I think a real simple solution is to use a variable and you can change it
when ever you want. So you can do
exten = XXX,X,Musiconhold(${VARIABLE})
and then you can create an extension that will change the value of
${VARIABLE}
something like:
exten = 123,1,Answer
exten =
On Monday 11 June 2007, Thomas Stein wrote:
Hello.
Is it possible to change the defined moh sound file within an extension?
I have:
exten = 18,1,Answer
exten = 18,n,Wait(3)
exten = 18,n,SetMusicOnHold(durchwahl)
exten = 18,n,Dial(SIP/118,15,m)
exten = 18,n,Hangup
It's possible to
On Tue, 12 Jun 2007, Deepak Naidu wrote:
We are planning to buy a wifi SIP phones to work with Asterisk 1.2-18
setup. I would like to get feedback views regarding Linksys WIP300
WIFI IP Phone or any other wifi phones which has been stable.
Please check the archives - there was a lot of
On Tuesday 12 June 2007, Thomas Stein wrote:
It's possible to change moh behaviour after the dial command.
exten = 18,1,Answer
exten = 18,n,Wait(3)
exten = 18,n,SetMusicOnHold(durchwahl)
exten = 18,n,Dial(SIP/118,15,m)
exten = 18,n,SetMusicOnHold(default)
Oops. Wrong. What works is:
Hi
does any one come acrosss the tool
which convert the normal config files of sip.conf, extension.conf...etc
will convert automatically to mysql. with with any problems
if yes, kindly point me to that toolm which iam looking
thanks
ram
___
Hello,
I'm using a dial command to make several phones ring. I use this format :
Dial(SIP/4029SIP/4030,15,tTr)
As soon as one of the phone is picked up, all the others should just stop
rining.
But the fact is that they continue to ring for several seconds (4-5s), and
this is quite annoying as
Your problem is 1.2.X. It has a poor DTMF. I broke my head for a few weeks
on the same issue. (My issue was Polycom + Asterisk 1.2.X + provider X. The
interesting thing is that the Polycom + asterisk worked fine. The issue I
had was just when I was calling out through a specific provider.) Any
Hi,
I posted earlier about 'sip show registry' not showing any trunks and there
seeming to be no attempt by one of my systems to register with our service
provider. Following some notes elsewhere I ran 'myisamchk *.MYI' on the
mysql database and it came up with an error in sip.MYI, which I
I now have both 1.2 and 1.4.4 boxes.
Each asterisk is behind NAT on a fixed ip with all the externip,
nat=yes, and forwarded ports etc set up.
I have two multiline SIP phones, Linksys 941 and a Polycom ip500.
THese both work normally on the 1.2 box. I took the exact configs from
sip.conf and
I'm using the Dial application in the extensions file with the G option to
execute an AGI script after the Dial (I need to track the call status) as
follows:
exten = _X.,3, Dial({DIAL_STRING},,G(_X.^4))
exten = _X.,4, Answer()
exten = _X.,5,AGI,agiScript.php
The problem is that testing between
Hi Guys
I am just looking to see if you can help me. I have been investigating the
SPA400 and it seems to run asterisk for the voicemail system. Does anyone
know if it could be programmed to also talk to the FXO ports?
___
--Bandwidth and
You could use it as a usually FXO Gateway. I have tested and it works fine.
2007/6/12, MBIT Technologies [EMAIL PROTECTED]:
Hi Guys
I am just looking to see if you can help me. I have been investigating the
SPA400 and it seems to run asterisk for the voicemail system. Does anyone
know if
Yes you can but I am looking to see if asterisk can use the FXO ports
directly. The proprietary SIP to FXO gateway on there is not as good as if
you could use asterisk with the FXO ports directly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Siemens Gigaset SL75 are just Great!
On 6/12/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Tue, 12 Jun 2007, Deepak Naidu wrote:
We are planning to buy a wifi SIP phones to work with Asterisk 1.2-18
setup. I would like to get feedback views regarding Linksys WIP300
WIFI IP Phone or
Hi,
I write the CDR of my Asterisk 1.2.17 server in MySQL database
using cdr_addon_mysql.so.
Now I'm trying to write some custom values to userfield column by
the SET(CDR(USERFILED)=SOME_TEXT) sintax, but nothing gets writeen in
MySQL cdr table!!
Why? I'm I skeeping something
Ricardo:
By default, the module cdr_addon_mysql doesn't store the userfield in the
table. You should configure this option by adding to the configuration file
cdr_mysql.conf the next line:
userfield=1
Hope this helps.
Tomás.
