Hi the list,
I am using Kirk DECT/SIP 600V3 every day.
This system run very very well behind an Asterisk, with transfert feature,
caller ID display and so...
Seen as an IP-Phone running a separate SIP account for each handset.
Consider the 600V3 server as a mediagateway converting DECT to SIP.
I
Hi to everybody,
I need some help in configuration of the spa 3102.
I created an account for line 1 (user 208, sip port 5061) correctly
registered in asterisk, then i create an account
in sip.conf like this:
[general]
register = line01:pwdsipura:[EMAIL PROTECTED]:5060/095377078
[line01]
We are having some major problems with app_swift since we went live. It
is regularly segfaulting.
I don't know if this is my fault or not, but here's the story:
Installed the cepstral voices (at the time, 4.0) on our test system
(2.6.9-42.0.10.ELsmp)
and later added some extra voices (now
Alex Crow wrote:
Alban,
Thanks! Where on earth did you source this? I can't seen to find hide
nor hair of it here in the UK :(
Alex
On Mon, 2007-06-04 at 16:01 +0200, Alban wrote:
Hi,
I've tested several wifi phone (UtStarcom, Hitachi 3000 and 5000, and one
Siemens). The Siemens is the best
Thanks Chris,
I tried:
debug = debug,dtmf
console = notice,warning,error
;console = notice,warning,error,debug
messages = notice,warning,error
;full = notice,warning,error,debug,verbose
And restarted the logger, but I don't see any DTMF output in the debug
log file when I call into meetme
On Monday 04 June 2007, Thomas Stein wrote:
Hello.
Is it possible to dial in background 2 or more different numbers while the
same uninterrupted soundfile is playing? Something like this:
exten = Answer
exten = Playback (hello-bla-bla-we are trying to connect you-play-music)
exten = Dial
You can use the small 'm' option in the dial command like this:
exten=1,1,Dial(SIP/123SIP/456SIP/789,,m(default)) ;default is the MOH
class name
This will play music on hold. To play specific msg instead of MOH, you can
create a different MOH class where you can specify to play that specif msg
i didnt see you were alreadyusing the m option. sorry about that. you just
need to make a new MOH class which play the msg instead of music. you dont
have to use the local dial option.
On 6/5/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
You can use the small 'm' option in the dial command like
Hello, everyone
I'm looking for a way to find out if there is a device registered on a
particular extension from dialplan. Does anyone know how to do that?
Thanks in advance.
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asterisk-users
We have a similar problem at our place, since a few months.
oej, mentioned a patch he has made after the release of asterisk-1.4.4. So
we're
all desperately waiting for asterisk-1.4.5 to be released; unless you want
to install
from svn.
2007/6/4, Compnet Bobby [EMAIL PROTECTED]:
We have
On Tuesday 05 June 2007, Rizwan Hisham wrote:
i didnt see you were alreadyusing the m option. sorry about that. you just
need to make a new MOH class which play the msg instead of music. you dont
have to use the local dial option.
On 6/5/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
You can
Hi
I've two boxes connected over IAX2 trunk before IAX I was using SIP trunk
and they were working fine b'coz of bandwidth issue I changed from SIP to
IAX now I'm facing a strange problem after some time on the cli of my
asterisk box I see lots of messages of IAX2 trunk and b'coz of that my
Hi all,
I have a problem to park the call with #700 when the SIP phone and the
asterisk PBAX run on the same machine.
Call parking works well if the SIP phone that I use is on a remote pc in
comparison with the asterisk on the which it is registered. If the
sipphone is in the same pc where
Hi All,
I'm still having trouble trying to figure out if it is possible to define
(in the dial plan) a context for meetme?
I'm using 1.4.4 with dialplan logic of:
exten = 123,1,Meetme(,Msa,)
This defaults to conferences defined within the rooms context of meetme.conf
Is it
Hi,
Is it possible to dial in background 2 or more different numbers while the
same uninterrupted soundfile is playing?
Try to use asterisk queues. with queue you can play music on hold etc/
IVR to caller while trying to connect it to the available agent. you can
use your target number(s with
Hi everybody,
From asterisk CLI we can add extensions in dial-plan dynamically using
dialplan add extension command.
but how we can dynamically create a context in dialplan. is that
possible?
