Try different music. Compressed codecs are optimized for voice. MoH
that is primarily vocals may be better than non-vocal.
Darryl Dunkin wrote:
Does anyone have any opinions on the music on hold quality over G729?
The stock files seem to sound terrible over it, this is enhanced further
by
Tilghman Lesher wrote:
On Monday 26 November 2007 11:14, shadowym wrote:
On Monday, November 26, 2007 at 4:34 AM, Steven wrote:
I also believe that supporting asterisk via a Digium purchase is the more
right thing to do.
How is buying Digium the right thing to do? It is not like they are a
Eric ManxPower Wieling wrote:
Tilghman Lesher wrote:
On Monday 26 November 2007 11:14, shadowym wrote:
On Monday, November 26, 2007 at 4:34 AM, Steven wrote:
I also believe that supporting asterisk via a Digium purchase is the more
right thing to do.
How is buying Digium the right thing
Asterisk does not detect analog ports with no line plugged in. It does
not test for dialtone before dialing (this applies to all analog cards
except the X100P).
Rilawich Ango wrote:
It works if it specified the port exactly plugged to PSTN. I want to
clarify the dial command here.
As SIP is not Analog FXO, my comments do not apply to them. I have no
idea if or which analog adapters might detect line voltage or dialtone.
Paul wrote:
Do the SIP-FXO gateway devices do any better?
Eric ManxPower Wieling wrote:
Asterisk does not detect analog ports with no line plugged
Remove callprogress=yes from /etc/asterisk/zapata.conf There is a
REASON it is listed as EXPERIMENTAL. It simply does not work well.
Rilawich Ango wrote:
HI,
I have 2 TDM400s plugged in a PC. I failed to use same channels to
make a call to PSTN. It shows it can't establish connection
broadband Voice wrote:
Is Asterisk capable of sending text messages to a cell phone or is there an
application for that?
Yes. Any carrier that supports SMS over analog lines will work with the
Asterisk SMS application.
Generally carriers in the USA and Canada do not support SMS over analog
didier wrote:
Callerid(number) ? or callerid(num) ?
Grasshopper, you will find many answers you seek by looking in
/path/to/src/asterisk-1.4/doc/channelvariables.txt
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Vincent wrote:
Hello
Since SIP is a bit of a pain to use with NAT firewalls in the
way between clients and *, I'm considering IAX for soft/hardphones.
One thing though: Does the client have to also use UDP4569 as its
source port when connecting to * on UDP4569, or can the client use
broadband Voice wrote:
I looked through /etc/asterisk and could not find the folder sampl.call.
That is the Asterisk configuration directory. You are looking for the
Asterisk SOURCE CODE directory. If you installed from a package (.deb,
.rpm, etc) then you will have to contact the packager
Mark Quitoriano wrote:
that's the same question i got(regarding question 1). Is it possible
for PCI compatibility issue? i need to check for the motherboard specs
to post later :)
Hopefully someone will have someone smarter to say. This specific ioctl,
if it actually gets to zaptel, should
(Footnote: do I need a default context? I'd rather not having one... I'd
rather specify where
my calls go explicitly...)
I just set the context in [general] to be context=INVALID and not have a
context named INVALID.
___
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-11-16 at 22:58 -0600, Eric ManxPower Wieling wrote:
exten = _0111NXXNXX,1,Goto(${EXTEN:4},1)
exten = _NXXNXX,1,Dial(
Of course! Couldn't be any simpler. Almost!
User dials 6135551212, phone sends 01116135551212, above rules processes
as:
Goto(6135551212,1)
Fair enough
You have 18 channels defined in zaptel.conf, but 24 channels configured
in zapata.conf
Mark Quitoriano wrote:
Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5
im having an error message when in running asterisk with the tdm card
in.
here's the error from the console of
exten = _0111NXXNXX,1,Goto(${EXTEN:4},1)
exten = _NXXNXX,1,Dial(
Baji Panchumarti wrote:
I have no idea if this would work :
exten = _0111NXXNXX,1,Set(x=${EXTEN:4})
exten = _0111NXXNXX,n,Goto(${x},1)
exten = _0111NXXNXX,n,NoOp( Sorry it didn't work ! )
exten =
Erik Anderson wrote:
On Nov 13, 2007 11:21 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote:
Anyone knows what is wrong with this mailing list its a while all my new
posts appear as a reply (branch) for others post, is there any hints i
could prevent this issue??
