Hello,
my name is dominik, and i'm using asterisk with
voip without isdn, only sip.
I'm using Asterisk Version 1.0.7 on Debian
3.0.
I've configured the fax receive in the
/etc/asterisk/extensions.conf:
exten =
99,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten =
On 9 Sep 2006, at 00:42, Kevin Smith wrote:
Hi everyone,
I am looking to log CDR records to our MSSQL database for further
examination on the records. From what I gathered from the wiki I
have to choose between FreeTDS and unixODBC. Is there a better
choice? Which option would be better
Mike wrote:
Here it is:
dialplan dialplan.impossibleMatchHandling=1
dialplan.removeEndOfDial=1
digitmap dialplan.digitmap=[7]xx|[9]xxT|[9][1]xxT
dialplan.digitmap.timeOut=3/
When I dial 845, I get fast busy. When I dial 9-555-555-, it dials
without the need to
Hi
On Fri, Sep 08, 2006 at 04:54:32PM -0500, Iván Vega R. wrote:
Hi everyone,
I'm new on Asterisk.
One help item: please post messages with a descriptive subject line.
Something like:
problem configuring zaptel
or:
ZT_CHANCONFIG failed on channel 4: Invalid argument (22)
I'm
I’m currently trying to write a section into my dialplan that when a user
dials *78, it will beep 3 times, then wait 10 seconds for the user to enter
a 10 digit phone number, then beep 3 more times and put that number into my
AsteriskDB. I’m very new to this and I know this is probably very
Hello to all asterisk users,
I have a problem with call forwarding.
My extensions.conf:
[outbound]
exten = _*22*XXX,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten = _*22*,1,DBdel(CFIM/${CALLERID(num)})
Have three stations, 301, 302 and 303. When dial on 301 following
number:
*22*302
it
Same problem here on CentOS 4.4 :(
Strange that apparently the tarball was not tested if it would even
compile
On Fri, 8 Sep 2006, Stuart Sheldon wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I can confirm the same problem, it looks like the oct612x directory tree
is missing from
lol
Matt Riddell (IT) wrote:
Arun Kumar wrote:
hi
thanks for reply.
I'm using vicidial to make calls at 2.0 dial level it is able to make calls
but when I see the asterisk -r most of the time it shows Outgoing Spool
Failed. Which Spool File ?
Er, probably the best place to ask would be
Remco Barendse wrote:
Same problem here on CentOS 4.4 :(
Strange that apparently the tarball was not tested if it would even
compile
I just tried to apply the 1.2.9 patch to 1.2.8 and that fails to patch.
Looks like someone had a bad day
--
Bill Maidment
Maidment Enterprises Pty Ltd
hello ,
Which channel do you want to set chan_h323 chan_oh323
or chan_ooh323 ?
Harry
--- Wasif [EMAIL PROTECTED] a écrit :
Hello,
Could anyone tell me how to install/configure H323
with Asterisk 1.2.11 .
Thanks
Wazb
___
--Bandwidth
http://www.eflo.net/VICIDIALforum
and VICIDIAL does not use call files for Originate spooling. It uses
the Manager API.
MATT---
On 9/8/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Arun Kumar wrote:
hi
thanks for reply.
I'm using vicidial
Thanks all. It works fine now.
-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after
Actually, as soon as you hit 8 you will get the fast busy.
Is that your full dialplan? What about an emergency (911) or
other N11 calls? What about direct dial international calls (011...)?
Its my full test dialplan for now. I do get fast busy as soon as I hit 8,
so that part works.
Hi
all,
That's my last one
for a while (I hope).
How can I (if at all
possible) make the 501 turn on the speaker phone as soon as a digit is dialed
(if the handset is not lifted)?Sort of likewhat a normal
speakerphone does.
The reason I want this is I want the 501 digitmap to be taken
Hi Ivan. As you see in this page:
http://www.neobits.com/do/dtls?pid=9583
This card is a bundle, wich means supports both, FXO, and FXS. FXO
ports should be used to connect your 3 telco lines, and FXS port to
connect some phones.
Regards
On 9/8/06, Iván Vega R. [EMAIL PROTECTED] wrote:
Hi
Yeah I figured after some experimentation :) Thanks.
I got confused while configuring zaptel and asterisk, hehe.
On 9/9/06, Moises Silva [EMAIL PROTECTED] wrote:
Hi Ivan. As you see in this page:
http://www.neobits.com/do/dtls?pid=9583
This card is a bundle, wich means supports both, FXO,
Hello,
I have found and read Steven Critchfields writeup on how to use ZapRas
(Thanks Critch!), however I am a bit confused.
