independently.
Sorry but other than this there's little I can do, maybe someone else has
experience with this.
Alyed
2010/3/29 Ott Rose sixfourimp...@hotmail.com
i posted this on the freepbx site. here is the response
from the trace, everything is working. Check your asterisk log for file
place.
Pls look for them in the server you are actually having the problems with cause
I can't remember that sound file being on the official's asterisk release.
Alyed
2010/3/30 Ott Rose sixfourimp...@hotmail.com
where are those sound files kept? i looked last night in
/var/lib/asterisk
of the FreePBX users list.
Alyed
2010/3/26 Ott Rose sixfourimp...@hotmail.com
i have posted this question couple of times and never really got any hits i
wasn't able to provide any debug info
Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309)
Verbosity is at least 4
list.
Alyed
2010/3/26 Ott Rose sixfourimp...@hotmail.com
i have posted this question couple of times and never really got any hits i
wasn't able to provide any debug info
Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309)
Verbosity is at least 4
== Using
Date: Fri, 26 Mar 2010 00:30:50 +0200
From: tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] new server install errors starting asterisk
On Thu, Mar 25, 2010 at 09:58:17PM +, Ott Rose wrote:
well here is what i did to solve it but i
i have posted this question couple of times and never really got any hits i
wasn't able to provide any debug info
Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309)
Verbosity is at least 4
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using
Date: Thu, 25 Mar 2010 11:30:49 +1300
From: li...@venturevoip.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] new server install errors starting asterisk
Just try running:
asterisk -vcd
here you go.
[r...@phoneserver src]# asterisk -vcd
Asterisk
@lists.digium.com
Subject: Re: [asterisk-users] new server install errors starting asterisk
On 25 Mar 2010, at 13:08, Ott Rose wrote:
Can't find indications config file indications.conf.
Thats the last line. Probably the problem... Amazing what reading
instructions does...
S
:02, Ott Rose wrote:
well i followed the same directions i used like 3 weeks ago with 1.6.0 and
didn't have any issue. Not sure what went wrong. That why i posted it.
how can it work one time and not the next.
Does the file exist? If not, then something is different. You probably
like amportal script should start asterisk for me.
S
On 25 Mar 2010, at 16:46, Ott Rose wrote:
so i went back to 1.6.1.18 and didn't have any issue with the install.
following the same setups as before with 1.6.2.
finished the install and now i have an issue were asterisk doesn't
are not in there.
looks like amportal script should start asterisk for me.
S
On 25 Mar 2010, at 16:46, Ott Rose wrote:
so i went back to 1.6.1.18 and didn't have any issue with the install.
following the same setups as before with 1.6.2.
finished the install and now i have an issue were asterisk
what is the general view about the versions of the packages that are used with
asterisk.
lame
asterisk
asterisk-addons
dahdi
libpri
i like to say on a version and not upgrade due to my experience with Linux and
upgrading screwing up things. When it comes to Asterisk i have only one server
thanks for hijacking my thread.
i have an idea don't help him/her so that people will help me!
now i am going to re-post this.
Date: Wed, 24 Mar 2010 09:08:02 -0700
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] software version
On
what is the general view about the versions of the packages that are used with
asterisk.
lame
asterisk
asterisk-addons
dahdi
libpri
i
like to say on a version and not upgrade due to my experience with
Linux and upgrading screwing up things. When it comes to Asterisk i
have only one server
/3/24 Ott Rose sixfourimp...@hotmail.com
thanks for hijacking my thread.
i have an idea don't help him/her so that people will help me!
now i am going to re-post this.