Nasir Iqbal
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On 04/06/07, Caio Zanolla [EMAIL PROTECTED] wrote:
Hi everyone,
in ldap realtime sip peers i need fullcontact set to
sip:[EMAIL PROTECTED] for asterisk to correctly match the peers (at least
for the natted peers to reach them)...
anyway, how do I populate fullcontact on the fly with
In article [EMAIL PROTECTED],
Chris Blunt [EMAIL PROTECTED] wrote:
I'm still having trouble trying to figure out if it is possible to define
(in the dial plan) a context for meetme?
Just have a look in apps/app_meetme.c in the functions find_conf() and
conf_exec(). You can see in there that
On Tuesday 05 June 2007 3:25 am, F6HQZ wrote:
I am using Kirk DECT/SIP 600V3 every day.
This system run very very well behind an Asterisk, with transfert feature,
caller ID display and so...
Seen as an IP-Phone running a separate SIP account for each handset.
Consider the 600V3 server as a
Hi everybody,
as far as i remember you don't need following line in configuration
register = line01:pwdsipura:[EMAIL PROTECTED]:5060/095377078
its your spa3102 who should register itself on asterisk. as i understood
the register = . is used when you need asterisk to behave like
Hi all,
I'm planning to buy a X100P clone and would like some feedback about
this card.
Does anyone already used this card? Does anyone recommend it ? or not?
I'd appreciate any comments.
Thanks.
Ronaldo.
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On 6/5/07, Ronaldo [EMAIL PROTECTED] wrote:
Hi all,
I'm planning to buy a X100P clone and would like some feedback about
this card.
Does anyone already used this card? Does anyone recommend it ? or not?
I'd appreciate any comments.
If you have a new 3.3v only motherboard then make very sure
Hi All!!
I have my asterisk working in my house (working with mandriva 2007 and
asterisk 1.4 svn). I´ve looking on the net but i couldn´t find the way of
making work my asterisk in a real enviroment. Seems that the problem of NAT
is a big problem. How can I sort out this, a mean crossing the
I've been using one for several years now. It works ok but not
spectacular.
You can almost forget ever sending or receiving faxes thru it tho. They
only work about 5% of the time.
Henry Cobb wrote:
On 6/5/07, Ronaldo [EMAIL PROTECTED] wrote:
Hi all,
I'm planning to buy a X100P clone and
Hello!
We have a strange problem, which doesnt let us receive faxes, because
during the initial fax whistling there are suddenly some background
beeps showing up, which then naturally disable any successfull fax
transfer, and we cant figure out which service or device (cisco,
asterisk)
On 6/5/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote:
Hi All!!
I have my asterisk working in my house (working with mandriva 2007 and
asterisk 1.4 svn). I´ve looking on the net but i couldn´t find the way of
making work my asterisk in a real enviroment. Seems that the problem of NAT
is a
Hi there,Do I need to do anything to optimize the use of asterisk on a
multi-processor x64 system before, during or after compilation?Responses would
be appreciated.Rgds,Akpome
_
With Windows Live Hotmail, you can personalize your
On 6/4/07, Jordan Novak [EMAIL PROTECTED] wrote:
I am having a difficult time with the transition from agentcallback
login...
Here are a few of the isssues, I am logging in using chan_ local
ie:local/8000 as the extension
I'm not sure if this will solve any of your problems or not, but
I would leave the CFLAGS etc as they are and just compile with your 64bit
GCC. That is what I have done and it works just fine.
On Tue, 5 Jun 2007 13:33:11 +, Akpome Akpoguma [EMAIL PROTECTED]
wrote:
Hi there,Do I need to do anything to optimize the use of asterisk on a
multi-processor x64
Hi to everybody,
I have an spa 3102 where i connected an analog phone (in the fxs port) and
the pstn line (in the fxo port).
This is my problem:
the incoming call doesn't arrive to asterisk.
In the spa web page i configured this dialplane:
(:[EMAIL PROTECTED]:5060)
where line01 is the
NAT is not that big of a problem, not anymore.
Do a NAT search on http://www.voip-info.org - it'll get you started (got me
started at least)
--
Cosmin Prund
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Henry Cobb
Sent: Tuesday,
On Mon, 2007-06-04 at 09:19 -0700, Douglas Garstang wrote:
Where’s Steve Murphy when you need him? J
I'm right here! :)
This doesn?t seem to work in AEL2?
Macro foo(arg1) { ?..
