Turn off relaxdtmf in zapata.conf if that does not help play with the
rxgain, if that does not help, play with the txgain. If the volume is
too loud or too soft on zap channels, Asterisk can sometimes miss or see
double DTMF.
Brian Candler wrote:
On Thu, Sep 14, 2006 at 10:37:59AM -0500,
Doug Lytle wrote:
Jamin W. Collins wrote:
Doug Lytle wrote:
callprogress = yes
The only thing I'm iffy about is the above entry.
Maybe it's mistaking the progress as disconnect?
You should never, ever use callprogress or busydetect when using a PRI.
In fact, you could not use it in
Jamin W. Collins wrote:
Doug Lytle wrote:
Jamin W. Collins wrote:
Doug Lytle wrote:
callprogress = yes
The only thing I'm iffy about is the above entry.
Maybe it's mistaking the progress as disconnect?
The calls in question are connected for varying time frames. In some
cases 5
You should never have callerid=xxyyzz as some devices (as you just
discovered) choke on the since that is not a valid Caller*ID
character. I think some versions of the Cisco phone SIP firmware also
has a similar problem.
George Pajari wrote:
Just a short problem description/resolution so
Forum wrote:
I have a Polycom 500 that I am having issues with provisioning via an
ftp server. I have a bunch of 301’s that find the server and configure
without an issue. For some reason the 500 gives me an error that it
‘could not contact boot server’ and will reboot continuously. I also
You can't dial from exten = h
You could use an AGI with a .call file, or you could create the .call
file from inside the Asterisk dialplan. Heck, you could do it with
System() commands. See sample.call in the asterisk source directory. as
well as docs/ in the asterisk source directory.
What other ones are there?
Porier, Jeremy M. wrote:
They're not the only ones :-)
Jeremy Porier
Senior Director of Information Systems and Technology
Colorado Christian University
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
BerkHolz, Steven wrote:
All circuits are busy now makes perfect sense in my PRI trunk is full.
How do I stop asterisk from playing this recording when it is a
wrong/bad number?
I gat a call today that a user was trying all day to call a number in
Mexico and kept getting the above
Mike wrote:
Let's just take 1) and 2). Why is Asterisk not going into the i extension
like it should?
Because the i extension is for IVRs and things like that.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
Remove immediate=yes from /etc/asterisk/zapata.conf
Henrik Woffinden wrote:
That's exactly what happens:
When I pick up the handle, this is what I get:
-- Extension 's' in context 'from-inside' from '11' does not
exist. Rejecting call on channel 0/2, span 2
Do you know what to do in the
For some reason your phone is dialing an empty extension as soon as you
go off hook.
exten = s would be the same as exten = ''
Henrik Woffinden wrote:
immediate is already set to immediate=no, so that's not it.
Best regards,
Henrik Woffinden
Eric ManxPower Wieling wrote:
Remove
Mike wrote:
It's not a silly idea, I've been doing some sniffing and debugging with my
limited knowledge of the whole process. I found this in the debug stream
after having dialed 965).
Notice this line: SIP/2.0 484 Address Incomplete.
Is this what I was suspecting, that it knows it could
You do not mention the device you are using.
I'll assume Zap.
Enable three way calling and conference in zapata.conf then use FLASH.
Bart Fisher wrote:
It appears the only way to cause a 3-way call (or a screened transfer)
is by using conference - nasty
This mean SLT would need to transfer to
]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: September 8, 2006 5:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What don't I get about SIP?
Mike wrote:
It's not a silly idea, I've been doing some sniffing and debugging
busy. That`s what I want. And apparently, I can`t
get that.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: September 8, 2006 6:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
What IS your Polycom dialplan, and do you have the digit.impossiblematch
set?
Eric ManxPower Wieling wrote:
Then you are doing something else wrong. If the call gets to Asterisk
then the exten = lines I gave should match if they are in that context.
I use this all the time.
Mike wrote
wcfxo is used only for the X100P (and some clones). It is not used for
any other card. wctdm supports both FXO and FXS. Maybe you are just
confused about which module is associated with which channel.
Iván Vega R. wrote:
Upon further investigation, I tried the following:
lsmod | grep
. All good result.
When I dial 9-555-5 and wait, nothing happens
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: September 8, 2006 7:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Andrew Kohlsmith wrote:
On Thursday 07 September 2006 08:45, Matt Riddell (IT) wrote:
When and where did KPF admit to it being Digium's code?
Via psychic vibrations, obviously.