Date: Wed, 24 Mar 2010 09:08:02 -0700
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re
i have this card installed
Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express)
following the steps below found on freepbx site
cd /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0
make
make install
make config
/sbin/ztcfg
echo /sbin/ztcfg
/etc/rc.d/rc.local
cd
Date: Wed, 24 Mar 2010 21:42:09 +0200
From: tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] installing dahdi card
On Wed, Mar 24, 2010 at 07:18:52PM +, Ott Rose wrote:
i have this card installed
Digium, Inc. Wildcard AEX800 8-port
Date: Wed, 24 Mar 2010 22:11:28 +0200
From: tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] installing dahdi card
On Wed, Mar 24, 2010 at 07:59:56PM +, Ott Rose wrote:
Date: Wed, 24 Mar 2010 21:42:09 +0200
From: tzafrir.co
here is the issue
phones freepbx-2.7.0]# ./start_asterisk start
STARTING ASTERISK
Asterisk ended with exit status 1
Asterisk died with code 1.
Automatically restarting Asterisk.
Asterisk ended with exit status 1
Asterisk died with code 1.
Automatically restarting Asterisk.
mpg123: no process
Date: Tue, 16 Mar 2010 10:38:00 -0400
From: m...@mattgwatson.ca
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] dnd
On Mon, Mar 15, 2010 at 1:44 PM, Ott Rose sixfourimp...@hotmail.com wrote:
I did a clean install to freepbx 2.6.1 and now when i do *76 i get a 1 second
I did a clean install to freepbx 2.6.1 and now when i do *76 i get a 1 second
flash on the screen then the phone hangs up. the FOP says it is on DND
but some ext are still getting calls. once i do a *76 FOP still says I
am on dnd. I am running asterisk 1.6.0.21.
before i was getting a message
running freepbx 2.6.1 and asterisk 1.6.0.21
i did a clean install with the versions listed above. I was on freepbx 2.5.0
and asterisk 1.6.0 ( i think)
I have aastra 57i phones. with the old versions i could hit the dnd(*76) button
and i would hear dnd activated and the the light at the top
Is there a way to tell if an extension is in use? We run a call center and it
would be helpful for the people taking calls to see if we are on the phone or
DND. Is that setup in Asterisk or on the phone? the phone as busy lamp field
but i will just turn on after a while even if the extension
you can get debug info a couple of ways from the asterisk CLI. I like this
command the best. sip set debug ip xxx.xxx.xx.xxx where xxx.xxx.xxx.xxx is the
of the x-lite phone. It will give you a lot of info. I haven't figured out how
to redirect output yet.
Date: Fri, 30 Oct 2009 13:05:35
i don't know what your are talking about (sig)
B) what trash?
c) dont thinks so
From: h...@a-domani.nl
To: asterisk-users@lists.digium.com
Date: Wed, 28 Oct 2009 22:16:16 +0100
Subject: Re: [asterisk-users] need a local tech
On Wed, 2009-10-28 at 14:59 +, Ott Rose wrote:
I am sure
thanks Cohen
Date: Wed, 28 Oct 2009 23:46:12 +0200
From: tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] need a local tech
On Wed, Oct 28, 2009 at 10:16:16PM +0100, Hans Witvliet wrote:
On Wed, 2009-10-28 at 14:59 +, Ott Rose wrote
I am sure many of you have seen my post asking question that I cannot seem to
resolve. While the responses i have been getting have been helpful i still
cannot seem to get this working 100%.
So I have waving the white flag here. I give up. I need someone to come to my
office and help me get
, 2009 at 1:32 PM, Ott Rose sixfourimp...@hotmail.com wrote:
Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at
your install and they said we are having a NAT problem but didn'ttell me if
it was with the asterisk conf or the Cisco ASA
on it?
you can also try to use the stun server ... asterisk has it built in
...never used it but saw it's there
Martin
On Tue, Oct 20, 2009 at 1:32 PM, Ott Rose sixfourimp...@hotmail.com wrote:
Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at
your install
and SIP -
http://www.aocomputing.net/?p=3. This is the link given if you were to ask
this same question in the IRC channel...
--wcs
On Wed, Oct 21, 2009 at 9:59 AM, Ott Rose sixfourimp...@hotmail.com wrote:
Here is what i think the is helpful from wireshark
OPTIONS sip:216.82.224.202
Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at your
install and they said we are having a NAT problem but didn'ttell me if it was
with the asterisk conf or the Cisco ASA.