His write up is here:
http://copilotconsulting.com/mail-archives/asterisk.2003/msg01030.html
Currently we have a full PRI (23B Channels, 1D) coming into our
Asterisk Box
Does anyone know of a DID provider in Thailand?
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Thanks Tim,
That was my first thought as well but then I thought, might as well give
it a try. But it is turning into a hassle more then anything. I already
have a PHP script wrote to for MySQL so the conversion to MSSQL
shouldn't be bad.
Thanks,
Kevin
Tim Panton wrote:
On 9 Sep 2006,
I have a follow up question. How do I pass on the caller ID of the call I'm forwarding to the other party? I can pass on the channels caller ID but prefer to pass on the forwarding party's number instead.
-- Original message -- From: [EMAIL PROTECTED]
Thanks all. It
Hello Matthew,
It depends on the chipset on the mainboard. I had problems with a
SC1420, the only way to solve it was to get a new server (without
Intel chipset). So don't try a chipset which is listed on the Digium
compatibility site.
Wednesday, September 6, 2006, 8:55:58 AM, you wrote:
If you don't set the callerID in the channel, it will get passed on as-is.
Don't change it, and it will stay the same.
-TIm
On September 9, 2006 12:27, [EMAIL PROTECTED] wrote:
I have a follow up question. How do I pass on the caller ID of the call I'm
forwarding to the other party? I can
Hi,I was testing the intel based G729 codec on SVN-trunk-r42453 following the new instructions for compiling with SVN trunk and it in preliminary tests it works ok for some calls but I found when one end of the call is an IVR or Music On Hold the sound gets all distorted and asterisk segfaults.
I tried both of them but it still goes asID unavailable. First I commented it out, that did not work and left it blank and that did not work either. Below is the sample in sip.conf
[4305]type=frienduser=4305secret=xxx;context=from-sipcallerid= ; left it blank but did not get passed
In
case you use an adapter or voip phone: Did you try to press hash # after the
number ? - thenthe adapter/voip phonedials immediately and doesnt
wait for the next digit timeout.
Cheers
Gerry
-Original MessageFrom:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
Jason Lee wrote:
Hi,
I was testing the intel based G729 codec on SVN-trunk-r42453 following
the
new instructions for compiling with SVN trunk and it in preliminary
tests it
works ok for some calls but I found when one end of the call is an IVR or
Music On Hold the sound gets all distorted
Yes that works. I'm using Linksys adapter, is there a code I can put in the dial plan to prevent users from putting # after the number? I have a lot of people on the server and cannot ask them all to be pushing # after every call. Thanks for the tip and any help will be appreciated.
Yes you could script a dialplan putting ... and S0 (zero) at the end.
An example :
(xxS0) It will dial 6 digits directly when you enter the 6th.
You could learn how to adapt your Linksys dialplan looking this wiki.
http://voip.wikispaces.com/
[EMAIL PROTECTED] escribió:
Yes that
I recompiled with debuging options...both bt and btfull outputs http://pastebin.ca/165250Before I recompiled it gave me a second of audio then I got nothing but distortion for 5 seconds then asterisk would crash.
I retested after compiling it with just a call between two local devices one using
Mike wrote:
Did I misread the Asterisk wiki pages, because I believed that when a
pattern was present, the pattern takes precedence over any real
extensions? (i.e. if I have both 1234 and _1XXX as extensions in a context)?
It's the opposite. Asterisk always uses the most specific match for an
Jason Lee wrote:
I recompiled with debuging options...
both bt and btfull outputs http://pastebin.ca/165250
Before I recompiled it gave me a second of audio then I got nothing but
distortion for 5 seconds then asterisk would crash.
I retested after compiling it with just a call between two
Sorry about that. I thought I had the right core dump. I retried again and the output from bt and bt full is at http://pastebin.ca/165289It took 1min 50seconds of nothing but distortion before asterisk segfaulted
-- Regards,JasonOn 9/9/06, Daniel Pocock [EMAIL PROTECTED] wrote:
Jason Lee wrote: I
It certainly makes sense, and I tried it...it works, you are right.
So what do you make of this page :
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf
+sorting
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
John
Hi Mike,
As far as I know, you need to at least start the dialing (ie New call,
speaker, etc) for the digitmap to even come into play.
The only settings that I am aware of that you can try to change are
dialplan.impossibleMatch-Handling and dialplan.digitmap from sip.conf.
