Does 'show features' display the correct information?
PaulH
Thanks Paul
*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #
Attended Transfer
One Touch Monitor
On our system i got:
Zap/1-1 answered SIP/106-091a2750
-- User hit '*1' to record call. filename: wav|
auto-1205385048-106-0434225491|m
Our dialplan looks like:
_0X' = 1. Dial(zap/g1/${EXTEN}||Ww)
(from show dialplan)
PaulH
Thanks Paul.
I think the problem is *1 is
Thanks Atis
I'll give that a shot and see where I end up.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins
Sent: 12 March 2008 06:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call tracing -
I think the problem is *1 is being ignored or cannot be transmitted
successfully to Asterisk.
Finally I resolved the problem.
For some reasons, the * and 1 must be pressed pretty quickly
together on the Polycom phone before it can be transmitted successfully
to Asterisk.
I think I cannot deny
On Tuesday 04 March 2008 06:48:43 am marek cervenka wrote:
hi,
can you recommend cleansimplestable solution for incoming call popup
(in browser)?
i'm using flash operator panel now
but i want something without flash (maybe something in AJAX?)
thanks
My guess (from your features) is that the * for disconnect and *1 for
records are clashing - maybe set disconnect to **73 to avoid this.
And - yes, it can be tuned:
;featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation (default
Just chatted to one of the guys here - he said that if you set it to a
single DTMF digit, it can be pretty good...anything more, not so
reliable.
PaulH
On Thu, 2008-03-13 at 17:22 +1100, Lee, John (Sydney) wrote:
I think the problem is *1 is being ignored or cannot be transmitted
I think if you install sox you will get soxmix.
Thanks Paul
I did yum install sox and Asterisk will automatically combine .in.wav
and .out.wav together into .wav
That is excellent!
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On 18:12, Wed 12 Mar 08, Joshua Wilson wrote:
How did you like the gui interface?
For a standalone PBX it's the best gui I've seen so far.
On 3/12/08, Michiel van Baak [EMAIL PROTECTED] wrote:
On 15:32, Wed 12 Mar 08, Joshua Wilson wrote:
I don't believe it supports multi-tenant as of
On Thu, Mar 13, 2008 at 3:47 AM, Gonzalo Servat [EMAIL PROTECTED] wrote:
I can't find any channel variable that gives me this info.
Gonzalo,
With SIP callers you can get the address from the SIPURI channel variable.
IAX does not seem to have an equivalent var... The best I could find is
Improvement: also check the funcions SIPCHANINFO and IAXPEER... With
this and the SIPURI channel variable you should be able to have all the info.
Cheers,
--
exvito
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Hi,
This is a bit OT. If I have a phone behind NAT and the phone registers via an
SBC can I set NAT=NO and canreinvite=yes in asterisk or will I still have
issues ?
Thanks.
___
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- Original Message -
From: John Lee (Sydney) [EMAIL PROTECTED]
Does anyone know if that can be tuned?
Sure... go to features.conf, and change the value of the featuredigittimeout
option.
---
Jared Smith
Community Relations Manager
Digium, Inc.
Thanks for the help. I still had a misconfiguration in my res_odbc.conf, but I
figured it out and it appears my voicemail storage is working. I haven't had a
chance to get to the phone on the extension I'm using for it.
--
Mike Hammett
Intelligent Computing Solutions
A note to all:
http://www.testforme.com/download/polycom/
Polycom has apparently changed their tune. You can now download all
the latest firmware and bootrom releases directly from their website
(even sip 2.2.2 and bootrom 4.0).
- Noah
-Original Message-
From: [EMAIL
More information:
Wireshark running on the * box shows that the request is being received, but
rejected:
Request: INVITE sip:2247@(ipaddress of *):5060, with session description
Status: 407 Proxy Authentication Required
Request: ACK sip:2247@(ipaddress of *):5060
Did I miss a security setting?
First of all, if Asterisk is the client and it must register to the other side,
does the peer\user entry have to be in sip.conf, or can it be in ARA?
Second, why do all calls fall through to the last context specified, whether in
that peer\user definition or not? I'm assuming it's a typo
Hi,
Surely, I must be overlooking something. If I run the
following SQL queries I don't get the same number of
rows. Is this coherent?
mysql select * from queue_log where queuename =
'4010' and FROM_UNIXTIME(time) between 2008030800
and 20080313145900 group by callid;
357 rows in set (0.01
On 3/13/08, Vieri [EMAIL PROTECTED] wrote:
Hi,
Surely, I must be overlooking something. If I run the
following SQL queries I don't get the same number of
rows. Is this coherent?
mysql select * from queue_log where queuename =
'4010' and FROM_UNIXTIME(time) between 2008030800
and
Some light relief
SNOM say Please note that you will not be able to reach us by phone.
http://www.theregister.co.uk/2008/03/13/dont_call_us/
regards,
Drew
--
Drew Gibson
Systems Administrator
OANDA Corporation
www.oanda.com
___
--
Yes it is.