After i rebuilt my server i did default install of Asterisk using the steps off
freepbx site. i used these steps before without any issues. this time i have to
start Asterisk manually every time the server reboots. if i start it by using
./start_asterisk script in the freepbx directory i get
On Mon, 19 Oct 2009, Ott Rose wrote:
After i rebuilt my server i did default install of Asterisk using the
steps off freepbx site. i used these steps before without any issues.
this time i have to start Asterisk manually every time the server
reboots.
[snip]
i am guessing
here is the debug from the CLI. I think I know where the problem is I just can
figure out how to fix it. The IP in the From and To i think is where the
problem is. When I make an outbound call. i get the message the call cannot be
completed as dialed. if i call another ext it works. I posted
I would like to have my ip phone list a menu with different status on it. for
example, i could have button on the phone named status. when the button is
pressed a list would display on the screen like 1:dnd, 2:break, 3:lunch, etc.
we use aastra 57i phones. It looks like i could use xml some
i have posted this before but was unable to resolve it. i have some new info so
i figured i would try again. the trace from bandwidth.com are below. they are
telling me that the ip that is bold should be our ip not bandwidth.com. i have
changed every setting that i can see and nothing fixes
will
have to help you as I don't. May I also suggest that you bottom post in
future responses rather than top post; that makes it a little easier to
follow. Good luck - John
On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote:
here is my sip.conf. i don't see
I put a post on here about my issues with outbound calls not ringing but i
haven't resolved it. so i am trying again.
When i dial any outside number i dont get a ring tone at all. when the person
picks up and starts to talk i can hear them fine. it sounds great. How do I
start to troubleshot
we are using Aastra 57i
i don't see that setting. where is it at?
From: jsulli...@opensourcedevel.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 Aug 2009 11:07:21 -0400
Subject: Re: [asterisk-users] outbound calls not ringing
On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
I
-19 at 15:55 +, Ott Rose wrote:
we are using Aastra 57i
i don't see that setting. where is it at?
From: jsulli...@opensourcedevel.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 Aug 2009 11:07:21 -0400
Subject: Re: [asterisk-users] outbound calls not ringing
how do i troubleshoot no ring tone. It was working and all i added was the
lines below now it doesn't ring.
Edit sip_nat.conf for proper NAT:
localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external
hostname name here)
externrefresh=10
fromdomain=DOMAIN.com (Set your
yes i just copied that form the freepbx site. sorry about that
From: st...@geekinter.net
To: asterisk-users@lists.digium.com
Date: Fri, 14 Aug 2009 15:43:00 +0100
Subject: Re: [asterisk-users] no ring tone
On 14 Aug 2009, at 15:18, Ott Rose wrote:
how do i troubleshoot no ring tone
\
externip=public ip
And work fine
Regards
On Fri, Aug 14, 2009 at 9:18 AM, Ott Rose sixfourimp...@hotmail.com wrote:
how do i troubleshoot no ring tone. It was working and all i added was the
lines below now it doesn't ring.
Edit sip_nat.conf for proper NAT:
localnet=192.168.1.0
] no ring tone
Hello
One question
In sip.con or sip_additionals.conf, in freepbx, the context of your client do
you put
nat = yes
externip =
You put your public ip.
Are you sure that?
Regards
On Fri, Aug 14, 2009 at 10:43 AM, Ott Rose sixfourimp...@hotmail.com wrote:
i changed
lines and a bad call fails on all 3? Are
there numbers that alternate between good and bad?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Wednesday, August 12, 2009 11:39 AM
? Are
there numbers that alternate between good and bad?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Wednesday, August 12, 2009 11:39 AM
To: asterisk-users@lists.digium.com
Subject: Re
I have setup my asterisk box using freepbx. I can call extension and make
outbound calls. the outbound calls drop between 10-30sec. we are using
bandwidth.com and they have logged our call. below is your bad followed by what
they say is a good call. I can't figure out where the problem is on
to both to
use their latest firmware.