2007/6/12, [EMAIL PROTECTED] [EMAIL PROTECTED]:
Hi,
I write the CDR
[EMAIL PROTECTED] wrote:
Hi,
it seems to be this, the purpose of this *Set(CDR(name))* sintax,
right?!
This should be:
SetCDRUserField()
show application setcdruserfield
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
Hello everybody, I have installed the Billion ISDN on a Debian machine.
I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So
I connect the Billion ISDN to the ISDN Box and I call from a extension to
outside.
But it doesn't work, that is what I have in the CLI:
*CLI
Hello everybody again.
I have a warning message in the CLI:
*CLI Jun 12 17:34:29 WARNING[10593]: chan_zap.c:8463 pri_fixup_principle:
Call specified, but not found?
*CLI Jun 12 17:34:29 WARNING[10593]: chan_zap.c:8463 pri_fixup_principle:
Call specified, but not found
I don't know what it
Thanks to all that replied!
Both suggestions you made, work just fine!
Regards,
Ricardo.
Quoting Doug Lytle [EMAIL PROTECTED]:
[EMAIL PROTECTED] wrote:
Hi,
it seems to be this, the purpose of this *Set(CDR(name))* sintax, right?!
This should be:
SetCDRUserField()
show
Dear Group,
I have a scenario where I would like to change the caller ID based on
the number dialled;
For example;
;Outbound UK and London Calls
exten=_8.,1,Set(CALLERIDNAME=0207100)
exten=_8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten=_80039.,1,Set(CALLERIDNAME=0039024070)
On Jun 11, 2007, at 7:26 PM, Zeeshan Zakaria wrote:
Also I recommend going with Sangoma. I hear a lot of bad stories about
digium cards imcompatibility with certain motherboards and conflicts
with USB modules on the motherboard, and conflicts with IRQs. Thats
why When I went for PRI, I used
On 6/12/07, Doug Lytle [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Hi,
it seems to be this, the purpose of this *Set(CDR(name))* sintax,
right?!
This should be:
SetCDRUserField()
show application setcdruserfield
Actually SetCDRUserfield is deprecated in 1.4, and you will get
I have GXP-2000 phones running against Asterisk 1.4. All phones are running G729 and this is witnessed by the fact that the phone shows the G729 codec.
I dial the first phone, place it on hold, dial the second phone, press CONF and the other line. The first connection goes away and the second
I think the problem is related to pattern matching your outbound numbers.
You have _8. for your first one listed and it will match for any of the
other 800 numbers you have so it goes there. Take a look at this webpage on
voip-info.org:
on sip show channels I do get a lot of entrys like
192.168.1.47 11 07ba5a490b3 00102/0 unkn No
Init: INVITE
192.168.1.47 11 19090f115b8 00102/0 unkn No
Init: INVITE
192.168.1.47 11 7d8b8fde46f 00102/0 unkn No
Init: INVITE
How do they
Hello,
I recently installed chanskype on my asterisk box and it works like a
dream, can phone out.
But no idea how to setup the incoming calls, every time I phone my skype
name it just connects and disconnect the call right away.
I get the following on asterisk -rvv
Verbosity was
Hello,
Have any one had any success installing asterfax.
Have installed and it seems like its working, it accepts the fax that
was sent via email, dail's out but then just freezes.
It then complains about rcfax that needs to be batched or it closes
asterisk down.
Any advice.
Kind Regards,
On 6/12/07, Tony Plack [EMAIL PROTECTED] wrote:
I have GXP-2000 phones running against Asterisk 1.4. All phones are
running G729 and this is witnessed by the fact that the phone shows the G729
codec.
I dial the first phone, place it on hold, dial the second phone, press
CONF and the other
Hi Tony,
Since G.729 codec requires a license unless using pass-thru, normal
calls probably pass without transcoding in your server to the other
end, but when transferring calls, Asterisk needs to transcode the RTP
flows, and that needs to be licensed!
You can however use other codec on
I ran the gambit and eventually, against my better judgement, I
finally broke down and installed HylaFax+IAXModem and I have had
absolutely zero problems with it. I'm extremely impressed.
This is the how-to I followed. A small amount of the instructions
aren't extremely clear; there is a
Hi,
Our operator frequently gets requests to transfer a call directly to
voicemail in order for the caller to leave a message without disturbing
the callee. Basicly, I'm looking for a blindxfer to vm.
