hello, all of users:
i want to test the dahdi with asterisk-1.6, but there is no much source for
this new project. the only information i got is from voip-info. my problem is
that, i can not enable the chan_zap, therefore i do not have chan_zap.so in
asterisk/modules, i can not remane it to
On Tue, 26 Aug 2008, aymen warfalli wrote:
Hi
as maximum link capacity could be calculated using codecs and channel
types so , regarding the CPU and processors load , Is there any formula
or (any relations could help ) that can give the maximum CPU load
(mainly processor and RAM ) or
On Tue, 26 Aug 2008, JR Richardson wrote:
Hi All,
I received a request for a special application and need some guidance.
Cust has there own Asterisk PBX with SIP phones, pretty standard
setup.
They want an after hours application that checks inbound caller ID
numbers and matches them to a
Hi list! Thank for the help. Now, I can call the 8500 to listen to the
inbound messages, change pin, but I have another problem. When I call a
SIP extension configured in the MySQL database it says: Call from '101'
to extension '102' rejected because extension not found. My vmusers
table:
On Tuesday 26 August 2008 11:44:42 pm Karl Fife wrote:
I'll be that none of the other coffee makers can handle anywhere NEAR 60
voice channels, and don't get me started about HPEC!
http://www1.shopzilla.com/8N_-_cat_id--13050802__oid--680459759
Good find! Does it grind it's own beans?
--
On Wed, Aug 27, 2008 at 02:09:27PM +0800, lizhong zhu wrote:
hello, all of users:
i want to test the dahdi with asterisk-1.6, but there is no much
source for this new project. the only information i got is from
voip-info. my problem is that, i can not enable the chan_zap,
therefore i do
On Tue, Aug 26, 2008 at 7:26 PM, Bob Pierce [EMAIL PROTECTED] wrote:
On Tue, 2008-08-26 at 17:53 +0300, Atis Lezdins wrote:
Are there any plans to back port this feature into upcoming 1.4
releases?
No, new features are added only in trunk, and released in next major
release (1.6).
So
Doesn't Queuemetrics run on a license basis?
Anything else that's probably open source and free?
Does anyone have any comments/experience about using asteriskguru queue
statistics?
http://www.asteriskguru.com/tutorials/installation_guide.html
___
Hi List,
I recently switched to asterisk-1.6-beta9 because of the RPID support,
but ran into the Problem, that the RPID-Header is not sent.
sendrpid is set to yes in my sip.conf, and i'm even sure that the
add_header() function is called in chan_sip.c, but when i capture the
SIP-Packets,
Hi Again,
Is there a way i can detect whether a user has been added into the regcontext?
Currently i'm seeing this and just gives a fast busy.
[Aug 27 16:44:46] WARNING[17402]: pbx.c:2483 __ast_pbx_run: Channel
'SIP/10..10.10.10-b63101d0' sent into invalid extension '141100' in context
Hello,
On a 1.2 Asterisk / Debian Sarge, I noticed that :
ipbx*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
4201/4201 192.168.100.111 D 5060 OK (8 ms)
4200/4200 192.168.100.110 D 5060 OK (8
Probably another left over word from another message. Is it repeatable?
On 27 Aug 2008, at 13:00, Olivier wrote:
Hello,
On a 1.2 Asterisk / Debian Sarge, I noticed that :
ipbx*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
4201/4201
A closer inspection shows ^@ between on and Name as if these letters came
from a word previously cut (from connexion ?)s o shell command would show
# asterisk -rx sip show peers
on
[EMAIL PROTECTED]/username HostDyn Nat ACL Port
Status
4201/4201
2008/8/27 Steven Howes [EMAIL PROTECTED]
Probably another left over word from another message. Is it repeatable?
At the moment, yes.
Now, I'm looking for a way to flush input/output, to protect shell script
from this type of side effect.
On 27 Aug 2008, at 13:00, Olivier wrote:
Hello,
On 27 Aug 2008, at 13:23, Olivier wrote:
2008/8/27 Steven Howes [EMAIL PROTECTED]
Probably another left over word from another message. Is it
repeatable?
At the moment, yes.
Now, I'm looking for a way to flush input/output, to protect shell
script from this type of side effect.
[EMAIL
Is this a one VIP to one cell number match? Or is it on VIP to multiple
cells?
On Tue, Aug 26, 2008 at 7:28 PM, JR Richardson [EMAIL PROTECTED]
wrote:
Hi All,
I received a request for a special application and need some guidance.