I believe your posts are
Artifex Maximus wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have tried 'failed' and 'h' extension. No luck. Extension 'failed'
does not know anything about ISDNCAUSE and 'h' extension is
You need to look at the files in /path/to/src/asterisk/doc (or /docs, I
don't recall) there is much information there, including a file named
README.variables (1.2) or channelvariables.txt (1.4).
Vincent wrote:
On Sat, 10 Nov 2007 23:05:47 -0600, Eric \ManxPower\ Wieling
[EMAIL PROTECTED
Thomas Winter wrote:
Hi,
Iam running debian etch on thecus n2100 (Xscale 80219)
I do not have MoH because standard mpg123 gives only loud noise.
I can not compile mpg123 from asterisk because of option -m486.
Any way to get MoH running on this board.
What version of Asterisk?
Asterisk
Mark Quitoriano wrote:
Hi im using centos 5 what is the prerequisite to be installed before
compilling asterisk 1.4?
http://asterisk.org/support/install
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asterisk-users mailing
Vincent wrote:
On Sat, 10 Nov 2007 21:16:44 -0400, Baji Panchumarti
[EMAIL PROTECTED] wrote:
TrySystem is passing the cmd to (bash) shell, just give it a file match
skeleton as long as you don't have other msgNNN.wav files that
shouldn't be moved.
Thanks, but it won't do, as I need to get
Johansson Olle E wrote:
Excellent article!
Just a small comment to clarify:
We have three namespaces to work with: Channels, Extensions and
Devices/lines.
* Channels are ongoing calls.
* Extensions are what you dial to setup something, an entry in the
dialplan.
* Devices/lines
Steve Davies wrote:
On 11/9/07, Johansson Olle E [EMAIL PROTECTED] wrote:
Hi,
I got the following warning from CLI when I try to execute the Dial
command. It makes the call failed. Anyone can tell me what does it
mean and how to solve?
-- Executing [EMAIL PROTECTED]:61]
Doug Lytle wrote:
Eric ManxPower Wieling wrote:
Doug Lytle wrote:
ANI is not Caller*ID. A caller can block their Caller*ID, but not their
ANI.
This I didn't know, Thanks!
exten = s,1,ExecIf($[${CALLERID(num)} = ],Set,CALLERID(num)=401212)
I wasn't aware of the ExecIf
Doug Lytle wrote:
Jon Weisman wrote:
All,
If someone calls into my asterisk box and has a private number I would
like to set the callers id to a specific telephone number, only when
the ANI is missing, otherwise if present just pass it along. Any ideas?
[incoming]
exten =
Paradise Dove wrote:
hi
is there any way to find out that an fxo module is connected to telco
line or not?
not in Asterisk.
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To UNSUBSCRIBE or update
Baji Panchumarti wrote:
what happens if you replace the pattern matching expr with _.XXX
Not what you expect, that is for sure! The . pattern MUST be the LAST
character in the pattern. Once Asterisk sees a . in a pattern it
stops looking for any more pattern characters.
The following two wiki pages explain it much better than I could.
http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching
I'm not a fan of using the Wiki as a reference, but there really isn't
any info like this in
Hans Feringa wrote:
I understood that a timing device (ztdummy if no zaptel hardware is
present) was not necessary anymore with linux kernel 2.6.
When I enable iax2 trunking I get this warning
chan_iax2.c:8908 build_user: Unable to support trunking on user 'xx'
without zaptel timing
Jeng Yu wrote:
Hi Gurus!