My first thought was to prepend a digit (eg 7) to the extension but this
does not fit well
Hello Drew;
Assuming your extensions is 105 let's see the dialplan:
exten = 105,1,Dial(SIP/105,30,Tt)
exten = 105,n,Hangup
exten = *XXX,1,Answer
exten = *XXX,n,VoiceMail(${EXTEN:[EMAIL PROTECTED])
exten = *XXX,n,Hangup
I think this should work for what you want.
Regards;
Leonardo Kamache
Drew,
I've written a tiny patch that duplicates the functionality of a blind
transfer but sets a variable (VMXFER) before transferring. The dialplan
simply looks for the VMXFER variable, and if found, will direct the call
directly to voicemail.
This way, instead of making the operators learn
Am Dienstag, den 12.06.2007, 09:57 -0400 schrieb Shad Mortazavi:
Dear Group,
I have a scenario where I would like to change the caller ID based on
the number dialled;
For example;
;Outbound UK and London Calls
exten=_8.,1,Set(CALLERIDNAME=0207100)
On Tue, Jun 12, 2007 at 03:54:13PM +0200, Josu Lazkano wrote:
Hello everybody, I have installed the Billion ISDN on a Debian machine.
I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So
I connect the Billion ISDN to the ISDN Box and I call from a extension to
Hello,
Has anyone tried latest HPEC software with Asterik 1.2 ?
It was published a couple of weeks ago to improve clipping issues.
Regards
___
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To UNSUBSCRIBE or update
Shad Mortazavi wrote:
Dear Group,
I have a scenario where I would like to change the caller ID based on
the number dialled;
For example;
;Outbound UK and London Calls
exten=_8.,1,Set(CALLERIDNAME=0207100)
exten=_8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
You must have forgot to read
G729 and annex A differ in the perceived quality and cpu requirements. The
annex A version requires less CPU at the cost of loss of quality. The
bitstreams are compatible with each other in that a G729A codec can decode a
G729 stream and vice versa.
Craig
-Original Message-
From:
Of course, thanks for respose Tzafrir.
Here is my zapata.conf:
[trunkgroups]
[channels]
language=es
context=default
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan = unknown
overlapdial = no
usecallerid = yes
callerid = asreceived
callprogress = no
hidecallerid = no
nationalprefix
Hello Steve,
Thank you for the detailed answer. The first solution you mentioned
seems to be good enough for me. So I have to wait for Asterisk 1.6.
That's bad, but I have to wait.
My hope was a way with Asterisk 1.2 (or 1.4) and CDR-functions like
ForkCDR or with some local channels. I worked on
We are trying to use a softphone from a location that is behind a
firewall. We are using x-lite as the softphone.
So far, we've been able to get the phone to register with the asterisk
server, and it can receive voice from the asterisk server (IE,
voicemail, etc).
However, asterisk can't hear
Paco Brufal wrote:
I have an asterisk 1.2.18 working fine, the only problem is that all
applications that play audio, sound like tremolo or vibrato, but
musiconhold plays fine.
The same audio file (wav, mp3, ...) works fine with Musiconhold()
but not with Playback() or Background()...
Do you
theoretically, with canreinvite=yes, it's phone - phone. with
canreinvite=no, it's phone - asterisk - phone. BUT there are a few
reasons which canreinvite=yes will not be this way. If for example you
have a T or a t in the Dial string, asterisk will _remain_ in the media
path so it can
In [general] section:
externip=your_extern_ip_address
localnet=your_local_net/bits i.e. 192.168.0.0/24
Try this...
On 6/12/07, Rob Schall [EMAIL PROTECTED] wrote:
We are trying to use a softphone from a location that is behind a firewall.
We are using x-lite as the softphone.
So far,
Hello,
Sorry if this is a real dumb question but when sending a fax and the end
user does not enable fax on their side and then just hangs up does not
force asterisk to end the call.
So it keeps the trunk open until its killed by a Flash Operator.
Please assist if any one understands me.
Hi all,
I have the following in my extensions.conf:
exten = s,4,GotoIf($[${CALLERID(number)} = 8585979857 |
8585970327]?15:5)
The numbers listed above are known spammer numbers. However, when I call
from any other CALLERID, it still directs me to s,15 which is the
Hangup() application. Here are
On 12 Jun 2007, at 17:53, Rob Schall wrote:
We are trying to use a softphone from a location that is behind a
firewall. We are using x-lite as the softphone.
Alternatively you could use an IAX softphone. They generally don't
have a problem with NAT or firewalls.
Tim Panton
No luck. Still no outbound sound.
Leonardo Kamache (Gmail) wrote:
In [general] section:
externip=your_extern_ip_address
localnet=your_local_net/bits i.e. 192.168.0.0/24
Try this...
