On Thu, 10 Jun 2010, Michelle Dupuis wrote:
I'm looking for a small formfactor mobo for an install that needs to
handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz
mobo - anyone know what kinds of call volume that will handle?
This question comes up all the time - I've
-- Forwarded message --
From: nikhil singhania niksingha...@gmail.com
Date: 10 June 2010 14:08
Subject: asterisk registration
To: asterisk-users@lists.digium.com
Cc: Ma Hu Ma anshumishra6...@gmail.com
Hi all,
I think i understand the problem, actually I have two asterisk
Hi,
I am trying to build asterisk with xtensa compiler on embedded platform.
I am trying to integrate my driver code to asterisk. For this tying to call
driver code IOCTLs from chan_dahdi instead of dahdi IOCTLs.
While compiling asterisk with xtensa, Observed chan_dahdi is not compiling
- Original Message -
On Thu, 10 Jun 2010, Michelle Dupuis wrote:
I'm looking for a small formfactor mobo for an install that needs to
handle 25 phone sets (no transcoding). I found a new dual atom
1.66GHz
mobo - anyone know what kinds of call volume that will handle?
On Thu,
Hi,
In BRI/TE/PtmP mode, pri show span 1 used to display Type: CPE (PtMP)
(setup is bristuffed 1.2 asterisk)
It seems, it displays now Type: CPE without any mention to PtMP (with
asterisk 1.6).
I'm far from 100% sure it used to include the PtMP word but do you think it
should display it ?
Does
Hello,
I've got a running system in which logs are full of messages such as:
[Jun 10 07:24:14] NOTICE[2414] chan_dahdi.c: PRI got event: HDLC Bad FCS (8)
on Primary D-channel of span 2
The strange thing is those messages are coming from a single span.
My setup is :
Asterisk 1.6.1.18
Junghanns
Basically it means that one of the messages it received on the PRI D
channel failed the checksum.
I take it that in your span command you have 'crc4' or similar specified
as an option for all of your spans?
If thats the case its probably a faulty port on the card, cable, or a
card in the
Hi,
Am Freitag, den 11.06.2010, 11:54 +0100 schrieb Gareth Blades:
Olivier wrote:
Hello,
I've got a running system in which logs are full of messages such as:
[Jun 10 07:24:14] NOTICE[2414] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 2
The strange
2010/6/11 Karsten Wemheuer k...@gmx.de
Hi,
Am Freitag, den 11.06.2010, 11:54 +0100 schrieb Gareth Blades:
Olivier wrote:
Hello,
I've got a running system in which logs are full of messages such as:
[Jun 10 07:24:14] NOTICE[2414] chan_dahdi.c: PRI got event: HDLC Bad
FCS
(8)
Hi!
Have anybody experience with ZA16E Analog card (manufactured by
Zycoo) for Asterisk PBX (with two FXO-200 modules)?
Which dahdi or zaptel and asterisk version used?
I have several problem with that card (ZAP - SIP no audio, but
ztmonitor indicate audio level from zap)
Thanks!
crowd
--
I have a box (Genband) expecting the following:
100 trying
180 ringing with SDP
Or
100 trying
183 with SDP
And asterisk is sending:
100 trying
180 ringing
183 with SDP
Any way to modify asterisk to send what he is expecting?
Thanks,
Dave
--
What is the interest to supply binary of Asterisk, under debian for example,
while to use MeetMe it is necessary to COMPILE Asterisk ??? :-))
Mickael.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Ditto I'm running a Supermicro Atom based dual core server and it's rock
solid!!!
These make excellent servers for Asterisk installation IMHO.
On Fri, Jun 11, 2010 at 05:42, --[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
On Thu, 10 Jun 2010, Michelle Dupuis
On 06/10/10 23:19, Philipp von Klitzing wrote:
Hi!
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
CentOS 5.5. The only thing, i want to do is a call-redirection from an
isdn-call to my mobile via sip-account.
Unless you are using mISDN v2: Do yourself a favour and
On Friday 11 June 2010 09:31:43 dave george wrote:
Any way to modify asterisk to send what he is expecting?
