--- On Fri, 12/7/12, Steve Totaro stot...@totarotechnologies.com wrote:
Why don't your span numbers match? 1-4 but you have
3-6 in your .conf.
What do you mean?
I have the following:
span=3,1,0,CCS,AMI
span=4,2,0,CCS,AMI
span=5,3,0,CCS,AMI
span=6,4,0,CCS,AMI
The first parameter is the
- Original Message -
From: Matthew Jordan mjor...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, December 08, 2012 12:43 AM
Subject: Re: [asterisk-users] BLF and call-limit in 1.8
Thanks for your reply. I just
Am 10.12.2012 06:37, schrieb Chandrakant Solanki:
Hi All,
OS : CentOS 5 64bit OS Machine
Asterisk: 1.8.13.0
ODBC Packages:
unixODBC-2.2.11-7.1
mysql-connector-odbc-3.51.12-2.2
unixODBC-devel-2.2.11-7.1
res_odbc.conf
[telco-ops]
enabled = yes
dsn = telco-ops
username = dba
password =
It seems I only assumed a call-limit value of 1 in the DB would make call
waiting not work. I tested it now, and because that sets the value in
Asterisk to INT_MAX, a call-limit value of 1 in the DB does allow for call
waiting. The same value in sip.conf does not.
-Pan
- Original Message
/etc/odbc.ini
[telco-ops]
Description = Asterisk realtime and other FUNC_ODBC access
Driver = MySQL
Server = 172.18.100.18
Socket = /var/lib/mysql/data3306/mysql.sock
User= dba
Password= c3podb@2012
Database= mytelcoexample
Port
I finally found the real culprit. The call-limit DB field was mapped to both
call-limit and callcounter in the view asterisk uses. The latter is what
caused the strange behaviour. Removed both and everything works as expected
now.
-Pan
- Original Message -
From: Pan B. Christensen
I am running 11.0.2 from 1.4.43 previous.
When I start up and do a sip show peers all devices are on and show an
IP Address.
After some time sip show peers shows two devices of my 12 as
(Unspecified).
I never had an issue with 1.4.43.
Is there some issue with 11.0.2 and registration?
Jerry
How can extensions.conf be changed to work with both
Asterisk 11 and 1.4.X such that 1.4.X calls deadagi and 11 just calls
agi as deadagi is no more.
Thanks,
jerry
--
_
-- Bandwidth and Colocation Provided by
Put a GLOBAL in extensions.conf with the version and use GOTOIF to run
AGI/DEADAGI dependent on it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, December 10, 2012 10:01 AM
To:
From the last time you sent this to the list, here's the response from Richard
Mudgett rmudg...@digium.com...
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk
jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk
and
other sip
On Mon, Dec 10, 2012 at 5:23 AM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Password= c3podb@2012
In case you didn't realize you were sending this out publicly to a publicly
archived and searchable list, you might want to change that password now.
--
-Chris Harrington
On Mon, Dec 10, 2012 at 8:24 AM, Jerry Geis ge...@pagestation.com wrote:
When I start up and do a sip show peers all devices are on and show an
IP Address.
After some time sip show peers shows two devices of my 12 as
(Unspecified).
When you say two, is it two every time? The same two? Is
When you say two, is it two every time? The same two? Is there something
different about the two that show this behavior? There isn't enough
information in your message.
Yes it is the same two devices every time.
I have the server running 11.0.2 , I have 8 asterisk devices (1.4.43),
I have two
Sounds like a registration timeout issue. What does the sip.conf entry look
like for these?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, December 10, 2012 10:30 AM
To: Asterisk Users Mailing List -
Hello,
On 10.12.2012 18:30, Jerry Geis wrote:
When you say two, is it two every time? The same two? Is there
something different about the two that show this behavior? There
isn't enough information in your message.
Yes it is the same two devices every time.
Try pedantic=no in
On 10 Dec 2012, at 16:13, Christopher Harrington wrote:
On Mon, Dec 10, 2012 at 5:23 AM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Password= c3podb@2012
In case you didn't realize you were sending this out publicly to a publicly
archived and searchable list, you
On Mon, Dec 10, 2012 at 10:52 AM, Steven Howes steve-li...@geekinter.netwrote:
On 10 Dec 2012, at 16:13, Christopher Harrington wrote:
Hostname address is RFC1918, he'll probably be ok ;)
Private subnet or not, that's a social engineering and recon target. If all
it takes is a Google search
I have a non-standard SIP client that I am trying to integrate with an Asterisk
10 server.
