Hello,
I want to monitor my Asterisk 1.8, inbound, outbound, status calls, queue
call? Any suggestions?
I found Monast, I'm having issues configurating.
Thanks,
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it happened to me once, make sure dahdi is on;
/etc/init.d/dahdi restart
Also
Chckconfig dahdi on
If all fails, I had to re-install dahdi, and Asterisk should not be a big
deal,
cd /usr/lib/asterisk/
mv modules ./old-modules-backup
And re-install Asterisk, make sure you backup your current
Hello,
I posted this problems in the past and was not able to find the solution, at
the time I posted this issue I had a equipment malfunction which has been
fixed but I still have the -- Requested transfer capability: 0x00 - SPEECH
error.
Any suggestions?
Asterisk 1.8.17.0 built by root @
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, October 17, 2012 7:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8.17.0 Requested transfer
capability: 0x00 - SPEECH
motty.cruz wrote:
Hello,
Hola
Hello,
I have a local Asterisk server that keep loosing its registration to main
Asterisk server. The local asterisk server is on the local subnet, it acts
as a client with extension 808.
Local server
Sip.conf
register = 808:passw...@as2.x.com
registertimeout=20
registerattempts=10
Main
I can't find a clear procedure to lower musicOnHold volume!
Any suggestions?
Hereis my music.conf file
[default]
mode=files
directory=moh
Thanks in advance!
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Hello,
I'm having issues connecting throu PRI with the following error Requested
transfer capability: 0x00 - SPEECH
Below are the logs:
== Using SIP RTP CoS mark 5
-- Executing [97052660@voipphones:1] Set(SIP/4856-0003,
CALLERID(num)=x) in new stack
-- Executing
transfer
capability: 0x00 - SPEECH
On 12-09-26 10:35 AM, motty.cruz wrote:
Hello,
I'm having issues connecting throu PRI with the following error
Requested transfer capability: 0x00 - SPEECH
Below are the logs:
== Using SIP RTP CoS mark 5
-- Executing [97052660@voipphones:1] Set(SIP/4856
Hello, would like to have distintive ringtone for internal calls, google
gave me blurr answer.
My extensions are 46**, any calls made within 46** I want to ring
differently than external calls.
Thanks in advance.
--
_
--
Hello,
I have anolog lines coming throug Dahdi to Asterisk Server, one of the
anolog lines is used for fax line. I received fax fine without any problems
using Iaxmodem with Hylafax Server. Outgoint fax is the problem, when
IAXMODEM dial out using Dahdi channel, dahdi answers and start to dial
Discussion
Subject: Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem
Am 01.08.2012 17:15, schrieb motty.cruz:
Hello,
I have anolog lines coming throug Dahdi to Asterisk Server, one of the
anolog lines is used for fax line. I received fax fine without any
problems using Iaxmodem
Thanks Tim,
I tried your suggestion below the logs:
-- Accepting AUTHENTICATED call from xxx.xx.xx.xx:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing [xxx1463@fax-out:1]
...@lists.digium.com] On Behalf Of motty.cruz
Sent: Wednesday, August 01, 2012 1:11 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem
that is correct! The reason I think is because when Dahdi answered
iaxmodem thinks
Thank you very much!
I added the following line to /var/spool/hylafax/etc/config.ttyiax0
ModemWaitTimeCmd: ATS7=120
Thanks,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Wednesday,
can you post your sip.conf for Exten. 1000?
it does not seem like you have
[1000]
mailbox=1000@default
Thanks,
-motty
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, July 26, 2012 10:35
are you able to received Fax?
is sending fax the only problem?
what do you have on chan_dahdi.conf?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Agustina
Berretta
Sent: Tuesday, July 24, 2012 1:47 PM
To:
Hello,
I'm trying get fax working over VOIP lines. I'm running Asterisk 1.8.12
server, working fine, however I would like to get rid of our anolog fax
lines and integrate with our fax to our Asterisk Server.
Any recomendataion from this list? I had done some research but nothing
solid.
Hello,
I'm trying to figure out how to change the redial, thus far if I hit redial
it will redial the last called I made that was answered, not the last call I
made that was not answer.
I'm using Asterisk 1.8
Thanks,
Motty
--
Hello,
How to change ring tone on interncal call? I'm using Centos 5.8 Asterisk 1.8
exten =
666,1,SIPAddHeader(Alert-Info:http://1.2.3.4/ringtones/ghost.wav)
exten = 666,n,Dial(SIP/10)
The above would not how to defirenciate from internal call or external call?