It's not Digium's code, IIRC. It's ITU code. You can download the ITU
reference code (in C) from the ITU for
Your problem is caused by using exten = _. DON'T DO THAT!
When Hangup() is being run then Asterisk will jump to exten = h Since
_. will match h it will go there.
Marco Mouta wrote:
Hi all,
I think i'm missing something very very basic! I want my calls with DID
48XX
(From pstn E1
Steven M. Sawczyn wrote:
Greetings, I finally got my Asterisk server up and running and now am in
the process of looking for a provider to use as a SIP trunk.
Unfortunately, I'm realizing that unlimited really is in fact limited --
Galaxy Voice's unlimited plan, for example, translates to a
This is why we set the SIP user ID to be the MAC of the device. It
helps us remember that EXTENSION != DEVICE.
Joshua Colp wrote:
Brandon Galbraith wrote:
I'm attempting to have multiple phones (geographically seperated)
register to a single extension, so when the extension is dialed, any
Dean Collins wrote:
Yes it is possible.
May I suggest you spend more time with www.voip-info.org
Or even better download www.trixbox.org on an old server to get an idea of how configs work.
Getting Trixbox would help him understand how Trixbox configs work, not
how Asterisk configs work.
andrutto wrote:
Hi,
Does anyone know how to solve this issue.
I have Asterisk box on public IP and three clients connected to it. Unfortunately they
are behind NAT (simple one-to-one). Those three clients can make outgoing calls hassle
free, but when I try to make a call between them
Untested:
exten = _,1,System(/bin/mail -s \Happy Message: ${EXTEN}\
[EMAIL PROTECTED])
This assumes you can send mail outside of Asterisk from that host.
Damien Gabrielson wrote:
I'm looking for a simple way to send email from a dial plan. I have
searched around quite a bit looking for
The call is not being picked up.
Manrique Feoli wrote:
thanks CF,
I did change the PRI CAUSE to unavailable, or reject.
only that it still shows Accepting overlap call from. just before
this -Executing SetVar(Zap/12-1, PRI_CAUSE=27)
does anyone knows if this call being picked
Luciano Moreira wrote:
I trying to setup a outbound trunk with IPSmarx. It's working, but when I make
a call, the ring dialtone stills ringing on my side, even after the other side
picksup the phone. I got a NOTICE message from Asterisk that I hope you can
help me:
--
Why make things so much more complicated than they need to be.
Asterisk has had support for doing this for ages. The term you are
looking for is contexts.
Brandon Galbraith wrote:
You could use Xen on Fedora Core 6 and virtualize each instance if you feel
the need is there.
On 8/16/06,
chan_zap won't build if Zaptel isn't installed when you build Asterisk.
Rebuild Asterisk after installing Zaptel.
Ken D'Ambrosio wrote:
Hi, all. I've just set up an Asterisk box -- to the best of my knowledge,
no differently than any of the others that I've set up. Only one minor
caveat:
That's kind of useless since progressinband only applies to digital
interfaces.
Try callprogress=no
Brodie Macleod wrote:
Try setting:
progressinband=no
in your sip.conf
-Brodie
On Monday 14 August 2006 10:20 pm, Don Fanning wrote:
Greetings List,
I'm having a strange problem with my
Rushowr wrote:
Hey Attilla, thanks for the update. I'm also working on a solution, but
unfortunately the system I'm working with needs the separate macros. I'll
update the list if anything gets worked out.
pbx-1*CLI show application gosub
pbx-1*CLI
-= Info about application 'Gosub' =-
Any reason that you can't set variables before you use Gosub, then
access them in the subroutine?
Attilla De Groot wrote:
On Aug 14, 2006, at 6:39 PM, Eric ManxPower Wieling wrote:
Rushowr wrote:
Hey Attilla, thanks for the update. I'm also working on a solution, but
unfortunately
Jason Parker wrote:
I think you misunderstand what qualify is/does. It appears that you believe that
qualify=1000 means that it'll send out a qualify packet every 1000ms. This isn't an
unreasonable assumption, but it is wrong. The qualify=1000 means that Asterisk will wait
1000ms for the
Douglas Garstang wrote:
Yes, it might be a problem in our situation. We have three Asterisk boxes in a 'cluster'. The sip.conf is identical on all three. In that case, all three of the Asterisk boxes in our cluster are going to send sip options messages to the phones, which is silly.