Please excuse this pesky Asterisk rookie:-)
I just wanted to know which channel variable tells
asterisk the number of rings before an incoming call
on FXO channel is answered?
I looked through zapata.conf.sample and other places
and could not find
Doug Lytle wrote:
I've moved 1 of our facilities over from 1.2 to 1.4 two weeks back. So
far, the only issue that I've encounted is.
I have a scheduled CRON job that runs at 3am every Sunday, that issues a:
asterisk -rx 'restart when convenient'
The first Sunday that it ran, Asterisk
Look at the section starting on line 100 in
/path/to/src/asterisk-1.4.13/UPGRADE.txt
You should have read this file before upgrading to 1.4.
Jean-Yves Avenard wrote:
Dear all
I am trying to upgrade our asterisk from 1.2 to 1.4.x
There is something that now fails to work, reading the
Jean-Yves Avenard wrote:
Hi
On 11/5/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Look at the section starting on line 100 in
/path/to/src/asterisk-1.4.13/UPGRADE.txt
You should have read this file before upgrading to 1.4.
Excellent. Thank you!
I've added a WaitExten() just
Doug Lytle wrote:
Eric ManxPower Wieling wrote:
Doug Lytle wrote:
document was called UPGRADE.txt. You might want to look at that, in
addition to the 1.4 UPGRADE.txt, as there are many features that were
The document was reviewed and the appropriate changes to the dial plan
Eric ManxPower Wieling wrote:
Doug Lytle wrote:
Eric ManxPower Wieling wrote:
Doug Lytle wrote:
document was called UPGRADE.txt. You might want to look at that, in
addition to the 1.4 UPGRADE.txt, as there are many features that were
The document was reviewed and the appropriate
exten = _XXX**,1,Set(CALLERID(num)=${EXTEN:0:7)
exten = _XXX**,n,Goto(${EXTEN:8:4},1)
The first line extracts the Caller*ID number from the incoming digits
and sets the Caller*ID number to that value.
The second line goes to the actual extension or DID in your dialplan.
I have
Tony Plack wrote:
I have an interesting issue. I am running Asterisk 1.4 (SVN branch latest)
and
same with Zaptel.
If I load ztdummy, my audio in BackGround (or Playback) cannot be heard. If
I
rmmod ztdummy and restart Asterisk, Background works. What am I missing?
What things
Tilghman Lesher wrote:
On Friday 02 November 2007 12:07:48 Eric ManxPower Wieling wrote:
exten = _XXX**,1,Set(CALLERID(num)=${EXTEN:0:7)
exten = _XXX**,n,Goto(${EXTEN:8:4},1)
The first line extracts the Caller*ID number from the incoming digits
and sets the Caller*ID number
Yes, but if you didn't need MeetMe it would be silly to try to diagnose it.
I suspect ztdummy is loading correctly, but the system is not generating
interrupts for ztdummy sync to. Traditionally ztdummy used a USB
interface to get timing, but at some point I think it switched to using
the
Asterisk used to NOT support more than one arg to an AGI. I thought
this was fixed in 1.2, but since I now use dialplan variables to pass
data to AGIs, I don't know if it really was fixed or not.
Arpit Mehta wrote:
Hello * users,
I know that passing variable in the AGI script is by
James FitzGibbon wrote:
On 11/2/07, Arpit Mehta [EMAIL PROTECTED] wrote:
Hello * users,
I know that passing variable in the AGI script is by
exten = _.,1,AGI(simple_c_prgm|123|789) ; 123, 789 are arguments being
passed and simple_c_prgm is C code
Now how will I receive these variables
James FitzGibbon wrote:
On 11/2/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Sadly those docs cover the situation of this:
exten = 666,1,Set(MY_VAR=fred)
exten = 666,n,AGI(simple_c_prgm)
Not this:
exten = 666,1,AGI(simple_c_prgm|123|789)
Looking back over it, you're
Asterisk wrote:
Hello everyone,
I'm trying to bridge 2 SIP channels together via AGI script. The AGI Script
is written in C#. I'm able to get the unique name of the channel and insert
that into my database and need to know if i need to do anything in the
dialplan so i can run EXEC in the
It is up to your carrier to permit you to send different Caller*ID.