This client requires that it register with the Asterisk server and that this
registration not be authenticated.
When a call is passed from Asterisk to the SIP client, the client does require
Asterisk
Here's where I am baffled and I am hoping someone with intricate
knowledge of this implementation may be able to explain it to me. What
we had to do to get this working was to set the host= parameter to the
respective endpoint IP's of the VPN tunnel, 172.10.1.1 in my case, and
172.10.1.2 in
Consider using a sip proxy server such as OpenSIPS or Kamailio.
Regards,
Ali Pey
On Mon, Dec 10, 2012 at 12:59 PM, John Gilbert
jo...@motorolasolutions.comwrote:
I have a non-standard SIP client that I am trying to integrate with an
Asterisk 10 server.
This client requires that it
Try pedantic=no in sip.conf.
Also, enable a SIP debug on the peers, check if anything out of the
ordinary appears.
seems as though pedantic=no was the issue. they are staying online.
further looking (which I seemed to miss) was in 1.4 pedantic as default no,
in 11 default is yes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf
#include siptrunk.conf
siptrunk.conf:
Does each box show up in the others SIP SHOW PEERS?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject:
On Mon, 10 Dec 2012, Jerry Geis wrote:
How can extensions.conf be changed to work with both Asterisk 11 and
1.4.X such that 1.4.X calls deadagi and 11 just calls agi as deadagi is
no more.
On Mon, 10 Dec 2012, Danny Nicholas wrote:
Put a GLOBAL in extensions.conf with the version and use
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Looks like a connectivity issue, doesn't it?
IP of box2, 172.17.145.145, doesn't show up even once in the SIP dialogues.
What happens on box2 (asterisk -vvvr and tcpdump port 5060) in the
moment that you place a call through box1 to box2?
Also what's strange is that you are trying to call
The Asterisk Development Team has announced the release of Asterisk 1.8.19.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.19.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team has announced the release of Asterisk 10.11.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 10.11.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 11.1.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.1.0 resolves several issues reported by the
community and would have not been possible
How can I monitor channel that hangup?
I'm using asterisk 1.8.15.1 and there are many times that nobody is using the
line but when I run:
asterisk -rx core show channels it show:
Channel Location State Application(Data)
SIP/pstn--00 (None)
Hi, Ken
I have almost the same setup as yours: new
asterisk-SIP-Trixbox(Asterisk 1.4)---PRIpots
Here are my configs:
new box sip.conf:
[126]
directmedia=no
type=friend
host=trixbox_IP_addr
secret=my_secret
username=126 ;this is for outgoing calls from new asterisk via trixbox
When a call comes in asterisk records the date correctly but when I cake a call
out I get only something like:
Date: 60
here is an example:
From: 7807560785
To: s
Date: 2012-12-11 00:46:04
Status: ANSWERED
From: 5
To: 4331235
Date: 60
Status: 4
--
Joseph
--
Hi,
Thank you for your reply.
77 ext. number is connected with my asterisk. so any one want to talk with
jitsi(pc), they have to dial 77 then 2000#(jitsi sip user number).
my pbx is sending callerid. i can see on other analog phone display.
Yes pbx is sending callerid. When i dial any ext.
On Mon, 10 Dec 2012, Joseph wrote:
When a call comes in asterisk records the date correctly but when I cake a
call out I get only something like:
Date: 60
here is an example:
From: 7807560785
To: s
Date: 2012-12-11 00:46:04
Status: ANSWERED
From: 5
To: 4331235
On 12/10/12 20:45, Steve Edwards wrote:
On Mon, 10 Dec 2012, Joseph wrote:
When a call comes in asterisk records the date correctly but when I cake a
call out I get only something like:
Date: 60
here is an example:
From: 7807560785
To: s
Date: 2012-12-11 00:46:04
Status: ANSWERED
On 12/10/12 20:45, Steve Edwards wrote:
On Mon, 10 Dec 2012, Joseph wrote:
When a call comes in asterisk records the date correctly but when I cake a
call out I get only something like:
Date: 60
here is an example:
From: 7807560785
To: s
Date: 2012-12-11 00:46:04
Status: ANSWERED
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