Only the
Curt Shaffer wrote:
I am having a weird issue with my zap channel (Digium TDM01B). Randomly it
appears that the POTS line is not seeing all of the digits passed. We have
to dial a 1 and the area code to call most numbers here, and we get the
error that we need to dial a 1 and the area code when
RetryDial (and DIALSTATUS) won't work on analog lines.
John Novack wrote:
Also many so-called legacy hybrid PBX switches have had this for many
a year
Hard to compete when well used features that have been around for 20
years are lacking
John Novack
Rushowr wrote:
The reason he might want
Also BUSY != BUSY Remember, pretty much any place in Asterisk quotes
are literal.
If you want to test if DIALSTATUS is equal to BUSY you want either:
GotoIf(${DIALSTATUS} = BUSY
or
GotoIf(${DIALSTATUS} = BUSY
Ira wrote:
At 11:54 AM 8/11/2006, you wrote:
Thanks for the
J. Oquendo wrote:
Asterisk Admin calls T1 Provider
Asterisk Admin -- T1 customer service -- Do you see 2125551212 dialing
in? T1 Cust Svce -- Asterisk Admin -- Nope
Asterisk Admin -- T1 Cust Svce -- OK, YOU try calling 2125551212 from
both on net and from off net.
Where off net is
pbx-1*CLI show application MailboxExists
pbx-1*CLI
-= Info about application 'MailboxExists' =-
[Synopsis]
Check to see if Voicemail mailbox exists
[Description]
MailboxExists([EMAIL PROTECTED]|options]): Check to see if the specified
mailbox exists. If no voicemail context is specified,
Ira wrote:
At 07:59 AM 8/10/2006, you wrote:
is there a way that asterisk can detect when someone speaks ? Like
answering a phone? i dont need speech recognition or anything like
that, just something that lets me know that any sound is originating
from the other end.
Play a recording that
Dean Collins wrote:
Is there a command for setting of a DID number?
Eg below I can set callerid
[custom-fromiaxfwd]
exten = s,1,Set(CALLERID(number)=2125316214)
Butw what I would prefer to do is set DID -like this (it doesn't work
[custom-fromiaxfwd]
exten =
number.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Thursday, 10 August 2006 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set DID
George Gardiner wrote:
Digium is not being given a whole load of money - the investors will
want a slice of the company and the future profits. That's how VC
funding works.
More like selling your soul to the Devil, actually.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Louis-David Mitterrand wrote:
Hello,
I am looking for the latest 1.6.7 Polycom firmware?
Is it available somewhere?
What issues are you experiencing that 1.6.7 fixes?
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
On Digital interfaces (PRI, SIP, etc) you are expected to check the
value of HANGUPCAUSE and play the correct message to the caller. The
telco does not do this for you on these types of interfaces.
Carlos Prieto wrote:
OK, sorry for not being so explicit.
Here is the console output when i
Generally yes, but keep a copy of the old files around just in case.
Stephen Murphy wrote:
Can you simply replace your current sip.Id and sip.ver files with the latest
firware files or is this dangerous?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
In my experience Yellow Alarm (AIS) on Tellabs indicates that the box
does not see a T-1 on one side.
marvin horst wrote:
Bad card?
I wired up another card and got the same result .
On the red Rcv In, have you tried swapping out the cable for the
opposite cable type. If xfer, change to
Adtran TA750 or TA850
Roger Workman wrote:
Can someone recommend a good FXS gateway/Channel bank that will intergrate
smoothly with * I need to port over 158 analog lines
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
Check that status of:
${RDNIS} and/or ${CALLERID(rdnis)}) in
/path/to/src/asterisk/docs/README.variables
C F wrote:
First my little Sunday story.
A client of mine with a big factory calls me up that he is trying to
call in to his place because the fire alarm went off. He is dialing
the
Andrea Spadaccini wrote:
Ciao Eric,
If you had a PRI (not just a T-1) AND your telco permits you to set
it.
Is there any hope to change the caller-id on a BRI line?
Sorry, I was being USA-centric. It's a bad habit to get into.
As I understand it, if you have a BRI and your telco allows
Marcus Carlson wrote:
Thomas Artner skrev:
Hi!
I have two extensions (25 and 26, and so two phones) for one person in
an office.
I can dial 25 or 26 and always both extensions are ringing. Thats okay!
exten = 25,1,Dial(Sip/25Sip/26)
exten = 26,1,Dial(Sip/25Sip/26)
The problem with this
The Tellabs cards I used were not configured for ESF/B8ZS when I got
them. If you have the Tellabs chassis, try connecting with a serial
connection.