On 6/12/07, Rob Schall [EMAIL PROTECTED] wrote:
We are trying to use a softphone from a location
This is probably what we'll have to do. We wanted to try to use all SIP
though. As I read through the documentation, it seems possible though.
Not sure where i'm off.
Rob
Tim Panton wrote:
On 12 Jun 2007, at 17:53, Rob Schall wrote:
We are trying to use a softphone from a location that is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steve Finkelstein wrote:
Hi all,
I have the following in my extensions.conf:
exten = s,4,GotoIf($[${CALLERID(number)} = 8585979857 |
8585970327]?15:5)
The numbers listed above are known spammer numbers. However, when I call
from any other
Moj,
Does this mean that even out-of-band DTMF still gets sent
SIP-phone--SIP-phone without Asterisk hearing them? (eg RFC DTMF,
can't remember the number right now)
Forgive me for butting into this thread but this is interesting...
Cheers
Alex
On Tue, 2007-06-12 at 09:21 -0800, Mojo
On Tue, Jun 12, 2007 at 06:13:19PM +0200, Josu Lazkano wrote:
Of course, thanks for respose Tzafrir.
Here is my zapata.conf:
[trunkgroups]
[channels]
language=es
context=default
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan = unknown
overlapdial = no
usecallerid =
Hi people!
Sorry for bringing up some annoying issue.. yes, it's AMD again...
But I was searching the last days for a solution for my problem and
didn't really find anything. Now I'm hoping that someone of you has
maybe an idea for me. :)
My setup:
-
I use the Asterik Manager API to
On Tue, 12 Jun 2007, Steve Finkelstein wrote:
Hi all,
I have the following in my extensions.conf:
exten = s,4,GotoIf($[${CALLERID(number)} = 8585979857 |
8585970327]?15:5)
The numbers listed above are known spammer numbers. However, when I call
from any other CALLERID, it still directs me to
Here's an update.
We now see RTP traffic being generated by the laptop using the
softphone. However its destination address is the asterisk servers
internal IP address and not the outside address it needs to be pointing
to. We set the externip setting, but that doesn't seem to change xlite's
Hello All,
How can I do to record my asterisk's CDR in a Oracle database?
I have to use unixODBC?
Can anybody send me a step to step to do this configuration?
Thank's All
Everton Goularth
___
Yahoo! Mail - Sempre a melhor
I have created an asterisk.org blog entry:
http://www.asterisk.org/node/48358
to describe what I will shortly be committing to trunk to correct the
weaknesses of CDRs, that asterisk users and developers have been
complaining about for quite some time.
Highlights: Restructuring the code and
Don't forget that you might be able to write that:
exten = s,3,...
exten = s/8585979857,4,Goto(15)
exten = s/8585970327,4,Goto(15)
exten = s,4,Goto(5)
exten = s,5,...
The closest-matching priority 4 will be chosen, even if it's simply
well go on to priority 5 then
Moj
C. Chad Wallace wrote:
Well that rabbit trail paid off thanks David.
It does appear as if the GXP-2000 only supports one channel of G729. You can have many different lines on hold with G729, but the conference feature requires two channels be active and only one of them can be G729.
This is true even if you force
Hi Steve,
Please look at my asterisk-dev post from a few minutes ago about
dcontext and dst where the behaviour changed in a bad way in svn trunk
recently.
Thanks,
Steve
___
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asterisk-users
Steve Finkelstein wrote:
Hi all,
I have the following in my extensions.conf:
I use the mysql addon and create a subroutine that checks for black
listed numbers. I then call it at each entry point (For faxes as well):
; **
; Auto attendant
; **
exten =
Murf,
you crack me up, but I totally agree with the vote or don't complain
model.
Thanks,
Dave
On Tue, 2007-06-12 at 13:05 -0600, Steve Murphy wrote:
I have created an asterisk.org blog entry:
http://www.asterisk.org/node/48358
to describe what I will shortly be committing to trunk to
Hello,
Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone.
Did I miss something ?
Regards
___
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To UNSUBSCRIBE or update options visit:
Looking at the archives, it looks like MeetMe Realtime is in 1.4 but alas the documentation is lacking.
Is it as simple as add the SQL table and placing a meetme family in the extconfig.conf?
It also looks like Dan Austin at phoenix dot com was working on a scheduler for this. Any news on that?
On 6/12/07, Olivier [EMAIL PROTECTED] wrote:
Hello,
Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone.
Did I miss something ?
I don't know of any other GE phones.
However...
