On Thu, 3 Sep 2009, Asterisk Security Team wrote:
++
| Discussion | A lot of time was spent trying to come up with a way to |
|| resolve this issue in a way that was completely backwards |
|
3 sep 2009 kl. 11.40 skrev Michiel van Baak:
On 14:24, Thu 03 Sep 09, ABBAS SHAKEEL wrote:
Hello
Previous context :- After Looking up sip and IAX2 that require
configuration at router level which may cause some problems like
connection
break etc... so i left them . and start
4 sep 2009 kl. 00.44 skrev Matt Riddell:
On 4/09/09 10:41 AM, Doug Lytle wrote:
Todd Routhier wrote:
Trying to do something like this in the sip.conf under my incoming
provider profiles:
setvar=CDR(accountcode)=${EXTEN}
Set(CDR(accountcode)=${EXTEN})
Nah he's trying to do it in
On Thu, 3 Sep 2009, Tilghman Lesher wrote:
On Thursday 03 September 2009 02:47:05 Armin Schindler wrote:
Hello,
I try to move our asterisk installation (3 Asterisk servers in different
offices connected using IAX and a lot of SIP phones, as well as ISDN
connections using CAPI) to use G.722
Thanks Michiel and OLLE ..
I will try and let you know later what is the result
On Fri, Sep 4, 2009 at 11:16 AM, Olle E. Johansson o...@edvina.net wrote:
3 sep 2009 kl. 11.40 skrev Michiel van Baak:
On 14:24, Thu 03 Sep 09, ABBAS SHAKEEL wrote:
Hello
Previous context :- After Looking
4 sep 2009 kl. 08.05 skrev Gordon Henderson:
On Thu, 3 Sep 2009, Asterisk Security Team wrote:
+
+
| Discussion | A lot of time was spent trying to come up with a
way to |
|| resolve this issue
From an off-list comment, I think the explanation is :
- some apps are opening and closing log files before and after each writing,
- some are leaving log files open.
When a log file is currently rotated, both apps can't append anything
anymore to log files.
Apps like Asterisk with wide open log
Hi matt
I use Asterisk release 1.6.1. So if it's my version which is source of
problem could you suggest an application like web-meetme to manage asterisk
conferences ?
Regards
Harry
2009/9/3 Matt Riddell li...@venturevoip.com
On 4/09/09 3:24 AM, harry R wrote:
Hi everybody
I have a
Hello,
I'm experiencing a weird problem when using the VoiceMailMain application.
If I use the application after dialing a Local channel, there's strange beep
just after asterisk answers the call and before the first locution. The
extensions.conf I'm using is:
Ruido extraño al llamar a la
Hi.
I've been using the Asterisk Manager Interface to originate calls from
Console/dsp.
I get the following form the server.
Response: Success
Message: Originate successfully queued
Uniqueid: asterisk-3301-1252055630.26701
Event: Newchannel
Privilege: call,all
Channel: Console/dsp
2009/9/4 Olivier oza-4...@myamail.com:
From an off-list comment, I think the explanation is :
- some apps are opening and closing log files before and after each writing,
- some are leaving log files open.
When a log file is currently rotated, both apps can't append anything
anymore to log
On 4 Sep 2009, at 10:36, Guillaume Yziquel wrote:
Uniqueid: asterisk-1252055630.26702
I'm really wondering how the Uniqueid works. Why is it incremented?
What
is the dot for in the Uniqueid?
The bit before the dot is a unix timestamp (Fri, 04 Sep 2009 09:13:50
GMT in this case). The bit
Steve Howes a écrit :
On 4 Sep 2009, at 10:36, Guillaume Yziquel wrote:
Uniqueid: asterisk-1252055630.26702
I'm really wondering how the Uniqueid works. Why is it incremented? What
is the dot for in the Uniqueid?
The bit before the dot is a unix timestamp (Fri, 04 Sep 2009 09:13:50
GMT
Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through
which I connected an external PSTN line. I use it as carrier for VoIP
calls. I can make successfully calls, but there's one problem, I receive
200 OK with SDP with delay (sometimes more than 30 seconds).
So when I make a call
Description
* Amatix Office is a complete communication platform for small and medium-sized
businesses. It provides email, calendar, contacts, conventional telephony like
ISDN, new generation VoIP telephony like SIP and more other features needed to
power a business.