The reason you get more entries in queue_log is that there are several
queue_log events per call - most commonly you get an ENTERQUEUE,
CONNECT and COMPLETECALLER/AGENT for each call.
Vieri wrote:
Hi,
Surely, I must be overlooking something. If I run the
following SQL queries I
Hi:
yes. that is very interesting. i hope they can resolve that and
back to work soon.
i have many technical issues need their support.
i have heard many positive reports about snom. in my testing
experience, their product seems a little better than grandstream. snom
has more functions and
Matt Riddell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tzafrir Cohen wrote:
Let's be more specific here, folks:
What version numbers?
Asterisk, spandsp, agx-addons / rx-tx-fax?
Asterisk: yesterday's 1.4 SVN
SpanDSP: tried with pre 15, 16 and 18
AGX-Addons: tried
On Mon, Mar 10, 2008 at 02:59:04AM -0500, John Faubion wrote:
But, just to clarify, please remember that using music as MoH
is considered a public performance, and if the pieces in
question do not include a buyout license *for the performance
Ok now I am curious, if a radio is playing
On Mon, Mar 10, 2008 at 03:55:52AM -0500, KodaK wrote:
On the subject of hold music, I've been using stuff from stock20.com.
I'm very impressed with their license contract; it's the clearest I've
seen.
I'm not sure how they get ASCAP and BMI, having been granted any rights
to collect on their
Updated with a smaller sip.conf that also doesn't work right.
[EMAIL PROTECTED] asterisk]# cat sip.conf
[general]
port=5060
canreinvite=no
rtcachefriends=yes
disallow=all
allow=ulaw
allow=alaw
register = 8157879826::[EMAIL PROTECTED] ; ottos 815-787-9826
register = 8159092443::[EMAIL
--- Ex Vito [EMAIL PROTECTED] wrote:
...as long as the destination does not answer
you'll get
a NO ANSWER disposition.
So, if in your case you want to know if a user
answered
the phone, then, yes, you will have to add the
DIALSTATUS
value to the CDR, probably in the CDR's
Hi,
Unfortunately that is true for the time being. Since we moved our main
office to new premises, our telecom provider has failed setup services
in time. Forums and otrs is online and we hope to have the phones
working ASAP.
We appreciate your understanding.
Regards,
Usman.
pls kindly respond to this email address
thx !
_
Connect and share in new ways with Windows Live.
http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008
___
--
Using Asterisk 1.2, still.
We are issuing a callback. User rejects the first two calls, but answers the
third. For some reason, the Manager Interface outputs a CDR with disposition
'NO ANSWER' for all three attempts, eventhough the 3rd call worked.
Is this an asterisk 1.2 bug?
Doug.
Hello,
This can help: http://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas van
dem Helge
Sent:
Heres a weird one...
Call comes in on mISDN channel. Little bit of dialcode (in a macro) looks
up the number in the astdb and puts an name to it. No real magic there,
and it works well.
Same macro also has parameter passed in to put a prefix on the name - this
is set in the DDI handling and
Hi, I have implemented a custom application module based in some esqueletone
code I will provide below. I have tested it with asterisk 1.2.23 and it works
fine. But when I tested the same application with a newest version of asterisk
like 1.4.* it always returns an error trying to load the
Gordon Henderson wrote:
Then no amount of Set(CALLERID(name)=somethin) will work. Even if I
explicitly do a Set before the dial, it seems to get ignored.
Trying doing that while using:
SetCallerPres(allowed)
Within your dial plan
Doug
--
Ben Franklin quote:
Those who would give up
On Thu, 13 Mar 2008, Doug Lytle wrote:
Gordon Henderson wrote:
Then no amount of Set(CALLERID(name)=somethin) will work. Even if I
explicitly do a Set before the dial, it seems to get ignored.
Trying doing that while using:
SetCallerPres(allowed)
Within your dial plan
Well there you go.
Michiel van Baak [EMAIL PROTECTED] writes:
On 15:32, Wed 12 Mar 08, Joshua Wilson wrote:
I don't believe it supports multi-tenant as of yet. It could be requested I
am sure.
I created a new VM and installed it.
Guess what, no multi tenant support.
Too bad all them good GUI tools never
That is something that would have to be tested and tried. I haven't ever
needed to run multi-tenant so it has never been a concern for me.
On 3/13/08, Benny Amorsen [EMAIL PROTECTED] wrote:
Michiel van Baak [EMAIL PROTECTED] writes:
On 15:32, Wed 12 Mar 08, Joshua Wilson wrote:
I don't
I've been asked to look at deploying Asterisk in a high availability
environment and I've been looking so I've been searching for methods to
decouple the voice PRI circuits from the Asterisk server so failover to
another server could take place.
I've been looking at the RedFone foneBRIDGE2 2e1
We used it in our installation and had some issues. We were passing fax and
modem calls through via the second port as a TDM bridged call. For some reason,
the timing was off even though we explicitly set the timing in the redfone.conf
file. We replaced it with a Sangoma A102d and haven't been
jonas boering wrote:
Have the registration way that the applications are registered in arterisk
1.2.* changed to much from version 1.4.* and 1.6.*?