Thanks,
motty
--
-users] Asterisk forward call
So Asterisk is playing no role during the times you have the calls
forwarded, and you just don't want it to ring? Why not just make it go
off-hook during those times?
On Tue, May 15, 2012 at 3:42 PM, motty.cruz motty.c...@gmail.com wrote:
Hello Carlos,
I'm
Hello All,
My Asterisk server is working fine except that at night I forward my number
to another phone number, however my asterisk server still rings once before
call is forward. My Local Phone provider is ATT and they said that there is
not way around it, I'm always going to get a partical
. This is by design
from ESS 1 days, and serves to remind the party that the line is forwarded.
Asterisk certainly can be configured to ignore the first ring in any number
of ways.
John Novack
Guy Gold wrote:
On Tue,May 15 02:00:PM, motty.cruz wrote:
Hello All,
My Asterisk server
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Tuesday, May 15, 2012 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk forward call
On Tue, May 15, 2012 at 2:00 PM, motty.cruz motty.c...@gmail.com wrote:
Any
@lists.digium.com
Subject: Re: [asterisk-users]Asterisk 1.8.10.1 ring tone in thebackground
after call sucessfully answered.
motty.cruz motty.cruz at gmail.com writes:
Hello I apologize for not being specific.
I'm using SIP. Our provider is call Wiline. I do not have Dahdi
install on this server
have to use this command
#pulseaudio -
Restart asterisk and it worked but if pulseaudio is not on asterisk won't
work, I'm in the process of fixing this issue.
Thanks,
-Original Message-
From: motty.cruz [mailto:motty.c...@gmail.com]
Sent: Thursday, April 19, 2012 2:30 PM
Hello, I'm trying to build a page system using a Dell Desktop PC optiplex
170L,
My sound card is working fine under /dev/snd/
exten = s,1,Dial(Console/snd/,20,A(trek))
exten = s,2,Hangup
But won't work! I get the following error
[Apr 27 11:44:46] WARNING[2950]: chan_oss.c:377 find_desc: could
Hello, I'm trying to get s extensions to autoanswer to Centos computer
speakers, the computer is a Dell Optiplex 170L embeded sound card. I'm
running Centos 6.2 i386 with Asterisk 1.8.10
Does anybody know how to fix error below?
-- Executing [s@default:1] Dial(SIP/publicip-0001,
Hello All,
I'm gettint this error, started recently when I upgraded to 1.8.10 from
1.8.4.
[Apr 17 08:03:52] ERROR[9099]: netsock2.c:263 ast_sockaddr_resolve:
getaddrinfo(external out, (null), ...): Name or service not known
[Apr 17 08:03:52] WARNING[9099]: chan_sip.c:26503 sip_request_call:
Hello All,
Is it possible to have an Asterisk server connect to a 2nd Server using
Extension?
For instance I have an Asterisk Server with public IP address then I have a
2nd Asterisk server in the local network that I want to do intercom pagin
with this server can I connect this server as
1.8.12.0-rc1
On Thu, Apr 12, 2012 at 11:04:22AM -0700, motty.cruz wrote:
Hello All,
Is it possible to have an Asterisk server connect to a 2nd Server using
Extension?
For instance I have an Asterisk Server with public IP address then I
have a 2nd Asterisk server in the local network that I want to do
Hello All,
I upgraded Asterisk from 1.8.4 to Asterisk 1.8.10.1 last week. After the
upgrade when I make a call and the other party answers the call a ringing
tone is heard in the background even thought the call was successfully
answered.
This issue is ramdom and is not consistent. I do think
tone in thebackground
after call sucessfully answered.
Which Technology are you using for the call (DAHDI/SIP/other)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Monday, March 19, 2012 10:42
Of motty.cruz
Sent: Monday, March 19, 2012 11:08 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk 1.8.10.1 ring tone in thebackground
after call sucessfully answered.
Hello I apologize for not being specific.
I'm using SIP. Our provider
Hi,
Is there a way to have a distinctive ring for the polycom phones when the
timeout is reached on a parked call? I have google this questions to no
success!
Thanks in advance!
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Hello All,
I'm having issues with asterisk 1.8.4 dropping calls during transfer, and
transfer to park extension. We're using polycom soundpoint IP 650. when the
park button is hit the response is i'm sorry not an extension at the same
time number 7 appers on the lcd.