* Amatix Office is very
What is the license for this product?
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX
Can you post the [bridgecall] context so we can see what it is trying to do?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faheem
Sent: Friday, September 04, 2009 12:32 AM
To: Asterisk Users Mailing List - Non-Commercial
Hello,
In the run up to Astricon [ http://astricon.net ] we'll be talking
today to Tim Panton about his experiences with SfA. You're welcome to
join in!
Speaking of Tim, you can join the conference in W I D E B A N D at
http://api.phonefromhere.com/gateway/zdx.xsql?conference=200901 - come
Well I tried Doug's suggestion and the echo is now better, but when I
call an outside analog line I still get some echo. I can hear my voice
in the ear piece of the phone with a slight delay. Is this just a fact
of life with VoIP, or is there a better way to reduce line echo?
Again, for
Hi all,
our asterisk is connected to a sip proxy through a sip trunk. Let's say we
have following dial plan (only an example)
[from_sip_proxy]
exten = 36122512,1,Answer()
exten = 36122512,2,VoiceMailMain()
exten = 3612252,1,Answer()
exten = 3612252,2,MeetMe(313,MI)
exten = 3612252,3,HangUp()
You can do this through your dial plan by using the failed extension within
your context.
Add this to your dial plan:
Exten = failed,1,NoCDR()
When using a call file and the call fails, the call will jump to the failed
extension within your dial plan. Any post-processing that you want to do
On 4 Sep 2009, at 10:36, Guillaume Yziquel wrote:
Uniqueid: asterisk-1252055630.26702
I'm really wondering how the Uniqueid works. Why is it incremented?
What is the dot for in the Uniqueid?
On Fri, 4 Sep 2009, Steve Howes wrote:
The bit before the dot is a unix timestamp (Fri, 04 Sep
Hello,
Just in case someone hasn't upgraded yet, and is using IAX2.
Original Message
Subject:AST-2009-006: IAX2 Call Number Resource Exhaustion
Date: Thu, 03 Sep 2009 17:47:35 -0500
From: Asterisk Security Team secur...@asterisk.org
To:
Hi All,
I am using asterisk-1.4.22 with mysql as my storage medium for voice
messages.Right now i am running 700+ extensions with this setup .
*System Configuration:*
asterisk-1.4.22.1
(using odbc storage for voice messages)
unixODBC-2.2.11-7.1
mysql-connector-odbc-3.51.12-2.2
4 sep 2009 kl. 13.40 skrev Marius Ciorecan:
Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through
which I connected an external PSTN line. I use it as carrier for VoIP
calls. I can make successfully calls, but there's one problem, I
receive
200 OK with SDP with delay
4 sep 2009 kl. 14.02 skrev Amatisoft SRL:
Description
* Amatix Office is a complete communication platform for small and
medium-sized businesses. It provides email, calendar, contacts,
conventional telephony like ISDN, new generation VoIP telephony like
SIP and more other features
4 sep 2009 kl. 16.05 skrev David Budny:
You can do this through your dial plan by using the “failed”
extension within your context.
Add this to your dial plan:
Exten = failed,1,NoCDR()
When using a call file and the call fails, the call will jump to the
failed extension within your
MURALI V escribió:
Hi All,
I am using asterisk-1.4.22 with mysql as my storage medium for
voice messages.Right now i am running 700+ extensions with this setup .
*System Configuration:*
asterisk-1.4.22.1
(using odbc storage for voice messages)
unixODBC-2.2.11-7.1
im confused:
http://www.h-online.com/open/Starfish-PBX-Management-Tool-for-Asterisk--/news/114154
whats that now? cloned - not cloned?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix,
tom wrote:
im confused:
http://www.h-online.com/open/Starfish-PBX-Management-Tool-for-Asterisk--/news/114154
whats that now? cloned - not cloned?
It specifically says it's not a clone, however I can't see any way in which
they
justify that statement.