Yes, they are almost completely different. You would be best off to
compare the source code of a simple application between 1.2 and 1.4 to
learn
Also to add to the last post does this device have hardware echo
cancelation?
if it does it could be a great replacement, if not may not be what I'd
really want to use.
Thanks,
tom
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of arkda
Sent: Thursday, March 13, 2008
For high-availability Asterisk when using PRI Lines, you might also want
to check out our product (FSV-4PFS). It's available at
www.failsafevoip.com
Bill
Also to add to the last post does this device have hardware echo cancelation?
if it does it could be a great replacement, if not may not be
Hi,
I have polycom 301 IP phones most of them especially when I call a
direct line with extensions, I cannot dial an extension. It does not
recognize my key inputs. If the number is an 800 number it seems to work
fine. I have used dtmfmode=inband with my sip trunks and my extensions
as
On Thu, Mar 13, 2008 at 03:32:08PM -0500, [EMAIL PROTECTED] wrote:
For high-availability Asterisk when using PRI Lines, you might also want
to check out our product (FSV-4PFS).
Which is something not unlike the Junghanns ISDNGuard.
--
Tzafrir Cohen
icq#16849755
I need to setup a small mail server on a local network. It only needs SMTP
ability as it's just so Asterisk can send out emails. The machine has sendmail
installed. My primary mail server seems to be rejecting the messages. Some
research says something isn't configured properly. What do I
We are in the process of building out www.dialaway4free.com, a free world
wide calling service. I am writing RFQ's for hardware, since we are going to
use asterisk as our call processor. I was wondering what is the best server
platform to use that will support digium cards and handle sip
Mike Hammett wrote:
I need to setup a small mail server on a local network. It only needs
SMTP ability as it's just so Asterisk can send out emails. The machine
has sendmail installed. My primary mail server seems to be rejecting
the messages. Some research says something isn't
On Thu, Mar 13, 2008 at 4:04 PM, Mike Hammett [EMAIL PROTECTED] wrote:
I need to setup a small mail server on a local network. It only needs SMTP
ability as it's just so Asterisk can send out emails. The machine has
sendmail installed. My primary mail server seems to be rejecting the
Mike Hammett wrote:
I need to setup a small mail server on a local network. It only needs
SMTP ability as it's just so Asterisk can send out emails. The machine
has sendmail installed. My primary mail server seems to be rejecting
the messages. Some research says something isn't
Through help from people on the lists and then further investigation based on
those results, here is what I did.
1) I set the office to a statically assigned IP instead of from the pool.
2) I made an A entry on one of my domains aiur.ics-il.net (where aiur is the
machine name).
3) I added
I am the ISP. ;-)
I'll have to look into that smarthost deal as there is no reverse DNS at
this time (my upstream's server times out).
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Erik Anderson [EMAIL PROTECTED]
To:
I was going to setup my extension on my employee's phone so he could answer
calls as well as myself. I noticed that once he registered, I could no longer
receive calls on my own phone. Is this a limitation of Realtime or something
else in Asterisk? I've had multiple devices register to the
As far as i know you cant have two devices register to the same
extension. We use -1 XXX-2 and so on if their is a need for the same
number to be on multiple phones. Now if you just want the calls to ring
his phone then all you have to do is just add his extension to the Dial
statement in
On Thu, Mar 13, 2008 at 9:07 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Which is something not unlike the Junghanns ISDNGuard.
...or the beroNet bero*fos
(https://shop.beronet.com/product_info.php/cPath/56/products_id/159)
Not affiliated, just a satisfied customer.
--
exvito
Has someone submitted a bugreport regarding enabling 9600kbps fax? I
always wonder why it would never negociate 14400kbps... when it did
work a single page on fine resolution would take 4 minutes.
Thank you very much for that link. I knew there had to be more
possible configurations for T.38. I
Below is the version I use in the system.
asterisk: 1.4.18
addon 1.4.5
zaptel 1.4.8
It happened suddenly. My colleagues noticed all call were failed and
asterisk process was hanged. There is no error in the system log
(message) and no core dump at all. I just find the error message in
the
The real queestion is:
What kind of provider is able to support such of a call volume?
How do you plan to provide the service ? I mean besides generated AD's will
you redirect your call to any sip providers ? Or we are dealing with mass
install in most of the regions of the world (where possible)
Ok, prepare yourself for a long week end;)
1) Replace all cables on isdn card
2) Read misdn-init.conf carefully, play with parameters, I don’t think that I
need to describe them, file is well commented.
3) Test communications port by port (dial out with each at least 5 times) If
you found that
Mike,
Most newer Linux distro's use Postfix. It's simple to setup Postfix to
use SMTP AUTH to send email. You need to figure out why the primary mail
server is rejecting the emails and go from there. Contact me off list if
you want more info.
I think I have a quick how-to I wrote for myself on
I just installed Asterisk 1.6 beta5 and moh is not working - is there a
trick? Or is something wrong with my system?
PaulH
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