Here's a copy of the manual: http://www.fnords.org/~eric/tellabs/ It's
in PDF format in 2 parts.
marvin horst wrote:
You can take a
If you had a PRI (not just a T-1) AND your telco permits you to set it.
hugolivude wrote:
That's what I feared. I could do it if I had a T1 is that right?
Thanks,
H
On 8/4/06, Steven Ringwald [EMAIL PROTECTED] wrote:
hugolivude wrote:
Redhat 9
Asterisk - 1.2.7
TDM 400 - 1 FXO, 2 FXS
I don't understand what the problem is. If you want to pass a variable
set the variable, but prefix it with __ (2 underscores)
Set(__DNID=${DNID})
Douglas Garstang wrote:
Oh... That's real nice. I was considering using SIP instead of IAX to trunk
calls between Asterisk boxes as IAX has some
Bart Fisher wrote:
I'm trying to detect when a T1 goes to Yellow or Red alarm. I noticed
these events will be displayed on the CLI.
What I'd like to do is cause an email to be sent when from a script on
these events, but somehow I would need to
capture the CLI outputs to detect messages
Pablo Mora wrote:
Did you saw my dialplan? I don't think I would have to add r.
You never want to add r option to Dial()
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
Pablo Mora wrote:
[outgoing]
exten = 0,1,Dial,Zap/g1
exten = 0,2,Hangup
exten = 0,102,Congestion
You NEVER want Dial,Zap/g1
You If you want to just get an outside dialtone you ALWAYS want a trailing /
Dial,Zap/g1/
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Eric ManxPower Wieling wrote:
Bart Fisher wrote:
I'm trying to detect when a T1 goes to Yellow or Red alarm. I noticed
these events will be displayed on the CLI.
What I'd like to do is cause an email to be sent when from a script on
these events, but somehow I would need to
capture the CLI
You do not have Caller*ID service, but Asterisk is configured to wait
for Caller*ID information. This information is delivered between the
1st and 2nd ring.
Joe Pokupec wrote:
Hey All,
I'm new to this list. I did some Google searching to find the answer but I
couldn't articulate the best
Koopmann, Jan-Peter wrote:
On Friday, July 28, 2006 3:12 PM Kai Ober wrote:
What about DIAL ( |M(macro-name))
and set the userfield in cdr during execution, ...
Set the userfield to what? That is the entire problem. ${CHANNEL} will give me
something like Zap/10-1. ${BRIDGEPEER} is empty. I
Zenone wrote:
But my question was, is it possible to free the channel if it rings too
long?
Yes. show application dial in the Asterisk CLI will show you where
the timeout goes on the Dial line.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
Bill Gibbs wrote:
Randomly, and this is very hard to debug because it happens so quickly
on outbound calls I get a one way screech, it's a steady tone that's
very loud. The remote end cannot hear it. You can hear the person
talking through the tone. I can't describe it but it's bad enough
] On Behalf Of Eric
ManxPower Wieling
Sent: Wednesday, July 26, 2006 9:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] One way screech or tone
Bill Gibbs wrote:
Randomly, and this is very hard to debug because it happens so quickly
on outbound calls I get
Then none of this applies.
Bill Gibbs wrote:
Ok, in my case it would be my Cisco 3660 since that's what talks to the
PRI. It talks sip to my Asterisk box.
Thanks!
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent
Someone connected to the Asterisk console using asterisk -r then typed
logger reload then exited the session.
Ira wrote:
This morning I found this message on my Asterisk Console. Does it mean I
should be concerned about the security of my system?
-- Remote UNIX connection
== Parsing
Sebastian Reitenbach wrote:
any idea what I can do? especially why it says it ignores the overlapdial
parameter, and why it is accepting them nevertheless?
are there any timing parameters to tell asterisk to wait a second longer for
the last digit? some rx.. tx.. parameters in the zapata.conf?
You are using quotes when you should not be. Notice the double quoting
of -- Executing NoOp(SIP/n-5d23, nothing) in new stack
Benjamin Stocker wrote:
Hi!
What's wrong with this?
exten = s,1,Set(myvar=nothing)
exten = s,2,Set(myvar = $[${CALLERID(num)} : ([a-z]+)])
exten =
Dean @ INKnBITs wrote:
I have setup the voicemail.conf as below, but I not receiving any emails.
Any thoughts?
voicemail.conf
[default]
3002 = 1234,Bob Wright,[EMAIL PROTECTED],,|attach=yes
I have also uncommitted the mailcmd=usr/sbin/sendmail -t
but that does not work.