Kevin
Mike
] Call Forwarding in SIP.conf
Date: Sat, 9 Sep 2006 16:52:40 +
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--
Message: 2
Date: Sat, 9 Sep 2006 19:17:23 +0300
From: G.Jacobsen [EMAIL PROTECTED
Thanks, I tried that and did not work for me. My users are calling US number and without the # at the end of the last digit dials it takes 11 seconds before it starts ringing.
-- Original message -- From: Alberto Sagredo [EMAIL PROTECTED] Yes you could script a dialplan
--- Bill Maidment [EMAIL PROTECTED] wrote:
Hi
I've just tried to compile the zaptel-1.2.9 release and I get the
following error:
Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors when
compiling zap:
make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command
HiI have successfully been running with several Polycom SoundPoint 501phones and recently purchased some Polycom 301 phones.However, I can't seem to get the phones to register. The phone seesthe asterisk server, but all calls our are busy.
The only difference for 'sip show peer xxx' for a working
The work around is at:
http://www.sineapps.com/news.php?rssid=1496
On 09/09/06, Samy Antoun [EMAIL PROTECTED] wrote:
--- Bill Maidment [EMAIL PROTECTED] wrote:
Hi
I've just tried to compile the zaptel-1.2.9 release and I get the
following error:
Same here, using CentOS 4.4 kernel
On a new set up Centos 4.4, kernel 2.6.9-42.0.2.EL, yum updated, 2
BRI-HFC cards, no digium hardware.
modprobe zaptel and modprobe ztdummy are both in rc.local, and lsmod gives:
[EMAIL PROTECTED] ~]# lsmod
Module Size Used by
ztdummy 3924 0
zaptel
zap show status
will tell you if Asterisk is really using ztdummy
Make sure you have chan_zap.so enabled in modules.conf (or that it isn't
disabled with a noload declaration)
Nigel Godfrey wrote:
On a new set up Centos 4.4, kernel 2.6.9-42.0.2.EL, yum updated, 2
BRI-HFC cards, no digium
--- Nigel Godfrey [EMAIL PROTECTED] wrote:
The work around is at:
http://www.sineapps.com/news.php?rssid=1496
Thanks, I'll give it a try.
__
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Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Hi all,
I am trying to understand contexts a bit better. The problem I have is
when you know when a context is finished. Is this when a new context
starts?
Example:
[context1]
exten = _9170X,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[context2]
exten = 6394,1,Dial(Local/6275/n)
Mike wrote:
It certainly makes sense, and I tried it...it works, you are right.
So what do you make of this page :
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf
+sorting
Interesting. I got my information from Asterisk: The Future of
Telephony (in the dialplan
Rene wrote:
Hi all,
I am trying to understand contexts a bit better. The problem I have is
This should help:
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/c650.html
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
Yep, that should help, and the short answer to you question is NO.
Regards
On 9/9/06, Doug Lytle [EMAIL PROTECTED] wrote:
Rene wrote:
Hi all,
I am trying to understand contexts a bit better. The problem I have is
This should help:
Thanks Daniel, your advice helped. It was the call waiting not working, which I thoughtwas working because I had selected 'No' beside 'Disable Call-Waiting'. But for call waiting to work properly, I also needed to select 'No' beside '
Enable Call Features' and then dial *70 to enable it on
I figured out how to send the caller ID, it is working. But now I am trying to send the Company Name along with caller ID, and that is not working. What could be the reason for that. I am using SerCallerID. Is there something else I should use to send the alphabets?
I am still trying to make it work. Where did you get firmware version 1.1.1.9. On there website they have only 1.1.0.16
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I'll try to use it, but in future I think I'll get some other phones with better configuration utilities.
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I sent the incoming calls to different queues, and now everything is working fine. Just changed the MoH option of the first queue to ring tone, so that the callers hears the ring going, otherwise they'll feel uncomfortable why music has started all of a sudden.
In my queues, I wanted callers to listen dial tone going, instead of listening the music, to I used option 'r' in queue command. Now it doesn't announce the position of caller and estimated hold time etc. Is this normal for this setting or I am soing something wrong. If I don't use option r and
I am having hard time with grandstream phones for a30 phone setup. When a change in configuration is required, I have to change their configurations manually for almost all of them. Their configuration utility is not very straight forward to use.
For my next installation, I would prefer some
On 18:47, Sat 09 Sep 06, Zeeshan Zakaria wrote:
In my queues, I wanted callers to listen dial tone going, instead of
listening the music, to I used option 'r' in queue command. Now it doesn't
announce the position of caller and estimated hold time etc. Is this normal
for this setting or I am
Try the Linksys ATA. I gave up on Granstream and have 4 sitting in around.