Probably, but what you really should be asking is why the endpoint is not
RFC-compliant.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC:
I did ask this question. However it's a big carrier using Genband and they
don't care.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Friday, June 11, 2010 11:28 AM
To: Asterisk Users
Asterisk 1.6
CentOS 5.0
All -
I'd like to offer my users the ability to hangup a call by pressing **. I'm
using an attendant, so when ** is dialled I'd like processing to return to
the attendant so the user can place a subsequent call. I have setup
features.conf to include:
[featuremap]
Hi Olivier,
Am Freitag, den 11.06.2010, 14:27 +0200 schrieb Olivier:
2010/6/11 Karsten Wemheuer k...@gmx.de
Hi,
Am Freitag, den 11.06.2010, 11:54 +0100 schrieb Gareth Blades:
Olivier wrote:
Hello,
I've got a running
Hi All,
We have built an asterisk server (asterisk - 1.4.26.2) where there would be
around 322 concurrent calls going on, but I can see that full log grows
rapidly, in one day it reaches to around 10-15 GB if I turn on the sip debug
and its tedious even by using any commands to get the required
Are you using logrotate?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-11 12:14 PM, das sandesh sandesh...@gmail.com wrote:
Hi All,
We have built an asterisk server (asterisk - 1.4.26.2) where there would be
around 322 concurrent calls going on, but I can see that full log grows
Sandesh,
I'm running 1.4.23.1 right now, created this script for the logs, and
set it to run as a cron every hour. Perhaps you can modify it to help you.
#3
#
# asterisk_log_rotate.sh
#
# Removes all log files older than 7 days, then
#
All,
I am contemplating moving static SIP users to SIP realtime, and I am
wondering if there is a nice simple tool to be able to do this with? I
am not concerned with something that would do all the work for me, just
something easier to use for a decent set of changes, than pure sql or
Hello,
Is it possible to connect two *callers* without going through a conference
(meetme) ?
Example:
06:50pm - User 1 call extension 600 and musiconhold / parked call ..
06:51pm - User 2 call extension 600 and connect to User 1.
Thank you in advance,
Mickael.
--
Hi!
But if i try to establish ISDN-SIP-Dialout, the redirection ist not
working.
Your logs are very sketchy and difficult to understand because you
stripped them of some details and cut out lines in between.
From: 5 sip:s...@sip;tag=as1ec770c5
This line does not make much sense.
Hi!
I did ask this question. However it's a big carrier using Genband and
they don't care.
Look at progressinband= in sip.conf.
http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband
Philipp
--
_
-- Bandwidth
I tried no, yes and never in the sip profile for that carrier and it did not
make a difference.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von
Klitzing
Sent: Friday, June 11, 2010 1:59 PM
To:
When I use an eagi script when play a message appear a lot of warning
messages, but it play very well
I´m using
Asterisk 1.4.32
dahdi-linux-2.3.0.1
chan_ss7-1.4.1
Any ideas??
-- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0)
[Jun 11 18:12:45] WARNING[15807]: file.c:1300
On Fri, 11 Jun 2010, equis software wrote:
When I use an eagi script when play a message appear a lot of warning messages,
but it play very well
I´m using
Asterisk 1.4.32
dahdi-linux-2.3.0.1
chan_ss7-1.4.1
Any ideas??
-- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0)
On Fri, 11 Jun 2010, Mickael Monsieur wrote:
Is it possible to connect two callers without going through a conference
(meetme) ?
0) A better subject may attract the interest of someone with relevant
experience. Contacting means nothing.
1) More details will yield better responses. What
Hi people, I have a problem with some extensions. The calls are ended after
31/35 seconds, also, it depends on the number which I call.
This is the log, but I've not been able to find something wrong...
Any ideas?
[Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: ExecIf
[Jun 11
OT = Old Topic.
Any suggestions for free DID/SIP accounts?
The local Linux User Group would like to see how to set up an Asterisk
system. And in case I can't find or remember who I loaded my TDM400
card to (and because it makes sense) I'd like do a (SIP) connection to a
DID provider. Heck
On Fri, 11 Jun 2010, Roderick A. Anderson wrote:
OT = Old Topic.
Any suggestions for free DID/SIP accounts?
The local Linux User Group would like to see how to set up an Asterisk
system. And in case I can't find or remember who I loaded my TDM400
card to (and because it makes sense) I'd
Hi!
I tried no, yes and never in the sip profile for that carrier and it did
not make a difference.
Look at progressinband= in sip.conf.
Just to make sure: Maybe you forgot the SIP RELOAD?
Are you 100% sure inbound calls arrive with the peer that you set
progressinband for? Verify this
I set it under the sip profile for the box sending calls to asterisk.