Many carriers that let you send your own Caller*ID number (you can't
send Caller*ID Name) but if you send an invalid number, they will drop
the call or override the Caller*ID number with the primary number of
your PRI.
;exten
The fact that you are sending 528 as your Caller*ID might be a problem
for your carrier.
Turbo Fredriksson wrote:
Quoting mail-lists [EMAIL PROTECTED]:
Turbo Fredriksson wrote:
We have now got our new PRI line (10 channels, 100 numbers) connected
and everything is working except the
Matt wrote:
This question is about 1.2.x asterisk. Please no flames, or you should
upgrade to 1.4.
Does anyone know what might be the cause for 'stuck voicemail's in
1.2.6asterisk? By stuck, I mean the phones show a voicemail, and if
you log in
you get you have 1 new voicemail, and if
Michelle Dupuis wrote:
Ok..so how would the CALLED and CALLERID ID be presented to Asterisk when
using PRI signaling.
The CALLING and CALLED numbers are sent automatically during the call setup.
CALLING NAME is usually sent right after the call setup happens.
Rizwan Hisham wrote:
I think you can use the 'ngrep' command to see the sip packets coming in
using the sip listening port. I dont know the exact command though, you will
have to lookit up urself. you will see the sip packets coming into ur system
and in those packets you can see the response
Steve Murphy wrote:
Although late model expression parser code allows ==, and treats it
like =,
and, and are interchangeable, and so also | and ||.
murf
late model is SVN TRUNK, 1.4, SVN BRANCH 1.4, or 1.2?
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Any echo you hear on pure IP calls is caused by the endpoint phone. You
cannot do ANYTHING about it on Asterisk.
Jonn Taylor wrote:
Any ideas ?
Jonn
Original Message
Subject: [asterisk-users] Internal LAN echo problem
Date: Wed, 24 Oct 2007 08:34:32
Jonn Taylor wrote:
Eric ManxPower Wieling wrote:
Any echo you hear on pure IP calls is caused by the endpoint phone. You
cannot do ANYTHING about it on Asterisk.
Jonn Taylor wrote:
Any ideas ?
Jonn
Original Message
Subject:[asterisk-users] Internal LAN
Rob Schall wrote:
Here's what I'm looking to do
exten = 10,1,Dial(SIP/1000SIP/1001,15,wW)
exten = 10,2,Voicemail(u1000)
This should ring both phones and they should keep ringing for the
alloted time before moving on. However, it appears that if one of the
phones is Busy, it returns
Voicemail will answer the line. 10 seconds is a pretty short timeout.
What you need to do is copy the CLI output of your failed calls from
BOTH servers and put them in this thread. Then we can SEE what Asterisk
is ACTUALLY doing.
Alan Lord wrote:
Eric ManxPower Wieling wrote:
Alan Lord
Alan Lord wrote:
Eric ManxPower Wieling wrote:
The remote server is where your problem is.
Thanks for the reply but I can call the extension in question normally
and it works fine. The problem is that the IAX trunk appears to be
answering before it knows if the physical destination
The remote server is where your problem is.
Alan Lord wrote:
Hello,
This message is similar to one I posted before, but with a different
subject line and I've revised the description to hopefully make it clearer.
The basic problem is I am trying to dial 2 numbers simultaneously using
ResponseTimeout was deprecated in 1.2 and removed in 1.4. Was this
information not in the upgrade.txt file in 1.2 and 1.4?
bilal ghayyad wrote:
Hi List;
My Asterisk version is 1.4 and I am trying to use the
ResponseTimeOut() application to control the response
time of the Background
Il Neofita wrote:
Hi,
I update from asterisk 1.2 to 1.4 and I have some problems.