Cust has there own Asterisk PBX with SIP phones,
Hi
There are some tools that you may hold serve, check these link:
http://www.bandcalc.com/
http://codec-calculator.softonic.com/mac
Miguel Otamendi
2008/8/27 Gordon Henderson
[EMAIL PROTECTED][EMAIL PROTECTED]
On Tue, 26 Aug 2008, aymen warfalli wrote:
Hi
as maximum link capacity
Sure, let me show you how I setup dundi on systems.
extensions.conf
exten = _1X,1,Goto(lookupdundi,${EXTEN},1)
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Playback(invalid)
You can have the i do whatever you want, and you can use the same
option in the macro
On Wed, 2008-08-27 at 11:21 +0300, Atis Lezdins wrote:
If you doubt about some part, you're welcome to ask, i'll try to help
you, but i don't want to provide complete backport to you, as i won't
be able to test it :)
Thanks Atis,
I'll probably try this in a few weeks when I start rebuilding
Ciao Noah,
What flags do you have in your Dial() statement? If you want both
parties to be able to transfer with the features.conf transfer, you
need to have 'Tt' in your dial statement, like this:
Dial(IAX2/user:[EMAIL PROTECTED]/exten,20,Tt)
Bingo. That was the problem.
Thanks a lot,
--
I think we're getting closer now as obviously Asterisk's greeting (...UNIX
connection) is mixed with its output.
(I can't understand why this happens now as I never noticed this before and
didn't change anything).
I tried to use asterisk -rx '!sleep 1 sip show peers' to works around but
:
1.
Does anyone know of a pri splitter device? Something that would take an
incoming PRI, and based on DID send that out one of other multiple PRI ports?
I'm needing to take a single PRI from the telco, and send it to two separate
phone systems(one asterisk) based on DID.
I know I could probably
On 27 Aug 2008, at 14:21, Olivier wrote:
I think we're getting closer now as obviously Asterisk's greeting
(...UNIX connection) is mixed with its output.
(I can't understand why this happens now as I never noticed this
before and didn't change anything).
I tried to use asterisk -rx
Jeremy Mann wrote:
I know I could probably achieve the same thing with a 3 port PRI card in
a server, but I’d like something braindead easy to configure from both a
hardware and software perspective.
Anything you use is going to (essentially) be a 3-port ISDN PRI capable
switch, because that
Hello again..
I am working on using call files to have a form of ringback - eg if an
extension is busy, the caller can dial a number and when the callee is
free, the call gets made.
I am trying to use a call file, which kind of works okay, however, if
users have voicemail, it connects to
On Mon, Aug 25, 2008 at 6:21 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Given that this is the case, we may want to do one of the following:
a) document that qualify=yes is incompatible with realtime, unless
rtcachefriends is turned on, b) automatically disallow qualify=yes if the
peer is
It does work, here !!
Thanks you very much !!
2008/8/27 Steven Howes [EMAIL PROTECTED]
On 27 Aug 2008, at 14:21, Olivier wrote:
I think we're getting closer now as obviously Asterisk's greeting
(...UNIX connection) is mixed with its output.
(I can't understand why this happens now as I
Hi everybody
Here is my zapata.conf file and extension.conf file
zapata.conf
[channels]
;switchtype=national
;pridialplan=national
;signalling=pri_cpe
context=test
group=1
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
In article [EMAIL PROTECTED],
Norman Franke [EMAIL PROTECTED] wrote:
Does any have some good experience with the various freetds variants?
Is 0.64 better or worse than 0.82? I know that to use 0.82 you have to
use ODBC, since libtds.a is not long installed. Which is more stable?
I plan
On Wednesday 27 August 2008 08:56:02 J.M. wrote:
On Mon, Aug 25, 2008 at 6:21 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Given that this is the case, we may want to do one of the following:
a) document that qualify=yes is incompatible with realtime, unless
rtcachefriends is turned on,
Hi All,
I'm using A2billing application in order to make callback calls through my
asterisk server...Everything looks fine except the voice quality...There is
a lot of cuts in the call with different codecs(G711, and G729)...Please
note that when making a call from any extensions to the same
We've done the asterisk passthrough route, but if the asterisk box is down for
whatever reason both systems are down.