To me, it appears to be a clone of
yeah that and good luck with their license .. moving on nothing new to see
here
On Sat, Sep 5, 2009 at 12:01 AM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
tom wrote:
im confused:
http://www.h-online.com/open/Starfish-PBX-Management-Tool-for-Asterisk--/news/114154
whats that now?
Hello everybody,
I try to install -Ubuntu 8.04 server- a B410P and a TDM2400P together
with Asterisk 1.4.26-2, dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0.
Problem I face is the following one:
CLI module load chan_dahdi.so
== Registered application 'DAHDISendKeypadFacility'
== Registered
Hi,
I've just had to enable the requirecalltoken=no option in iax.conf for
one of my IAX2 trunks, and I don't think it works properly in the
realtime version. I've created the requirecalltoken field in my
(Postgres via ODBC) database, type text, and have variously tried it
with 'yes', 'no' and
Jason Baker wrote:
Well I tried Doug's suggestion and the echo is now better, but when I
call an outside analog line I still get some echo. I can hear my voice
in the ear piece of the phone with a slight delay. Is this just a fact
of life with VoIP, or is there a better way to reduce line
On Fri, 4 Sep 2009, Leif Madsen wrote:
tom wrote:
im confused:
http://www.h-online.com/open/Starfish-PBX-Management-Tool-for-Asterisk--/news/114154
whats that now? cloned - not cloned?
It specifically says it's not a clone, however I can't see any way in which
they
justify that
On Friday 04 September 2009 12:08:26 Gary Hawkins wrote:
I've just had to enable the requirecalltoken=no option in iax.conf for
one of my IAX2 trunks, and I don't think it works properly in the
realtime version. I've created the requirecalltoken field in my
(Postgres via ODBC) database, type
Administrator TOOTAI wrote:
I try to install -Ubuntu 8.04 server- a B410P and a TDM2400P together
with Asterisk 1.4.26-2, dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0.
snip
Why signalling bri_cpe_ptmp is not recognized (for tests, pri_cpe is
loading well, bri_cpe or bri_net gaves also
Hi All,
on this hardware:
http://www.sun.com/servers/x64/x2200/specs.xml
would one of the following 4 ports PRI cards be ok ?
http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE420BF-T1/E1-PRI-PCIe-HW-EC.html
Hi all
I have asterisk 1.4.12 that was working on CCM 4.0
they updated to 6.1.3 and it no longer works.
I tried updating to 1.4.26.2 but still not working.
I get SIP error 503 service unavailable.
The guy says he has MTP enabled etc...
Anyone connected to CCM 6.1.3 and have it working?
No it's not a fact of life. VoIP works as fine as conventional telephony once
it's correctly set up.
Try echocancel=256 instead of echocancel=yes and also run fxotune (check the
man page). If that all fails, install OSLEC. It's an excellent free software
echo canceller, that works much better
Oops sorry, I didn't read you have a VPMADT032 module. (Damn ADD)
Don't bother setting echocancel=256, echocancel=yes should be working fine.
Also don't bother with OSLEC. Maybe the IP part of the call is introducing the
delay that is being perceived as echo. Do you have QoS set up to minimize
2009/9/4 Vinícius Fontes vinic...@canall.com.br
No it's not a fact of life. VoIP works as fine as conventional telephony
once it's correctly set up.
Try echocancel=256 instead of echocancel=yes and also run fxotune (check
the man page). If that all fails, install OSLEC. It's an excellent
Yes sir. Even upgraded to Cisco UC 7.0 and still works great with
Asterisk 1.4.23.
For me, I only had to change one thing. On the Cisco side, confirm the
SIP Trunk Security Profile has the Incoming Transport Type as TCP+UDP
and Outgoing Transport Type as UDP.
On the Asterisk side, a basic
Yes sir. Even upgraded to Cisco UC 7.0 and still works great with
Asterisk 1.4.23.
For me, I only had to change one thing. On the Cisco side, confirm the
SIP Trunk Security Profile has the Incoming Transport Type as TCP+UDP
and Outgoing Transport Type as UDP.