Why in the world would you ever need GigE sip phones?
-Erik
On 6/12/07, Steve Murphy [EMAIL PROTECTED] wrote:
I have created an asterisk.org blog entry:
http://www.asterisk.org/node/48358
to describe what I will shortly be committing to trunk to correct the
weaknesses of CDRs, that asterisk users and developers have been
complaining about for quite
Oliver
The thing you missed about Gigabit enabled SIP hardphones is the
demand for them.
Andrew
On 6/12/07, Olivier [EMAIL PROTECTED] wrote:
Hello,
Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone.
Did I miss something ?
Regards
___
Quoting Erik Anderson [EMAIL PROTECTED]:
On 6/12/07, Olivier [EMAIL PROTECTED] wrote:
Hello,
Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone.
Did I miss something ?
I don't know of any other GE phones.
However...
Why in the world would you ever need GigE sip phones?
unless
Erik Anderson wrote:
On 6/12/07, Olivier [EMAIL PROTECTED] wrote:
Hello,
Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone.
Did I miss something ?
I don't know of any other GE phones.
However...
Why in the world would you ever need GigE sip phones?
Maybe for people who
Hi. I am using latest asterisk 1.2 and it would be nice in a meetme
conference to be able to mute all but a particular user and then
unmute all those users again with one command. Am I missing something
or is this not available? Maybe I could write something, but I wanted
to check first.
--
1.4 does have support for MeetMe RealTime, and the docs are a tad
lacking.
I have a patch up on Mantis that extends/cleans up the RT features in
app_meetme
I made the column names configurable, with optionally enabled scheduling
features (start time, end time, maximum participant count per
Olivier wrote:
Has anyone tried latest HPEC software with Asterik 1.2 ?
It was published a couple of weeks ago to improve clipping issues.
It works just fine.
___
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asterisk-users mailing list
To
Andrew Latham wrote:
Oliver
The thing you missed about Gigabit enabled SIP hardphones is the
demand for them.
Not true. I can think of several places where I have or would like to
install phones where the end users currently have Gigabit ethernet feeds
to workstations. Specifically if you
I don't see this listed anywhere here in the replies so.
In your zapata.conf file try changing:
echocancelwhenbridged=no
to:
echocancelwhenbridged=yes
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To
Remember to restart asterisk and zaptel when you make this change.
On 6/12/07, Matt [EMAIL PROTECTED] wrote:
I don't see this listed anywhere here in the replies so.
In your zapata.conf file try changing:
echocancelwhenbridged=no
to:
echocancelwhenbridged=yes
On Tue, 2007-06-12 at 16:44 -0700,
[EMAIL PROTECTED] wrote:
Date: Tue, 12 Jun 2007 17:56:34 -0500
From: Darrick Hartman [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Gigabit SIP Phones
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
On Tue, Jun 12, 2007 at 07:44:02PM -0400, Matt wrote:
On 6/12/07, Matt [EMAIL PROTECTED] wrote:
I don't see this listed anywhere here in the replies so.
In your zapata.conf file try changing:
echocancelwhenbridged=no
to:
echocancelwhenbridged=yes
Remember to restart asterisk and
This should only be for TDM to TDM calls, SIP to SIP calls don't use the
zaptel driver.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, June 12, 2007 16:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I like the way people replied to this message of mine. It seems this thread is
going back to the hybrid echo issue(no this is not the problem). As said by
many ZAP is not in picture for SIP--SIP ie Ext-Ext internal calls.
To put my inputs I did tons of QA on this issue to ground on
What are the end devices? That seems to have been lost here. The real
issue is the handsets as those are the devices introducing the echo (the
only analog players here). Most likely a volume or gain issue on those
handsets, what SIP devices are the echo issues between? If both people
hear echo,
On 6/12/07, Olivier [EMAIL PROTECTED] wrote:
Hello,
Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone.
Did I miss something ?
I don't know of any other GE phones.
However...
Why in the world would you ever need GigE sip phones?
I think the advantage would be in the
Also, are there any IP phones that run apps other than telephony,
like
video, which could use more than 100Mb, even if just while switching
streams?
Video of 100Mb/s? ;-) HDTV doesn't consume more than 20Mb/s, Gige is an
overkill for IP Phone. Though it is used for switching, I assume
Deepak Naidu wrote:
I like the way people replied to this message of mine. It seems this thread is
going back to the hybrid echo issue(no this is not the problem). As said by
many ZAP is not in picture for SIP--SIP ie Ext-Ext internal calls.
To put my inputs I did tons of QA on this
84 matches
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