Check the logs on
C F wrote:
Feelings are for the ignorant.
In any case, if you have trouble pinging your phone then you have
something wrong on either your network, or you got a damaged phone.
Here is my output from pinging a Polycom 501 while in a conversation
with app_voicemail:
Ping statistics for
[9507] is the incoming User ID. user=8407 is the outgoing User ID.
Do you really want them to be different?
Dial() will stop processing of the dialplan until the call ends. Do you
really want this?
r option to Dial will force a ringing sound to the caller, even if the
caller should be
= _.,2,Authenticate(8675301)
exten = _.,3,Goto(whateverdialcontext,whateverexten,whateverpriority)
replace Allison's recording for authenticate with your own.
Unless I am totally missing what you are trying to do.
Thanks,
Steve
Eric ManxPower Wieling wrote:
[9507] is the incoming User ID. user=8407
brandon kruz wrote:
youll have to decide what context this goes in
either
[internal]
or [incoming]
but i hope you can figure this out yourself
here is an idea
[internal]
exten = s,1,Answer()
exten = s,n,Playback(pbx-invalid)
exten = s,n,Hangup()
*sigh*
Playback will BY DEFAULT answer the
Maxx Lobo wrote:
An update:
I've found that I can leave the TE110P card in the server, unload the
module and issue an 'amportal restart' - this brings the
IVR/meetme/internal directory voice prompts all back again.
So it looks like the issue is directly related to the TE110P module
Kai Ober wrote:
Eric ManxPower Wieling schrieb:
Grandstream seems unable to produce stable firmware. They have tried
for *YEARS* and still people have to try many different versions of
the firmware to find one that actually works in their environment.
okay, i see, thx :)
i will try
Turn off 3-way calling on your SIP device. Set the dialplan on your SIP
device to not wait 15 seconds after pressing 9.
Pablo Mora wrote:
Hi all,
Iv' got a problem taking lines to call from SIP to PSTN. I have to press #
after 9 to get ringtone, otherwise I would have to wait above 15
No. In SIP these features are configured on the SIP device. If you
cannot disable three-way calling, or modify the dialplan on your SIP
device, then there is nothing you can do to fix the problem.
Pablo Mora wrote:
I really don't understand what you say.
I've been searching in my SIP
Kai Ober wrote:
Has somebody done that with a Grandstream GXP-2000 or a BudgetTone
100/101 ?
Has somebody even a list which SIP phones have this funtion?
SIPura supports it, Cisco ATAs support it. I assume that Cisco phones
support it.
I don't know about Grandstream devices since they
Shaun wrote:
Is there not a way to manually configure these phones or at least configure
them to use a diffrent tftp server rather than it attempting to ask the
dhcp/bootp server? For users at home with dinky linksys/dlink modems you
cant set a tftp/bootp server.
Of course there is. You
Kai Ober wrote:
Eric ManxPower Wieling schrieb:
I don't know about Grandstream devices since they are banned from our
network.
Banned? I didn't try any other devices, but whats wrong with the
Grandstreams??
Grandstream seems unable to produce stable firmware. They have tried
for *YEARS
Angel Diaz wrote:
Hi list,
What is ZapRas used for ?
I would like to use asterisk as a RAS server replacing a Cisco RAS server
where users calls to a number directed to asterisk, and here, asterisk
answer the data calls and assign an IP address via PPP to calling user.
ZapRAS allows
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
does Asterisk 1.2.7.1 supporting VAD? because i am
running my asterisk on VPS and i want to save
badwidth.
If Asterisk supports VAD (or silence suppression) please tell me how to turn it
of! I don't care about
I believe that with immediate=yes Asterisk does not know what number is
dialed and so that information is not available. Stop using immediate=yes.
Giordano Grandis wrote:
I cannot use it, I have the immediate=yes in my zapata, the extension will be always 's'
Thanks again for all
Giordano
Sounds to me that the incoming call is providing the wrong userid/password.
voiplist wrote:
Anyone have any thoughts on this?
On 7/13/06, voiplist [EMAIL PROTECTED] wrote:
We have a situation where the wrong account code is being passed from
Asterisk to our AGI and then on into the
Martin Joseph wrote:
I would love to see a simple explanation of how to update to the
latest,
including patches. Although I am not using queues, I have wondered
about this ever since the change over to SVN, and this seems a good
place to ask.