In the extensions I used DIAL(SIP/100SIP/101,30,tTr) if I receive a
call from an external providers
now in 1.4 I recieve only one ring
What can I do to solve this problem?
You can start by removing the Fake
Tilghman Lesher wrote:
On Saturday 13 October 2007 18:08:49 Turbo Fredriksson wrote:
Quoting Philipp Kempgen [EMAIL PROTECTED]:
exten = s,1,Answer()
exten = s,n,Goto(s-${DIALSTATUS},1)
This still doesn't make sense because you did not Dial()
before jumping based on ${DIALSTATUS}.
Ok, make
Jeng Yu wrote:
Hi All,
I have a quick one for you. Is there a way to mask
(i.e. combine) the flags in the Dial() application? In
other words, a way to do something like
Dial(Zap/1,10,d|t|f)
to get the effects of the three flags together in one
shot? I have a need to combine the
Julian Lyndon-Smith wrote:
Hijacking a thread again - the only way I can post to the -user list is
by replying to another thread. (btw, this is getting really annoying.
Please, please, please, Digium, sort the filters out!)
You seem to be subscribed to the list as [EMAIL PROTECTED]. Is that
Andreas Bayer wrote:
is there a way to turn of SIP METHOD OPTIONS in asterisk?
I have a sip pbx which ignore Sip Option Messages from a unknown user.
Asterisk send Option Messages to peers with From: [EMAIL PROTECTED] The sip
server expects From: [EMAIL PROTECTED] server domain].
So i
Victor wrote:
I need to process a number of lines of code in the dialplan before answering a
call. Can standard ring back tones be played to the caller while this is
happening prior to answering the call. Which commands would facilitate this?
I strongly doubt those lines are going to take up
Steve Totaro wrote:
Eric ManxPower Wieling wrote:
Steve Totaro wrote:
Steve Totaro wrote:
I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it
worked fine except for audio issues that I believe are directly related
to IAX2 in version 1.2.x. I have four PRIs and want
Steve Totaro wrote:
Steve Totaro wrote:
I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it
worked fine except for audio issues that I believe are directly related
to IAX2 in version 1.2.x. I have four PRIs and want a separate context
for each going into the PBX. This
Christoph Adomeit wrote:
Hi,
I have the following problem: I want asterisk to dial
a chain of n-numbers until somebody picks up the line.
I am using Digium E1 Hardware (zaptel) for dialing out.
Dialing a Chain is basically no problem, I use somwthing like:
dial(no1,50)
dial(no2,50)
Let us not forget that AEL cannot be stored in a database therefore
rendering you unable to utilize realtime.
AEL converted into standard extensions.conf syntax in the dialplan.
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phone to dial that
extension, and you will
have
dial tone on selected line,
then as a traditional PBX you can send any digits to
your provider.
On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:
ignorepat continues dialtone after a leading digit
has been dialed
on
FXS ports
dial tone on selected line,
then as a traditional PBX you can send any digits to your provider.
On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
ignorepat continues dialtone after a leading digit has been dialed on
FXS ports. How does ignorepat help this guy?
Al lists wrote
Kenneth Padgett wrote:
Dear Atacomm Customers,
We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm
and its parent company Ataractic Corporation has ceased
operations. We appreciate the 7 years of loyalty and support from
our customers. We sincerely regret any adverse effects
ignorepat continues dialtone after a leading digit has been dialed on
FXS ports. How does ignorepat help this guy?
Al lists wrote:
ignorpat is your friend
On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote:
Dear List;
How can
Unfortunately 1.2 is no longer getting bug fixes (except for security
fixes). You will have to manually apply the patch for 1.2.
Yes the 1.2 maint policy sucks for many people, including me.
Doug wrote:
http://bugs.digium.com/view.php?id=10535
It is quite serious, costing us money and ill
Doug wrote:
At 14:14 10/1/2007, Eric \ManxPower\ Wieling wrote:
Unfortunately 1.2 is no longer getting bug fixes (except for security
fixes). You will have to manually apply the patch for 1.2.