I know it's not a definitive answer but I've never truly got down to the
heart of the issue as with us it would affect just one out of 100 or so
extensions.
Ish
Ott Rose wrote:
I have setup my asterisk box using freepbx. I can call extension
-c, 3-c and got failure on each of the 9
calls? And then replicated on the “good” call (1-a,2-a…)?
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Wednesday, August 12, 2009
11:08 AM
To:
asterisk-users
? Are there numbers that alternate between good and
bad?
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Ott Rose
Sent: Wednesday, August 12, 2009
11:39 AM
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call
2009, Ott Rose wrote:
I don't think the GUI is editing the conf files correctly. I am not sure
I have configure things right. At this point i think i am going to start
from scratch.
Yea!
--
Thanks in advance
.
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Monday, July 13, 2009 12:02
PM
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
setting up phones
Ok here is what i did.
reinstalled asterisk (i
#.
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Monday, July 13, 2009 12:49
PM
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
setting up phones
added that line to the extensions.conf file because
Of Ott Rose
Sent: Thursday, July 09, 2009 4:12
PM
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
setting up phones
I followed it the best I could.
the phones say no service. I haven't got to setting up the SIP trunk yet I was
told I could get the extensions to work so I could
server.
Date: Thu, 9 Jul 2009 17:42:43 -0400
From: stot...@asteriskhelpdesk.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] setting up phones
On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote:
I followed it the best I could. the phones say
to yes.
Thanks,
Steve
On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose sixfourimp...@hotmail.com wrote:
Here is my physical network.
We have a Adtran router that is plugged into the Asterisk server and into the
circuit provided by my tel co.
the other nic in the Asterisk box is plugged into your
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Friday, July 10, 2009 8:33
AM
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
setting up phones
Here is my physical network.
We have a Adtran router that is plugged into the Asterisk
Totaro
On Fri, Jul 10, 2009 at 10:58 AM, Ott Rose sixfourimp...@hotmail.com wrote:
Carrier is bandwidth.com
we are running Asterisk 1.6.1.1
i ran sip set debug on from the CLI
Once i did a module reload it started displaying all the debuging info. Here is
some of the debug info
--- (13
” to it and register.
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Friday, July 10, 2009 10:33
AM
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
setting up phones
so i filled
asterisk where your
phones can “talk” to it and register.
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Friday, July 10, 2009 10:33
AM
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Friday, July 10, 2009 11:05
AM
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
setting up phones
Great i changed it to my ip here
is the debug and sip show peers. phones still say no service i
-0400
From: stot...@first-notification.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] setting up phones
Change the address in sip.conf, not the phone.
On Fri, Jul 10, 2009 at 12:04 PM, Ott Rose sixfourimp...@hotmail.com wrote:
Great i changed it to my ip here
I don't see my extensions in my extensions.conf file. I see a bunch of other
stuff but nothing that looks like this
exten = 500,500,Dial (SIP/500,20,tr)
I am guessing there should be something in there.
Date: Fri, 10 Jul 2009 12:44:56 -0400
From: stot...@totarotechnologies.com
To:
you dial your 2 extensions and hear MOH until it picks up
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Friday, July 10, 2009 2:39
PM
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
setting up
the GUI.
If you want to mess with the conf files then download the source and compile it
for a vanilla, non-gui installation.
Pick one or the other until you know what you are doing.
Thanks,
Steve Totaro
On Fri, Jul 10, 2009 at 4:11 PM, Ott Rose sixfourimp...@hotmail.com wrote:
added
Can someone tell me how to setup a Aastra 75i phone? I have been trying to set
it up and have pointed it to our asterisk server and selected http for
download. What is the path? I have created two extension in asterisk for
testing. I can't even get the phones to call each other.
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Thursday, July 09, 2009 1:52
PM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] setting
up phones
Can someone tell me how to setup a
Aastra 75i phone? I have been trying to set it up and have pointed
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