I just upgraded my 7970G to the SIP firmware. What I'd like to do is have the 8
line buttons be able to make outbound calls using the same account (for
practical purposes, same caller-ID). Since the phone is going to have a single
public DID, when a call comes in, it should ring on the first ava
hugolivude wrote:
> Just upgrading to 1.4.6 from 1.2. SIP channels work OK, but not zap.
> I have a TDM400 w/ an FXO & 2 FXS. I built libpri 1.4.0 first then
> zaptel 1.4.3. Menuselect had a * beside chan_zap and I loaded the
> wcusb & wctdm before building asterisk. In the CLI "zap show
> ch
Hi,
Just upgrading to 1.4.6 from 1.2. SIP channels work OK, but not zap.
I have a TDM400 w/ an FXO & 2 FXS. I built libpri 1.4.0 first then
zaptel 1.4.3. Menuselect had a * beside chan_zap and I loaded the
wcusb & wctdm before building asterisk. In the CLI "zap show
channels" returns "no suc
Ed Nuñez wrote:
> For anyone interested on the crashes I was experiencing when using
> ChanSpy from SIP extension to SIP extensions with the group option. For
> the last couple of days, I’ve been monitoring from Zap extensions to SIP
> extensions, and the system has not crashed once. The probl
clive.chan(Alpha Trilogies Networks) wrote:
> I tried to define noload to the chan_00h323.so and res_config_mysql.so,
> my asterisk start but give me others problems as bellowing...
>
> [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'cdr_addon_mysql.so'
> did not register itself during load
>
Hi,
Does Asterisk_addons_1.4.2 cant be use in the new Asterisk release or no one
really want to share their experiences? Since this project is belonging to
everyone within this list, why still no one really want to share the
experiences and to growth the Asterisk to the next level by keeping t
shadowym wrote:
> Eagerly waiting for v1.4.x to mature a bit before getting serious about it.
> Is it ready for production yet? If that's too general, where is it in terms
> of stability compared to where 1.2.x is now. Anyone running it successfully
> in production environment and if so what sor
Is it taking a while for _your_ messages to post to the list, or do you
mean messages from the mailing list software take days to get to you?
Moj
Lenz wrote:
> Hello list,
> I am getting the list with days of delay, take for example this message:
>
> Received: from unknown (HELO lists.digiu
Andres Paglayan wrote:
> On Jun 29, 2007, at 12:50 PM, Lenz wrote:
>
>
>> Hello list,
>> I am getting the list with days of delay, take for example this
>> message:
>>
>>
>> As you can see, the message was posted on June 25th and was sent to my
>> email on June 29th! am I the only one who is g
Hi Russel,
I called on official distributor in the Miami area (Commlogik) and they
did not have it at the moment. I also tried several on-line resellers
like Voipsupply, Telephony Depot, Voxilla, and they do not have it
either. I was preparing a quote for a customer and was planning on
inclu
On Jun 29, 2007, at 12:50 PM, Lenz wrote:
> Hello list,
> I am getting the list with days of delay, take for example this
> message:
>
>
> As you can see, the message was posted on June 25th and was sent to my
> email on June 29th! am I the only one who is getting such an awful
> message
> tu
Andres wrote:
> Does anybody have any feedback on this new card from Digium? It was
> announced a couple of weeks ago but now Digium said they ran out and is
> no longer available for purchase via their web site. I find this kind
> off odd. If they are out why don't they just says its on bac
I jumped into asterisk 1.4 and its pretty stable .. i never got a core dump
but it did halt while reloading a few times . I am back on asterisk 1.2 now
but i think asterisk 1.4 is stable .
On 29/06/07, Bruce Reeves <[EMAIL PROTECTED]> wrote:
While I have not jumped all my systems to 1.4, there
Hello list,
I am getting the list with days of delay, take for example this message:
Received: from unknown (HELO lists.digium.com) (216.207.245.17) by
mxavas16.fe.aruba.it with SMTP; 29 Jun 2007 13:38:37 -
Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by
The Asterisk development team is proud to announce the releases of
versions 1.2.20 and 1.4.6!
These releases are regular maintenance releases. They have been made
just a couple of weeks after the previous set of releases because the
development team has been working especially hard on fixing bugs
Opps! Sorry wrong list
Robert A. Rawlinson wrote:
> I have Apache2 set up and running on a system I only use for testing. In
> trying to access a script that is an html and only points to a Perl
> script. When it reaches the Perl script I get this message:
> You have chosen to open filename.pl w
Rob Schall wrote:
> Anthony Francis wrote:
>> Eric "ManxPower" Wieling wrote:
>>
>>> Rob Schall wrote:
>>>
>>>
Eric "ManxPower" Wieling wrote:
> Rob Schall wrote:
>
>
>
>> I currently have about 50 polycom 501 phones on my
Sounds like your filename.pl script should be in a cgi-bin directory
rather than in a document directory?