Yes the 1.2 maint policy sucks for many people, including me.
Hmmm. Many people believe
The only time I have had this problem is when there was a version
mismatch between Zaptel and Asterisk. Once I resolved that issue
(latest asterisk + latest zaptel + reasonably recent wanpipe) everything
worked for me.
Alvin Austin wrote:
Hi everyone,
I'm trying to get a Sangoma A101D-X
There is no such thing as a SIP Trunk in Asterisk. Nope. It does not
exist. Some people (seems to me mostly GUI people) use the term SIP
trunk to mean SIP friend/user/peer.
John covici wrote:
I am not an expert on chanspy, but it seems to me spying on the trunk
would not work very well,
Those variables were deprecated in 1.2 and removed in 1.4. You should
read both the 1.2 and 1.4 UPGRADE.txt files. Also read README.variables.
Peter Kranz wrote:
When receiving inbound calls from a Vonage Softphone extension, I'm unable
to view/maniupulate calledid data. but it shows up in
Nicholas Blasgen wrote:
exten = 555,1,Dial(Local/1234567890/n)
note the /n
I'm going to try this in a bit (can't hurt anything, might as well), but I'd
like to understand you're reasoning. You're dialing an extra extension?
I'm also going to be trying this with Asterisk 1.6 TRUNK to
We use the MAC of the phone (all lower case) with a -a, -b, -c, etc
tacked onto the end of the MAC to specify the line appearance.
One thing you MUST remember is that a sip.conf entry is NOT an
extension. Extensions are totally different from sip.conf entries.
sip.conf entries are DEVICES.
If the #AsteriskNOW channel is dead on IRC that does not mean you can
bring your problems to a channel dedicated to Asterisk (i.e. no GUI).
Go ahead and use AsteriskNOW, but don't pester the people in #asterisk,
most of which have never used it and many have never even heard of it.
All the
Try adding canreinvite=no to the sip.conf entries for UA or UB.
Asterisk can't timeout the RTP if the RTP is not being handled by Asterisk.
Arun Kumar wrote:
Hi All,
UA Asterisk Server - UB
if there is no rtp for a specified number of minutes / seconds then I want
to
Vieri wrote:
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
You can probably get an answer to that if you enable
and log debug
messages of Asterisk .
Thanks, I thought that a pri debug was enough but now
I have the missing information:
Sep 16 12:37:28 VERBOSE[19175] logger.c: --
SIP response 486 is Busy Here according to RFC 3326. Polycoms at
least (and I think Cisco phones) do not send back a different message
depending on if DND is enabled .vs. the line appearance simply being busy.
Personally I can't see how the people that designed SIP could justify
not being
Anthony Francis wrote:
Eric ManxPower Wieling wrote:
SIP response 486 is Busy Here according to RFC 3326. Polycoms at
least (and I think Cisco phones) do not send back a different message
depending on if DND is enabled .vs. the line appearance simply being busy.
Personally I can't see how
Anthony Messina wrote:
I am working on getting freenum.org ISN/ITAD numbers integrated into my
exiting dialplan however I am having trouble getting the extension matches to
work as expected.
I would like to be able to do something like:
exten = _X.*.,1,Macro(isn-outbound...)
Where I
Jared Smith wrote:
On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote:
. matches any number of the preceding character, change it to _X.*X.
That still won't help. Once the Asterisk pattern matching parser sees a
period in the pattern, it ignores anything after it. (I'm not exactly
Jeremy Wadhams wrote:
In Asterisk 1.4, is there any way to force new users to configure
their mailbox? I'm thinking a simple IVR that holds a user's hand
through changing their PIN, recording their name, and setting up one or
both greetings, the very first time they use the account.
I
This error message means there was corrupted data received from the
card. This can be caused by many things. The T-1/E-1 could be having
errors, there could be a bad SmartJack (the telco box on your wall), it
can also be cause by lost or delayed interrupts.