Includes {
Hangup;
}
}
The error is: File: /etc/asterisk/extensions.ael, Line 59, Cols: 5-12:
I just solved a similar problem on my asterisk box. i just enabled nat=yes
and removed the externip from the nat portion in sip.conf. Try it.
On 6/4/07, Compnet Bobby [EMAIL PROTECTED] wrote:
We have the latest version of asterisk, on a xeon dell server (2gb ram),
with 6 snom320's(latest
Hi Damiano!
Take a look at this link:
http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_faqid=5159lid=6862769263B11
Best regards;
Leonardo Kamache
On 6/5/07, damiano bertuna [EMAIL PROTECTED] wrote:
Hi to everybody,
I have an spa 3102 where i connected an analog
I am using Local channel instead of callback agents and it works not as
good as I expected. If you add /n option then after the transfer queue
assumes that agent is still busy because asterisk doesn't hangup such
channels after tranfer. If you don't use /n then queue doesn't have info
about
Hello,
I'm not 100% sure if it's the same on 1.2 as I'm on 1.4 now, but when
I need to debug DTMF I add the following:
full = notice,warning,error,debug,verbose,dtmf
Then do a logger reload from the console
You should then see the following if you do a logger show channels in
the console:
- Steve Totaro [EMAIL PROTECTED] wrote:
Hello,
I just did a fresh svn install of 1.4 trunk everything. Everything
compiles and installs just fine.
When I get to asterisk-addons, I cannot select chan_mobile in
menuselect.
Chan_mobile is not even an option in menuselect for asterisk
Most small/medium companies have a T1 for all their phone needs.
Internally there is a need for some analog lines.
* Fax Machine - FXS
* Security System (most ask/demand two lines) FXS
* Paging - FXO
* Dialup systems
I think he's asking why both T1 and FXS/FXO need to be on a single card.
I don't see any mention of you adjusting gains on the card/phones. Also,
what are you doing for echo cancellation? Can you post your zapata.conf
file?
-Original Message-
From: Stephen Bosch [mailto:[EMAIL PROTECTED]
Sent: Monday, June 04, 2007 10:01 PM
To: Asterisk Users Mailing List -
On 6/5/07, Noah Miller [EMAIL PROTECTED] wrote:
Most small/medium companies have a T1 for all their phone needs.
Internally there is a need for some analog lines.
* Fax Machine - FXS
* Security System (most ask/demand two lines) FXS
* Paging - FXO
* Dialup systems
I think he's
I think you're on the right track. You need to decide where to store
the CID-data mappings (files on disk, astdb, mysql, generated
on-the-fly) and come up with what the wave files are (text to speech?
selected from pre-made recordings?) I would do the brunt of the work
with a script instead
Oops, I meant to include in my prior note that you could of course
generate the wave file with flite or swift for text-to-speech
Matthew Pease wrote:
Hi all -
Not really sure where to post this question as I am just starting to
research this issue.
We want to allow users to dial into our did
Hi Arun -
I've two boxes connected over IAX2 trunk before IAX I was using SIP trunk
and they were working fine b'coz of bandwidth issue I changed from SIP to
IAX now I'm facing a strange problem after some time on the cli of my
asterisk box I see lots of messages of IAX2 trunk and b'coz of that
I must be not understanding your question very well, because it seems
like an easy answer :)
In the following Dial event, we have Source and Destination. Like Eric
said, Destination can contain multiple devices, so can't be trusted.
But Source should only contain one device Does it have
On Tue, 5 Jun 2007, Cosmin Prund wrote:
NAT is not that big of a problem, not anymore.
Do a NAT search on http://www.voip-info.org - it'll get you started (got me
started at least)
And read the archives of this list - I've posted about NAT recently too,
and it all just works for most cases.
On Jun 5, 2007, at 9:46 AM, Cosmin Prund wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Henry Cobb
Sent: Tuesday, June 05, 2007 4:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] NAT
On 6/5/07, Noah Miller [EMAIL PROTECTED] wrote:
Hi Arun -
I've two boxes connected over IAX2 trunk before IAX I was using SIP
trunk
and they were working fine b'coz of bandwidth issue I changed from SIP
to
IAX now I'm facing a strange problem after some time on the cli of my
asterisk box I
All,
Since I have now (at least partially) got my 7961G phones working
with Asterisk, I have temporarily moved on to try to get the expansion
modules working. There doesn't seem to be much in the way of
documentation here either. Does anyone have this combination working
(or any 79X1) here?