How exactly are you doing this from asterisk? Is this for a
microbrowser in a desk phone?
Moj
Robert A. Rawlinson wrote:
> I have Apache2 set up and running on a system I only use for te
many thanks!
bye
On 6/29/07, Alexander Lopez <[EMAIL PROTECTED]> wrote:
> In the top directory of your asterisk source in the doc dir there is a
> file that explains channel variables.
>
> From that file:
> ${UNIQUEID} * Current call unique identifier
>
> BEWARE the UNIQUEID can be re
So if the it is only outgoing then no need for context
but if it is incoming or incmoing & outgoing then I
need context. Correct?
Regards
Bilal
> Yes, you can only send calls to peers, not receive
>them, so no context=
needed.
Moj
___
Hi,
Does anybody have any feedback on this new card from Digium? It was
announced a couple of weeks ago but now Digium said they ran out and is
no longer available for purchase via their web site. I find this kind
off odd. If they are out why don't they just says its on backorder or
someth
>
>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Andres Paglayan
>> Sent: Friday, June 29, 2007 12:46 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] FAX over T1
>>
>>
>> On Jun
I have Apache2 set up and running on a system I only use for testing. In
trying to access a script that is an html and only points to a Perl
script. When it reaches the Perl script I get this message:
You have chosen to open filename.pl which is a perl script from ---
What should I do with t
Hi,
Please bear with me if I'm asking stupid questions... I'm new to Asterisk,
newish to Linux, etc...
I've got MoH working nicely with my new Asterisk setup using the "files"
option; except that it always plays from the start of a (random) music file
when you first put someone on hold. Take them
For anyone interested on the crashes I was experiencing when using ChanSpy
from SIP extension to SIP extensions with the group option. For the last
couple of days, Ive been monitoring from Zap extensions to SIP extensions,
and the system has not crashed once. The problem only happens when I spy
On 6/29/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
>
> What is a good solution for playing music on hold on the 1.2 branch. I do
> not want to use mpg123 because last time I used it in a production server it
> caused many problems. The MPG123 process was taking about 60% of my Xeon
> CPU.
>
For mi
While I have not jumped all my systems to 1.4, there were some that I have
moved to 1.4 and I have found it to be as stable as 1.2 was on those
machines.One of the systems is a 10 user office with Sangoma cards and
another is a 70+ user pure voip system. In both cases I have warning
messages about
What is a good solution for playing music on hold on the 1.2 branch. I do not
want to use mpg123 because last time I used it in a production server it caused
many problems. The MPG123 process was taking about 60% of my Xeon CPU.
___
--Bandwidth a
Anthony Francis wrote:
Eric "ManxPower" Wieling wrote:
Rob Schall wrote:
Eric "ManxPower" Wieling wrote:
Rob Schall wrote:
I currently have about 50 polycom 501 phones on my asterisk setup.
The dialplan is set to work with mysql (realtime), and all of
I'm going to top post in this situation.
Kevin - Commands that operate on the channel variables won't help if we
are using a call file. We will have a new channel.
This syntax works with asterisk 1.2.x for me:
Application: AGI
Data: say_it.php|call_status_message
I have done other things where
hi,
i am using Asterisk 1.4. and unable to get Voice Mail below is my config
extensions.conf
exten => 50,1,NoOp(Failover)
exten => 50,2,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b)
exten => 50,3,Dial(SIP/101,18)
exten => 50,4,Goto(ss-${DIALSTATUS},1)
exten => ss-NOANSWER,1,StopMixMonitor()
ext
Hi All,
Eagerly waiting for v1.4.x to mature a bit before getting serious about it.
Is it ready for production yet? If that's too general, where is it in terms
of stability compared to where 1.2.x is now. Anyone running it successfully
in production environment and if so what sort of config d
Eric "ManxPower" Wieling wrote:
> Rob Schall wrote:
>
>> Eric "ManxPower" Wieling wrote:
>>
>>> Rob Schall wrote:
>>>
>>>
I currently have about 50 polycom 501 phones on my asterisk setup.
The dialplan is set to work with mysql (realtime), and all of the
extensions
Yes, you can only send calls to peers, not receive them, so no context=
needed.