[BREKEKE]
type=peer
context=wholesale
host=x.x.x.x
nat=no
canreinvite=no
progressinband=yes
dtmfmode=rfc2833
insecure=port
disallow=all
allow=g729
Thanks,
Dave George
-Original Message-
From:
This is a small 12 line system, internal extensions 150 - 180. I didn't
have a phone on 151. Here's the sip.conf stanza:
;;[151]
;;type=friend
;;context=longdistance
;;callerid=Conf Room 151
;;secret=
;;host=dynamic
;;qualify=yes
;;dtmfmode=rfc2833
;;allow=all
;;defaultuser=151
;;nat=yes
rant level=mild
I tried emailing support and got a rejection notice saying to go via
the website, which then wants registered product details before it
will let me enter a support request. Unfortunately, the card was
bought by my customer, not by me :-( So I'll just have to ask here...
/rant
I am
Steve Edwards wrote:
On Fri, 11 Jun 2010, Roderick A. Anderson wrote:
OT = Old Topic.
Any suggestions for free DID/SIP accounts?
The local Linux User Group would like to see how to set up an Asterisk
system. And in case I can't find or remember who I loaded my TDM400
card to (and
On Jun 11, 2010, at 5:55 PM, sean darcy wrote:
This is a small 12 line system, internal extensions 150 - 180. I didn't
have a phone on 151. Here's the sip.conf stanza:
--snip--
There's no DISA. And then somehow (how???) ip address 79.117.17.247
becomes extension 151 and starts making calls
Fred Posner wrote:
On Jun 11, 2010, at 5:55 PM, sean darcy wrote:
This is a small 12 line system, internal extensions 150 - 180. I didn't
have a phone on 151. Here's the sip.conf stanza:
--snip--
There's no DISA. And then somehow (how???) ip address 79.117.17.247
becomes extension 151
You`re using Xlite/eyeBeam by any chance?
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, June 11, 2010 16:12
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call ended after 31
On Fri, 11 Jun 2010, Fred Posner wrote:
On Jun 11, 2010, at 5:55 PM, sean darcy wrote:
This is a small 12 line system, internal extensions 150 - 180. I didn't
have a phone on 151. Here's the sip.conf stanza: --snip-- There's no
DISA. And then somehow (how???) ip address 79.117.17.247
On Jun 11, 2010, at 8:03 PM, sean darcy wrote:
Fred Posner wrote:
On Jun 11, 2010, at 5:55 PM, sean darcy wrote:
snipped...
What is the default context in sip.conf? Does it allow outbound calls?
;###
;DEFAULT CONTEXT
;###
[default]
When will you people learn ... you set the secret=
and it's one of the many frequent passwords most people sets out of
being lazy ...
that simply says ... guess my password and call through my pbx for free ...
so again ...
1) bad people scan extensions 100-199 and 1000- trying to guess
if you know IP then ban with iptables
iptables -A INPUT -s IP -j REJECT
Martin
On Fri, Jun 11, 2010 at 8:41 PM, Martin asteriskl...@callthem.info wrote:
When will you people learn ... you set the secret=
and it's one of the many frequent passwords most people sets out of
being lazy ...
On Fri, 11 Jun 2010, Martin wrote:
if you know IP then ban with iptables
iptables -A INPUT -s IP -j REJECT
Ever play http://en.wikipedia.org/wiki/Whac-A-Mole ?
--
Thanks in advance,
-
Steve Edwards
I am in Nigeria and I am wondering if I could get a free SIP/DID provider with
regards to my region
Thanks
_
Hotmail: Trusted email with Microsoft’s powerful SPAM protection.
lol when then if he knows the IP of his provider plus a few phones he
can just allow these ... and problem solved forever
Martin
On Fri, Jun 11, 2010 at 9:02 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 11 Jun 2010, Martin wrote:
if you know IP then ban with iptables
iptables
Roderick A. Anderson wrote:
Actually the in the US. Inland Northwest. North Idaho if anyone is
interested.
http://www.ipkall.com - Free WA state DID numbers.
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-- Bandwidth and Colocation Provided by
hello,
sounds like a T1 timeout hangupt. The T1 timeout has the default value
of 30 seconds and hangs up a call when for example the 200 OK to the
client doesnt get the ACK back.
you should look at the sip debug of client 3000 maybe you could see that
packets are resend to the client.
maybe
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