Splitter wasn't the right word, but yes I see your point, I'll look into the
Adtran.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
On Wednesday 27 August 2008 09:29:55 Tilghman Lesher wrote:
On Wednesday 27 August 2008 08:56:02 J.M. wrote:
On Mon, Aug 25, 2008 at 6:21 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Given that this is the case, we may want to do one of the following:
a) document that qualify=yes is
Friday August 29, 2008 at 12 Noon EDT - VoIP Users Conference
Hi,
I was surprised to learn last week from Digium that asterisk 1.6 has
fax supported included. No more copying source files and editing
Makefile. While those tasks are not a big deal, they sort of defeat
the idea of using make. I
Hi all, I'm trying to send fax from Hylafax to a remote fax machine
through Asterisk and cisco 2801 as E1 gateway.
This is my architecture:
sendfax - HylaFax - iaxmodem - Asterisk - (SIP) 2801 with E1 card
For incoming fax I don't have any problem, but I'm not able to send fax
out of 2801.
My
On Wed, Aug 27, 2008 at 10:16:34AM -0400, [EMAIL PROTECTED] wrote:
Hi everybody
Here is my zapata.conf file and extension.conf file
zapata.conf
[channels]
;switchtype=national
;pridialplan=national
;signalling=pri_cpe
context=test
group=1
usecallerid=yes
hidecallerid=no
On Wed, Aug 27, 2008 at 07:46:10AM -0700, randulo wrote:
Friday August 29, 2008 at 12 Noon EDT - VoIP Users Conference
Hi,
I was surprised to learn last week from Digium that asterisk 1.6 has
fax supported included. No more copying source files and editing
Makefile. While those tasks are
On Wed, Aug 27, 2008 at 6:20 PM, Enrico Pasqualotto
[EMAIL PROTECTED] wrote:
Hi all, I'm trying to send fax from Hylafax to a remote fax machine
through Asterisk and cisco 2801 as E1 gateway.
This is my architecture:
sendfax - HylaFax - iaxmodem - Asterisk - (SIP) 2801 with E1 card
For
On Tue, Aug 26, 2008 at 08:36:35PM -0400, SIP wrote:
Jay R. Ashworth wrote:
On Tue, Aug 26, 2008 at 05:10:35PM -0400, Asterisk wrote:
The shared desktop is available using a Java enabled browser at
???http://callin.xelatec.com/vnc??? with a password of ???aretta???.
Of course you
Hi all.
This is my first post here and I searched a lot for a solution without luck.
Heres the problem; When I make a call forwarding from a extension to an
external number (cell phone) it never work. Only work if the forward
goes to another local extension. I dont know exactly what kind of
Does anybody have an idea how to pass Off Hook caller ID to Asterisk via
Linksys ?
I'm getting caller ID type I OK but when another customer rings the phone (when
I'm on line) the CID off hook is not coming through.
I think Off-Hook CID is called CID type II, isn't it?
--
#Joseph
GPG KeyID:
i kinda have a relevant prob!
my sipura 3102 wont pass CID to asterisk even though ive enabled such a feature
in its web gui!
Date: Wed, 27 Aug 2008 12:07:51 -0600
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Off-Hook (type II) CID passing to
Dpto. Datos Television Costa Blanca schrieb:
This is my first post here and I searched a lot for a solution without luck.
Heres the problem; When I make a call forwarding from a extension to an
external number (cell phone) it never work. Only work if the forward
goes to another local
On Wed, Aug 27, 2008 at 8:42 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
part of the asterisk install. I wonder where the fax code comes from?
SpanDSP?
Yes. Spandsp (0.0.5pre) is required for that.
Hi Tzfrir,
So this will be an option in selectmenu? (or menuselect or whatever
it's called,
Hi,
I've had the following problem with all Polycom phones. They will dial
a real SIP URI such as [EMAIL PROTECTED] but they will not
dial [EMAIL PROTECTED] which is the Talkshoe SIP server. Yet, any software
client I use and my Linksys SPA 941 will call both. The same is true
for the [EMAIL
On 08/27/08 22:29, RoLaNd RoLaNd wrote:
i kinda have a relevant prob!
my sipura 3102 wont pass CID to asterisk even though ive enabled such a
feature in its web gui!
I can help you out with this, it easy :-)
Tell me what you have enabled?
In addition to the obvious one you need to set delay
Has anyone tried using a Linksys One phone (such as the PHM1100) with
Asterisk?
--
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca
www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
Open Source
One of the Asterisk people down here in Melb set it up for the company
they used to work for, and I played with it once and it seemed to be usable.
PaulH
Lee, John (Sydney) wrote:
Doesn't Queuemetrics run on a license basis?