- Eric Lubow [EMAIL PROTECTED] wrote:
All,
Since I have now (at least partially) got my 7961G phones working
with Asterisk, I have temporarily moved on to try to get the
expansion
modules working. There doesn't seem to be much in the way of
documentation here either. Does anyone
Awesome. Thanks for the tips. Today I'm going to install asterisk
for the first time try to set up the SIP/ DID trunk. I signed up with
Voicepulse connect.
Matt
On 6/5/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
I think you're on the right track. You need to decide where
Have you tried something along the lines of:
System(swift blah blah blah -o blah.wav)
Playback(blah.wav)
It does have an inherent delay for the generation step but maybe swift
binary segfaults less? I've only used cepstral via swift binary, and it
has never segfaulted for me. My swift and
Hmmm..
It does seem to be there:
ubiphone*CLI logger show channels
Channel Type StatusConfiguration
--- ---
/var/log/asterisk/messages File Enabled- Warning Notice
Error
All,
I've a question on A*k hardware.
I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz)
with 512mb RAM.
I'm supporting 60 users (Cisco 7940s each + Xlite PCs).
Call loads are low, max of about 10 simultaneous SIP/IAX calls.
CPU for A*k rarely goes above 2% as I can tell.
Its
Hello users,
I've searched the archives for information on training our end users on
how to use hard/soft phones and voicemail, and Asterisk in general -- I
couldn't find much that wasn't about echo. I've looked at the Asterisk
Documentation Project as well, but I'm more interested in users, not
Hi,
I just got a Polycom 330 and, of course, I don't have the firmware and
sip.cfg files to provision it. Where can I get those?
Mike
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To UNSUBSCRIBE or update
Hi Mike,
I believe Polycom has directed resellers to supply firmware updates
directly to buyers. I'd recommend you speak with whomever you purchased
the phone from.
Best,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352)
Anyone know how to change the From field in Asterisk PBX voicemail/email to
some other info of my choosing?
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Well John let em castigate away. While they are hrrumping their way to
perfection. My phone is ringing from those wanting PBXs.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Monday, June 04, 2007 8:09 AM
To: Asterisk Users Mailing
It could also be that network congestion is causing the quality degradation.
Do you have quality of service configured on the LAN?
You mention that you are IP only; does that mean you are doing local
traffic only or are you connecting to the public network via your internet
connection. If you
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, June 05, 2007 1:13 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Where to find
For the first of your two topics, there used to be an Asterisk
Voicemail User Reference PDF by Jeffrey C. Ollie floating around on the
net, but the link seems to be dead now. You may still be able to find
the DocBook XML sources out there if you search for them.
In any case, I will send the HTML
On 6/5/07, Jim Suber [EMAIL PROTECTED] wrote:
Anyone know how to change the From field in Asterisk PBX voicemail/email to
some other info of my choosing?
If you look in the sample voicemail.conf file that comes with
Asterisk, you'll see that there's a setting called serveremail:
; Who the
On 6/5/07, Ronaldo [EMAIL PROTECTED] wrote:
I'm planning to buy a X100P clone and would like some feedback about
this card.
Most of the clone cards don't support far-end disconnect supervision,
so you'll have problems where Asterisk can't tell that the other party
has hung up the phone. You'd
On Tue, 5 Jun 2007, Jim Suber wrote:
Anyone know how to change the From field in Asterisk PBX voicemail/email to
some other info of my choosing?
Edit the voicemail.conf file.
[general]
serveremail=Mr. Voicemail
[EMAIL PROTECTED]
Gordon
___
Jonathan, can you send that to me also please?
thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Barratt
Sent: Tuesday, June 05, 2007 11:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users]
On Tue, 2007-06-05 at 12:26 -0500, Jason Parker wrote:
- Eric Lubow [EMAIL PROTECTED] wrote:
All,
Since I have now (at least partially) got my 7961G phones working
with Asterisk, I have temporarily moved on to try to get the
expansion
modules working. There doesn't seem to be
On Tue, 5 Jun 2007, Adrian Marsh wrote:
All,
I've a question on A*k hardware.
I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz)
with 512mb RAM.
I'm supporting 60 users (Cisco 7940s each + Xlite PCs).
Call loads are low, max of about 10 simultaneous SIP/IAX calls.