-- Original message -- From: "Zeeshan Zakaria" [EMAIL PROTECTED]
I am having hard time with grandstream phones for a30 phone setup. When a change in configuration is required, I have to change
[EMAIL PROTECTED] wrote:
Thanks, I tried that and did not work for me. My users are calling US
number and without the # at the end of the last digit dials it takes
11 seconds before it starts ringing.
If you are dialing 11 digits then set the Linksys Dial Plan to:
(1xx), the phone
Zeeshan Zakaria wrote:
I figured out how to send the caller ID, it is working. But now I am
trying to send the Company Name along with caller ID, and that is not
working. What could be the reason for that. I am using SerCallerID. Is
there something else I should use to send the
Most phone
Before the phone starts it boot process, I edit the phone settings under server
information, enter the IP of the sip server and also the protocol to use, UDP
Only.
Chris
David Gagnon wrote:
Hi Chris,
I'm would like to get more information about this problem. Why the
phone
Polycom & Aastra are both great in this manner.
Michael
--Original Message Text---
From: [EMAIL PROTECTED]
Date: Sun, 10 Sep 2006 00:18:37 +
Try the Linksys ATA. I gave up on Granstream and have 4 sitting in around.
-- Original message --
From:
I'm trying to get a clear understanding just how calls are routed in a
mixed SPA3k and Asterisk system.
This is my present (incomplete) understand and I'd appreciate any
corrections. I'm especially interested in what happens between Asterisk
and an SPA3k.
Note:
-
POTSaudio refers to
Can you explain a little bit what make them better for mass deployment. Do they have Windows based software to communicate with all the installed phones and upgrade them and also to remotely monitor them. Is there a separate cost for these software tools or are they free?
I finally made the paging to work. But the only thing which I had to change was the number to dial. As in the instructions, it is _**1, but it didn't work for me and I used 333 instead. All other settings are the same. Now the reception phone's one Speed Dial key is assigned ext 333, pressing
Nigel Godfrey wrote:
The work around is at:
http://www.sineapps.com/news.php?rssid=1496
Thank you. I'd forgotten about SVN. Works like a charm.
Cheers
Bill
--
Bill Maidment
Maidment Enterprises Pty Ltd
www.maidment.com.au
si hoc non legere potes tu asinus es
for mass deployment the Linksys will allow you to update your routers with a tftp server.. You can have the routers always download their software from the tftp server, that way you have the latest on the server for upgrade software. The reason that I don't like granstream is their bad customer
Nope. A context ends when a new one starts. The only way for a call to
continue is to have a maching extension, and the next higher priority. If
you want a call to continue in another context, you need to use the Goto()
application.
On September 9, 2006 19:04, Rene wrote:
Hi all,
I am
I am a really big fan of Aastra phones.
It's a splinter company of Northern Telecom, so their quality is very good.
Provisioning is done via a text file on either a tftp or an ftp server. There
is a global file and a per phone file. When you have a good set of config
files built, you can
I'll speak on the Aastra, since that is what I know, although most of this
applies to Polycom as well.
There is no windows software needed at all. Personally, I haven't been a
Microsoft customer in more than half a decade. Their operating systems are
not appropriate for telecom applications.
Thanks for the info. In my next installation, I'll try those phones.
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On Saturday 09 September 2006 18:51, Zeeshan Zakaria wrote:
For my next installation, I would prefer some other phones with better
configuration and remote accress utility. My question to those of you with
more experience, what IP phones are better for mass deployment and easy
management of
On my grandstream phones, I am having problem that I can't login in remotely. With in the local network, they work ok, but outside the network, they don't. All the NAT settings are ok and I can log into the Asterisk server from X-Ten, no problem. But Grandstream phone doesn't login. I've port
I've noticed that Grandstream works better using stun and not port forwarding your router.
Try setting stun.xten.com or stun.fwdnet.net in your GS2000 and make sure sip.conf has nat=yes. It should work fine.
Also, i've noticed that Linksys wireless with speed booster has something in it that is
I am trying some streaming radio for the frist time, but it is not working. I have something like this in my musiconhold.conf:
stream = quietmp3:/var/lib/asterisk/mohmp3/stream,
Actualy it doesn't realy have to do with the phone company not letting
you set it, although sometimes between 2 PRIs what you set as the name
might come up. It has to do with the fact that when phone companies
send callerid to a ringing phone they don't look at what you set for
name, they look up
. Pierre [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Call Forwarding in SIP.conf
Date: Sat, 9 Sep 2006 16:52:40 +
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