The latest release is 1.2.9.1 Anything in SVN
Doug Lytle wrote:
Dan Elder wrote:
Hey All, probably missing something really obvious here, but when our
users
are trying to dial the phone, asterisk timesout really quickly if they
don't
press the digits fast enough. Is there a global timeout value for dialing
See:
Wes Santee wrote:
Greetings all,
I'm on 1.2.9.1, and I'm trying to get a dialplan working that uses text
labels, but it's not working. For instance, take the following macro
snippet:
[macro-dosomething]
exten = s,1,GotoIf($[${MACRO_EXTEN:1:1} != 1] ? scid)
exten =
Wolfgang Zweimueller wrote:
Hi all,
when I receive incoming SIP calls on my Asterisk (1.2.9.1) where the
caller has a username in it's From-Address which also exists in my
sip.conf then my system answers with 407 Proxy Authentication
Required. If it's nonexistent username then callin works
Dean @ INKnBITs wrote:
I'm trying to build another asterisk server as I'm having a problem with
the current one. Unless anybody can tell me how to compile the meetme
app? Everything else works fine, asterisk just will not compile
meetme?!? (Under kernel 2.4)
Meetme will not compile if zaptel
Martin Joseph wrote:
On Jul 10, 2006, at 1:23 AM, yusuf wrote:
Hi all,
I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also
have 2 Digium FXO cards, and I have premicells connected to the FXO's
. Calls come in off the Sangoma E1 cards, from a Philips PABX. The
problem
Michiel van Baak wrote:
On 16:44, Sat 08 Jul 06, Florian Overkamp wrote:
Point is, you do not really need a CH1 or CCME license, you are free to
combine the Spare phone with a separate SIP license - the price is
identical. It is NOT OK however to use a Spare phone without any
license, as far
Outdated, but some of the info may still be current:
http://www.tek-tips.com/viewthread.cfm?qid=583069
Edwin Lam wrote:
hi folks.
does anybody know what's the phone number for SBC Nothern
California's 102-type milliwatt test line? (specifically
in 415 area code)
--
Now accepting new
There are different versionsof the polycom phones. Depending on the
actual part number it can come with MCGP, SIP, or H323. Polycom does
not support customer migration from one protocol to another. Get the
version with the SIP firmware.
Jim Freeze wrote:
Hi
I was about to order a
This has been my experience as well. I also posted the issue to this
mailing list, but has not responses. I have not come up with a
workaround. If I have time I'll try to write up a bug report, but it
will be a while. You are welcome to document the issue with as much
detail as you can and
Yes. It does not seem to cause any problems.
Douglas Garstang wrote:
Is anyone getting '500 Internal Server' errors back from their Polycom phones
when Asterisk sends a SIP NOTIFY message to them?
I called Polycom tech support, who where utterly useless.
Of course Polycom won't officially
I'm using the M() option to Dial() and having problems. In the
following dialplan example ANY digit exits the macro. When the callee
presses 1 the Noop(Reset AbsoluteTimeout(0)) does not get run. Does
anyone have any ideas as to what I'm doing wrong? Asterisk 1.2.x
[extensions]
exten =
sdgesa gaeharth wrote:
I have blindxfer = #1 set in features.Doesn't this means #1 is the same as
transfer - blind, correct? Both are blind transfers..
Is so, why when I transfer using #1 do I hear what extension the call was parked at but not transfer - blind?
#1 is, for whatever
Matt wrote:
Interesting, I have #2 setup to do blind transgfers, and if I do a
#270 it tells me the number seven one and then hangs up on me and
the user is left on park 71.
Maybe Asterisk knows that doing a blind transfer to park a call is a
silly and pointless thing to do and does a
Juan Luis Moyano wrote:
Hi all, I have a Soekris net4801-50 board with OpenBSD 3.9 where I've
configured a dhcp server and tested it with a regular PC connected
directly via a crossover cable with success. The problem comes when I
try to connect my IAXy device instead of the PC. I can see with
Mojo with Horan Company, LLC wrote:
I will agree that switching to the TDM card significantly helped my echo
and sound quality, I would take a second to point out that interrupt
sharing on your * server might cause crackling-like noises. Try
lspci -vb
and
cat /proc/interrupts
to see if
Vincent Delporte wrote:
Thanks Noah for the help, but... no go :-/
From: Noah Miller
ONE: You should answer an incoming zap line before doing anything with
it, so do this:
exten = s,1,Answer
exten = s,2,Dial(Zap/2/014XX)
When I try this, instead of using the Zap/2 interface to ring
701 - 800 of 1072 matches
Mail list logo
