbitrary number of
times. The audio is a five-second segment of a sine wave. I need this
to be played repeatedly without gaps between playbacks. I've tried
doing this in the dial plan, e.g.:
exten =
Robson Ribeiro wrote:
Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. They are
installed respectively on banks 1,2,5 and 6. The problem I am having is that
when I make a call using the ZAP channel, I can hear perfectly but the
person on the other end is hearing my voice with
I suspect you need something like libcurl to build the CURL() function
when building Asterisk
Jerry Geis wrote:
You can always use the System() command in asterisk to call the curl
executable.
jerry
---
Ok, after requesting information to digium (no answer yet) and being
What is reported depends totally on the version of lspci and it's
libraries. It's cosmetic.
Robson Ribeiro wrote:
Erik, the TDM2400 is not in conflict with other interrupts BUT, i saw
something very weird:
Lspci
:04:09.0 Ethernet controller: Digium, Inc.: Unknown device 2400 (rev
11)
joea, j4computers wrote:
Ferguson, Michael[EMAIL PROTECTED] Wrote on: 9/20/2006 8:03 AM:
G'Day List,
Interesting article. Enjoy
http://www.networkworld.com/news/2006/091206-von-sam-houston.html?t5
Mike
The text states that asterisk cannot do secretarial functions, meaning one person,
. That
is why I think I must be missing something.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Tuesday, September 19, 2006 10:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
I suspect the article is referring to BLF, which is a traditional Key
System feature. It does not scale well in larger PBXs.
BLF support is not great (in Asterisk OR in phones) for SIP.
joea, j4computers wrote:
. .
What SPECIFICALLY are you trying to do that you are unable to do?
No
Jamin W. Collins wrote:
Doug Lytle wrote:
Jamin W. Collins wrote:
callprogress = yes
The only thing I'm iffy about is the above entry.
Maybe it's mistaking the progress as disconnect?
That does appear to have been the issue. We haven't had a new
occurrence of the random disconnects
Jamin W. Collins wrote:
Eric ManxPower Wieling wrote:
The comments in /etc/asterisk/zapata.conf didn't tip you off?
;
; On trunk interfaces (FXS) it can be useful to attempt to follow the
progress
; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
; progress attempts
Forum wrote:
Thanks for your response.
Unfortunately I still receive the same error - 'Error updating bootrom' - no
matter what version of sip and the bootROM I upload to the ftp site. I have
even used the latest release of the fimware - could I have somehow broke the
phone with a corrupted
a ton!
Brian
On 9/19/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Use type=user for inbound and type=peer for outbound. Have different
codec settings for each of them.
Mr. Jones wrote:
Hi Folks,
We're trying to roll Asterisk out to production and are having a few
complications.
Most
Jessee J Holmes wrote:
Get 3.2.2 from your reseller or installer. Bootrom 3.2.2 and firmware
2.0.1 is the latest available from Polycom. Your phone installer,
service provider, or reseller should be able to provide you with this
firmware.
I didn't think those worked with the 500 and 300
Damon Estep wrote:
Anyone encountered this on yet?
WARNING[23251]: chan_sip.c:2570 sip_write: Asked to transmit frame type
64, while native formats is 4 (read/write = 4/4)
Started after an upgrade from CVS 8/2005 to current 1.2.12.1
If I had a reference for what frame types 4 and
Anthony Cennami wrote:
I've also found similar problems with blindxfer -- such as when someone is
attempting to interact with an IVR using a '#' option. By default # seems
to transfer a call, but if you have blindxfer enabled with '#1' or ##, then
Asterisk hears the first # and waits for a
Use type=user for inbound and type=peer for outbound. Have different
codec settings for each of them.
Mr. Jones wrote:
Hi Folks,
We're trying to roll Asterisk out to production and are having a few
complications.
Most specifically we have G711 for our inbound origination, but would
prefer
.)
On 9/19/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Anthony Cennami wrote:
I've also found similar problems with blindxfer -- such as when someone
is
attempting to interact with an IVR using a '#' option. By default #
seems
to transfer a call, but if you have blindxfer enabled with '#1
Remove mpg123. In the Asterisk source directory type make mpg123 I
believe that make install is required to install it.
Jeronimo Romero wrote:
Running Asterisk 1.2.8 on Pound Key linux which I downloaded from Digium
site.
Uname output:
Linux localhost 2.6.13.4-1.x86.i686.cmov #1 Wed Nov
Christian Mohrbacher wrote:
In some cases : Yes.
But we have the following situation : We re using cisco 7960 phones in
each office (about 150 of them), but not every person has it's own
phone. Normally there are two employees in one office and they share one
phone, BUT have their own extension.
Rich Adamson wrote:
Julian Lyndon-Smith wrote:
I've got a cisco 7960, with (amongst many others) the following in the
RINGLIST.DAT file
Foghorn foghorn.raw
I can manually select this for the ringtone. However, I was wanting to
use a normal ringtone, with foghorn being used if the
That is not helpful in convincing my customers that there are many
companies using Asterisk.
Michael Welter wrote:
Yes. I don't use my customer's names on the list, so I can't say anything.
Porier, Jeremy M. wrote:
They're not the only ones :-)
Jeremy Porier
Senior Director of Information
exten = _X.,1,Playback(pbx-invalid)
exten = _X.,2,Goto(s,1)
Brian Candler wrote:
Hello,
In the process of finding my way around, I tried to get Asterisk to give a
recorded message if an invalid extension is dialled by a locally-attached
phone (FXS port on TDM400P)
Here's what I'm trying:
Douglas Garstang wrote:
Is there any documentation, maybe at voip-info.org, available for Asterisk 1.4?
Either in the form of _new_ docs, or docs that outline the differences and new
features that will be available in 1.4?
I'd like to avoid the months of trial-and-error that I went throo with
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Thursday, September 14, 2006 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] University switches to Asterisk
That is not helpful in convincing my customers
'i' is designed to be run when someone enters an invalid option in an
IVR during a WaitExten, Background, etc. i.e. AFTER a call has been
answered.
Brian Candler wrote:
On Thu, Sep 14, 2006 at 10:23:09AM -0500, Eric ManxPower Wieling wrote:
exten = _X.,1,Playback(pbx-invalid)
exten = _X
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