Moj
bilal ghayyad wrote:
> Hi Noah;
>
> The reason that I am asking wether I need to determine
> the context is what I read in the documentation (about
> configuring outbound IAX connections), it did not
> mention th
Hi Steve;
I did what I told me below, and look like going fine
but I do not know how can I know that zaptel
compilation was implemented successfully specially I
do not have a message in the end indicate this, please
find below what the make and make install commands
(for zaptel compilation) was en
You're including a context in your dialplan that doesn't exist. Given
that it has been prefixed with AEL, I'd check extensions.ael for the
Asterisk Extension Language sample file. I bet it does some including.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Resear
I have been trying for a very long time to get asterisk to detect and
utilize dtmf tones from my sip clients within my dial scripts. I have
set automon=>#9 in my features.conf, I have Dial(,gWw) in my dial
scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in
my extension. I can
Dear Users !
I have recently installed asterisk 1.4 i got a warning message whenever i
use reload or extensions reload.
[Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes:
Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls'
[Jun 29 19:22:11] WARNING[
hi all!
I have the following situation:
1 2
¦¦
¦¦
3--4
¦¦
¦¦
5--6
where 1 ... 6 are nodes and every direct neighbor is specified as a dundi peer
(in *). When I start a dundi request, every queried node is m
Nitesh Divecha wrote:
> Hello All,
>
> Is there any way to pass additional parameters while calling AGI from
> *.call file?
>
> Channel: Local/[EMAIL PROTECTED]
> MaxRetries: 0
> RetryTime: 15
> WaitTime: 15
> Application: AGI
> Data: recordvoice.php
>
> Something like Data: recordvoice.php?id=345
I think in your SIPDefault.cnf you disable VAD
enable_vad: 0 ; VAD setting 0-disable (Default),
1-enable
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
Sent: Friday, June 29, 2007 9:28 AM
To: asterisk-users@lists.digium.com
S
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andres Paglayan
> Sent: Friday, June 29, 2007 12:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FAX over T1
>
>
> On Jun 22, 2007,
Hi Everyone
Got a newbie type question regarding MOH & Cisco phones.
I'm still new to Asterisk (very new in fact) & built up a AsteriskNOW box
just to get something going.
My simple test system has just 3 Cisco phones a 7905, 7940 & 7960. -
Everything's running SIP.
The 3 phones can call ea
Hi,
I am trying to establish call through sip phone between two PC connected to
linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53
I am trying to dial from 1st PC to 2nd PC through asterisk server
The problem is 1s
Hi Noah;
The reason that I am asking wether I need to determine
the context is what I read in the documentation (about
configuring outbound IAX connections), it did not
mention the context at all, please read the below
paragraph (I copy it from the documentation and paste
it):
Configuring Outboun
I am trying to implement a Centralized Call Waiting System. I have red
some document about asterisk group features to manage group and
category of a sip channel.
I have done a lot of test about it but always it doesn't work
correctly if I transfer the call.
This is the macro code I use for inboun
I'm using asterisk 1.4.5 , on Centos. My kernel version is 2.6.9-55.ELsmp
I've configured the nway call. I made entries in extension.conf, feature.conf,
as per required.
I'm trying to make a 3-way conference with the 1 user myself ( using asterisk),
and two others are PSTN line users.
I'm maki
Hi
how can i retrieve the call unique id of asterisk in the dialplan?
I have enabled the cdr logging on a postgres database.
In the table cdr each record has a field that assumes an unique id
(for example: 1141628669.51)
Can i retrieve this from the dialplan?
For example:
exten => 203,1,Answe
http://www.voip-info.org/wiki/view/Asterisk+dimensioning
Santosh S Kumar wrote:
> Hi,
>
> We are planning to develop a product making asterisk as base, I love
> that asterisk is open source and eager to start working on it. But
> before even we get into start working on asterisk we want to know
Hi,
We are planning to develop a product making asterisk as base, I love that
asterisk is open source and eager to start working on it. But before even we
get into start working on asterisk we want to know how many number of
parallel calls can be made from a single asterisk box, considering we
in
Hi Remco
I have used the IP600 v3 with SIP support on Asterisk... apparently I
was the 1st person globally to run it at a site. The 1st firmware was a
bit buggy at times, but seems to be much better on the later versions.
Kind Regards
Garth
Garth van Sittert
BSc (Physics & Computer Science)
-
If you flash new sip flash firmware into 7941 look at tftp log, you will
see, that after firmware flashing and phone reboot, it will download and
flash localization files in next flashing cycle,
if you copy this files from callmanager tftp dir to your tftp server it
will work.
before flasing loc
here's my output of /proc/zaptel/4:
Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4" HDB3/CCS/CRC4
94 TE4/0/4/1 Clear (In use)
95 TE4/0/4/2 Clear (In use)
96 TE4/0/4/3 Clear (In use)
97 TE4/0/4/4 Clear (In use)
98 TE4/0/4/5 Clear (In use)
99
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