Thanks in advance.
-motty
Hello,
I have a server that connects to my Voice Server provider so far is working
great! I have a second server that I want to set caller id to a different
number second server I'm going to call it server B. And server B will go
through server A which is connected to my Voice Server Provider.
not detect menus
We had similar problems, updating to the latest 1.8.x seems to have solved
the issue for at least one number we were having issues with.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent
Hello,
When I called companies with auto animate menus my system does not seem to
detect menus on ther other side. For instance I called this number (407)
886-3338 when I input the ext. number of any option on the list I don't get
a response however if I called the same number from my google
to the latest 1.8.x seems to have solved
the issue for at least one number we were having issues with.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Monday, January 23, 2012 2:00 PM
To: asterisk
Hello,
I see the following error in the logs
[Jan 4 11:37:35] NOTICE[21]: chan_sip.c:15388 check_user_full: From address
missing 'sip:', using it anyway
Does anybody know how to stop this error? It does not seem to be affecting
performance on the Asterisk 1.8.4.1 running on Centos Linux 2.6. I
: Thursday, October 27, 2011 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Executing outbound dial number twice
On 10/28/2011 01:02 AM, motty.cruz wrote:
Hello,
I noticed Asterisk 1.8.4.1 execute number dial twice
Log
== Using SIP RTP
outbound dial number twice
On 10/28/2011 01:02 AM, motty.cruz wrote:
Hello,
I noticed Asterisk 1.8.4.1 execute number dial twice
Log
== Using SIP RTP CoS mark 5
-- Executing [912066604@sipphones:1] Set(SIP/4773-0003e920,
CALLERID(num)=2066604) in new stack
== Extension Changed 4773
'
Subject: Re: [asterisk-users] Asterisk Executing outbound dial number twice
Please post output of CLI command dialplan show sipphones
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Friday
4783
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Friday, October 28, 2011 9:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk Executing
, October 28, 2011 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Executing outbound dial number twice
On Friday 28 October 2011, motty.cruz wrote:
Hello,
I noticed Asterisk 1.8.4.1 execute number dial twice
Log
== Using SIP RTP CoS
Hello,
I noticed Asterisk 1.8.4.1 execute number dial twice
Log
== Using SIP RTP CoS mark 5
-- Executing [912066604@sipphones:1] Set(SIP/4773-0003e920,
CALLERID(num)=2066604) in new stack
== Extension Changed 4773[sipphones] new state InUse for Notify User 4701
-- Executing
Hello,
I'm using Asterisk 1.6 with Polycom SoundPoint 650, everything is running
fine except that I can't program a button on Polycom to transfer inbound
call to Voicemail directly.
I have the following in my extension.conf
exten = _547xx,1,Voicemail(${EXTEN:1}@default,u)
Reception can
Hello All,
I have asterisk server running on Centos, some of our users are spreadout
throut the states. I want the time zone to reflect our users repective time
zones. My questions is how to customize their timez zone accordingly? Is
that done in sip.conf? or extensions.conf?
Thanks in
motty.cruz would like to recall the message, Time zone on phones.
attachment: winmail.dat--
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New to Asterisk? Join us for a live introductory webinar every
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog
telephone line coming on
Wildcard TDM400P REV E/F Board 5
I can't get asterisk to dectect call coming from analog line.
Here is my /etc/dahdi/system.conf
fxsks=1
# global data
loadzone = us
defaultzone = us
My mistake I had fix that typo but no luck
Thanks,
motty
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
Mudgett
Sent: Tuesday, June 28, 2011 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial
analog call.
Third, I think you've got some issues with your Dial statements, but I'm on
my phone right now and can't really diagnose them. I'll take a look later
when I'm back at a desk, if no one else has commented by then.
Thanks,
--Warren Selby, dCAP
On Jun 28, 2011, at 12:30 PM, motty.cruz
Hello, I'm new to this list. I'm trying to configure my Asterisk to have
user access their email. SO far users can leave voicemail but they can't
access voicemail. As you can see I had sip.conf and extensions.conf below.
Please advice how to access configure extensions.conf to have users access
Hello All,
I'm installing asterisk 1.6 on FreeBSD from ports; I'm not sure what options
should install; can anybody points to good howto on FreeBSD, I defenetely
appreciate! There are a log info for Linux but very little for FreeBSD.
Thanks,
-motty
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