As you can see below your T-1/E-1
HNAGUPCAUSE is more specific. Cause 31 is Normal, Unspecified end of
call. Chances are it is a harmless message and is a telco caused issue.
See
http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf
gincantalupo wrote:
Hi satish,
I get that error too (my
Looks like the Alcatel is sending back a busy. Check the value of
HANGUPCAUSE with a Noop as the priority after the Dial. You may also
want to do a pri debug span X to see the actual Q.931 ISDN messages that
are exchanged.
Vieri wrote:
An Asterisk extension calls an Alcatel extension via a
I've not seen an EM/Wink that supported Caller*ID. You can fake it by
sending something like *CALLERID*DID and then on the far end break that
out and set the callerid and goto the DID.
Willy Wouters wrote:
Hi,
Normally my T1 implementations are PRI.
However, I do have a customer who uses
Polycoms work just fine behind NAT.
Mike Clark wrote:
Chris Mason (Lists) wrote:
Mike Clark wrote:
Yes, the Asterisk boxes were on private addresses. The Polycoms are also
behind a NAT. Yes, I tried using externip in sip.conf and this allowed
registration, and calls to be placed, but
;)
- Original Message -
From: Eric ManxPower Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 2:43 PM
Subject: Re: [asterisk-users] Linux-HA and Asterisk
Polycoms work
THAT is an issue with externip= and localnet= not being correct.
Mike Clark wrote:
Eric ManxPower Wieling wrote:
Polycoms work just fine behind NAT.
Yep, we have lots of Polycoms behind NAT working fine with Asterisk
servers on *public* IPs. However, with the HA cluster, we had
Rizwan Hisham wrote:
well he does not have access to hi sip settings, so he cant edit the
host=differentIP every time he moves or registers from anyother place.
Actually he should be able to register from anywhere in the world but once
he has registered with us, i dont want anyone else to
It is not possible to do this the way you want. Most phones will
display the called name if that name/number is in the phone's directory.
Peder @ NetworkOblivion wrote:
We have users with Cisco 7900 phones running sip. When user A calls
user B, we want user B's name to appear on user A's
\ManxPower\ Wieling [EMAIL PROTECTED]:
It is not possible to do this the way you want. Most phones will
display the called name if that name/number is in the phone's directory.
Peder @ NetworkOblivion wrote:
We have users with Cisco 7900 phones running sip. When user A calls
user B, we want
The SIPuras support it, Asterisk analog does not, as far as I know.
Matt wrote:
The answer, I believe, is yes... but I'm not sure how We had
this working on some SPA-2002s from Sipura... but then after an
asterisk upgrade it stopped working. I'm not sure if it's a setting
in the ATA or
The correct term for this tone is howler. I'm surprised it is not in
indications.conf
Robert Lister wrote:
On Tue, Sep 04, 2007 at 07:41:53AM -0500, Anthony Messina wrote:
well i'm looking for the feature that the telco provides where, if you've
left
the phone off-hook for 60 seconds or
Tzafrir Cohen wrote:
On Sat, Sep 01, 2007 at 01:23:47PM -0500, Eric ManxPower Wieling wrote:
Tzafrir Cohen wrote:
On Sat, Sep 01, 2007 at 07:30:37AM +, wassim darwish wrote:
Hi:
Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel are
being installed to /lib/modules/2.6.8.1
Tzafrir Cohen wrote:
On Sat, Sep 01, 2007 at 07:30:37AM +, wassim darwish wrote:
Hi:
Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel are
being installed to /lib/modules/2.6.8.1-12mdkcustom
how can i fix this up, any one have an idea?
What is the output of:
rpm -qa |
Many of these issues only appear when you put it into production and/or
after a period of time. Most of the crashes I've seen are like this. I
simply to not have the resources to run simulations to try to find these
types of issues.
I can do one of several things. I can simply not upgrade
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