Anything else that's probably open source and free?
Does
Asterisk?
PaulH
Jeremy Mann wrote:
Does anyone know of a pri splitter device? Something that would take
an incoming PRI, and based on DID send that out one of other multiple
PRI ports?
I’m needing to take a single PRI from the telco, and send it to two
separate phone systems(one
Hi guys,
What are your suggestions to people who have pbx systems that interface with
the world over pri and want to convert them to sip interfaces so that they
can use sip trunking?
Tom
___
-- Bandwidth and Colocation Provided by
Why a three-port PRI card?
Just put a two-port card into your Asterisk server, pull off those DIDs
you want to process locally, and send the rest over the second port to
the PBX. In the reverse direction, intercept calls from the PBX to the
Asterisk DIDs but pass everything else to the telco.
Adtran Atlas 550. We were bring in a single pri into an atlas 550 and then
splitting it up so that 6 channels went to a video system (h.320) and 17
channels to our PBX. You can also convert the signaling or send out on
different type of connections like v.35. Pretty cool device and rock solid.
Are you using an Asterisk PBX?
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 27, 2008, at 7:06 PM, Tom Moore wrote:
Hi guys,
What are your suggestions to people who have pbx systems that
interface
Asterisk.
PaulH
Tom Moore wrote:
Hi guys,
What are your suggestions to people who have pbx systems that interface with
the world over pri and want to convert them to sip interfaces so that they
can use sip trunking?
Tom
___
-- Bandwidth and
No, these are mainly Samsung pbx systems.
I know I can use Asterisk to do this but what be a solid platform to go with
that can go in the phone closet?
tom
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Sessions
Sent: Wednesday, August 27, 2008 9:22 PM
To:
Joseph wrote:
Does anybody have an idea how to pass Off Hook caller ID to Asterisk via
Linksys ?
I'm getting caller ID type I OK but when another customer rings the phone
(when I'm on line) the CID off hook is not coming through.
I think Off-Hook CID is called CID type II, isn't it?
You can use an extremely simple Asterisk config to do the SIP-PRI
call conversion that'd be very solid.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 27, 2008, at 7:37 PM, Tom Moore wrote:
No, these
Are you looking for a hardware suggestion or a software suggestion?
PaulH
Tom Moore wrote:
No, these are mainly Samsung pbx systems.
I know I can use Asterisk to do this but what be a solid platform to
go with that can go in the phone closet?
tom
- Talk to a service provider that provide VoIP services.
- Does your PBX support SIP ?
- Does your PBX also provides Topology hiding and NAT traversal , otherwise,
you may need a session border controller .
- Does your Service provider's softswitch has proven interworking tests with
the brand of
Hi all,
I have the following queue and members. I found that there is a
call stuck in the queue so other call can't enter the queue. I want
to know whether we can remove the call (by CLI) to free the queue.
ango
2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s
holdtime),
On Wed, 27 Aug 2008 14:05:05 -0700, randulo wrote:
Hi,
I've had the following problem with all Polycom phones. They will dial
a real SIP URI such as [EMAIL PROTECTED] but they will not
dial [EMAIL PROTECTED] which is the Talkshoe SIP server. Yet, any software
client I use and my Linksys SPA 941
2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s
holdtime), W:0, C:134, A:48, SL:88.8% within 120s
Members:
Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
This an FXO card, but for some reason it is configured as an FXS card on
asteriskas per asterisk install Guide...
--- On Tue, 8/26/08, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
From: Eric ManxPower Wieling [EMAIL PROTECTED]
Subject: Re: [asterisk-users] X100P Card in OFFHOOK state
To:
Thx I will try that.
--- On Tue, 8/26/08, Guillermo Salas M. [EMAIL PROTECTED] wrote:
From: Guillermo Salas M. [EMAIL PROTECTED]
Subject: Re: [asterisk-users] X100P Card in OFFHOOK state
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Tuesday,
Hi,
Try CLI soft hangup Local.
Andy
On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
I have the following queue and members. I found that there is a
call stuck in the queue so other call can't enter the queue. I want
to know whether we can remove the call (by CLI) to
hello, all of users:
i have a problem with loading chan_dahdi.so. when i start asterisk, it always
reports the can not open channel 1 in ...
here is my setting: in etc/system/dahdi.conf:
# Global data
fxsks=1
fxsks=2
fxoks=3
fxoks=4
loadzone= us
defaultzone = us
-
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