CPU for
At 11:57 AM -0800 2007/2/21, Brad Templeton wrote:
On Tue, Feb 20, 2007 at 12:08:15PM -0700, Natambu Obleton wrote:
Why not make it like DNS and have each provider have their lookups
deligated to a local server and then each ISP will run a caching
server that will use a serial number system
I have a feeling that many folks will want you to send that to them
(myself included). Might it be easier if you just post it somewhere and
send the URL to the list?
Eric
On Tue, 2007-06-05 at 13:11 -0700, Compnet Bobby wrote:
Jonathan, can you send that to me also please?
thanks
Yeah I've heard the same breaks in conversations myself. It simply goes
silent for a few seconds - making both parties say the usual sorry..
Missed that can you say again?...
Connection quality via remote SIP (outside our network via internet) can
be terrible (using GSM), though obviously theres
shadowym wrote:
I don't see any mention of you adjusting gains on the card/phones. Also,
what are you doing for echo cancellation? Can you post your zapata.conf
file?
I had actually tried to adjust the gains, but it actually seemed to make
the problem worse.
I turned echo cancellation off
Good call Eric, thanks for the suggestion.
Voicemail user guide now temporarily available at:
http://pbx.openface.ca/vmug.htm
Yours,
Jonathan Barratt
Openface Internet Inc.
514-315-3652
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Lubow
Sent:
Thanks for your answer, I've now investigated this further and it was
really easy, it was right in front of my eyes...
Using the java library from asterisk-java.org it was extreamly easy,
start monitoring on a new channel, check that to is ment for me and add
a property listener for
Hi all,
This may be a really stupid question, but, what preset global dialplan
variables can I use to determine the calling leg when using Dial()?
Say I have phones (SIP peers) originating calls out of the same context,
and I need to set the ANI differently depending on who is calling out in
I installed a hardware g729 codec card in my asterisk, and I'm getting the
following error when calling from a g729 sip extension to a SIP trunk also set
to g729. The call goes through just fine, but these error messages keep flying
by until I disconnect the call.
Any ideas?
Hi,
Has anyone on this list connected with Verizon's SIP product? We are
currently undergoing interop testing with Verizon, and honestly, it seems
like the most convoluted process. I'd be interested in talking with
someone else who has gone through this and run a few things past you.
Why would calls be coming in on the Guest IAX account?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett
Sent: Monday, June 04, 2007 6:56 PM
To: 'Asterisk Users Mailing List -
Fuermann, Jason Bryce wrote:
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html
This only works if you have a reseller account.
-Stephen-
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asterisk-users
If I understand your problem correctly you need to set ANI/CALLERID on a
peer by peer basis.
You can use the accountcode variable in the sip.conf file and set that
to the DID or you can use another variable.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
On Tue, 5 Jun 2007, Alexander Lopez wrote:
If I understand your problem correctly you need to set ANI/CALLERID on a
peer by peer basis.
You can use the accountcode variable in the sip.conf file and set that
to the DID or you can use another variable.
Thank you for that suggestion.
But
Dear All,
I looking to implement asterisk solution for 2000 sip registrations and
expecting con-current call about 250.
Can some one provide me guide line that what kind of server will fullfil the
requirment.
what is the Processor, RAM ???
--
Thanks Regards,
Vidura Senadeera,
Network
You can use an ex girlfriend type extension, aka if your internal DID is
1000 and that is their caller id on your outbound macro or prefix doctor the
callerid
exten = 1000/_9XXX,1,SetCallerID(20843471000)
;)
Brian
On 6/5/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Tue, 5 Jun 2007,
The bar is getting raised yet again
http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack
et8+Virtual+Office.aspx
I personally use Snapanumber $30 or there abouts (after trialing a few
other TAPI solutions and finding them sub-par) and think it's a great
product but
Stephen Bosch wrote:
Stephen Bosch wrote:
Stephen Bosch wrote:
Hi:
I have a Sangoma A200 card installed in a server with two FXO modules
and one FXS module.
Analog sets connected to the FXS module have a squeaky static -- it's
like static mixed with the sound of someone vigorously
Every Sangoma A200 card I have ever connected to the PSTN required a rx gain
of at least 10. Yours is commented out which I believe would make it
default to 0? I am guessing that because the rx gain is so low the users
are cranking up the phone volume all the way and maybe your hearing
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