On the Asterisk side, a
I have a SIP trunk between CCM 6.1.2 and Asterisk 1.6.1.1 working
without any issues. What does your peer section of the sip.conf look
like? When do you get the error (call direction)?
-Jonathan
On Fri, Sep 4, 2009 at 12:00 PM, Jerry Geisge...@pagestation.com wrote:
Hi all
I have asterisk
4 sep 2009 kl. 21.00 skrev Jerry Geis:
Hi all
I have asterisk 1.4.12 that was working on CCM 4.0
they updated to 6.1.3 and it no longer works.
I tried updating to 1.4.26.2 but still not working.
I get SIP error 503 service unavailable.
From Cisco or from Asterisk?
/O
Tilghman Lesher wrote:
On Friday 04 September 2009 12:08:26 Gary Hawkins wrote:
I've just had to enable the requirecalltoken=no option in iax.conf for
one of my IAX2 trunks, and I don't think it works properly in the
realtime version.
[snip]
Please try the attached patch.
I've just tried the
Steve Totaro wrote:
I was also under the impression that rx/tx gain was only for POTS lines.
Our Indianapolis plant had echo on our PRI until I reduced the TX
volume. And, it did work.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
We are using using what Cisco's Port Address Translation, so that all
SIP traffic is done through %EXTERNIP%. To any outside box, it should
look like the asterisk server is actually on %EXTERNIP%.
My SIP packet gets sent to the ITSP with a Call-ID:
2fd557964ca936b1d72f1328c...@%externip% ,
You should also get your t1 carrier to provide echo cans on the
circuit. Fairly common and easy for them to set up.
John Chastain
On Sep 4, 2009, at 3:07 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
2009/9/4 Vinícius Fontes vinic...@canall.com.br
No it's not a fact of life.
Here is my chan_dahdi.conf
[channels]
; configuration for T1 card as PRI
language = en
group = 1
echocancel = yes
echotraining = yes
echocancelwhenbridged=yes
rxgain=-1.0
txgain=-4.0
signalling = pri_cpe
switchtype = 4ess
usecallerid = yes
context = incoming
channel = 1-23
Here is my
worked for me, but would lock up on my 3845:
!Image: Software: C3845-IPVOICE-M, 12.4(19), RELEASE SOFTWARE (fc1)
!Image: Compiled: Sat 01-Mar-08 01:57 by prod_rel_team
!Image: flash:c3845-ipvoice-mz.124-19.bin
stopped locking up my 3845 but rarely properly negotiated T.38:
!Image: Software:
Title: Jason Baker - Signature
I am using the VPMADT032 echo cancellation module
attached to a Digium
TE121B PCI express card. The incoming phone service is a PRI. Can I
still use fxotune if my incoming lines are all digital?
I will try echocancel=256.
I have also tried OSLEC and still get
Title: Jason Baker - Signature
I have not really messed with the QoS on my network, I guess that will
be my next step. I am running 1 - 3Com Unified Gigabit Wired and
Wireless PoE Switch and several 3Com Baseline Switch 2924-PWR Plus
Switches. I know these switches support VoIP. I guess I'll
Title: Jason Baker - Signature
I have not tried it without any echo cancel. I'll have to try that.
Thank you. I'm not sure about rx and tx gain. I adjusted the values and
it seemed to make a difference.
Jason Baker
IT Coordinator
Glastender,
Inc.
5400 North Michigan Road
Saginaw, Michigan
Title: Jason Baker - Signature
Thank you. I will do that.
Jason Baker
IT Coordinator
Glastender,
Inc.
5400 North Michigan Road
Saginaw, Michigan 48604 USA
Phone: 989.752.4275 ext. 228
Fax: 989.752.4276
www.glastender.com
John wrote:
You should also get your t1 carrier to provide echo
/ Hi all
//
// I have asterisk 1.4.12 that was working on CCM 4.0
// they updated to 6.1.3 and it no longer works.
// I tried updating to 1.4.26.2 but still not working.
//
// I get SIP error 503 service unavailable.
/ From Cisco or from Asterisk?
Calls from CCM to asterisk work.
Calls from
Jason Baker wrote:
#---Glastender PRI--
loadzone = us
defaultzone = us
span = 1,0,0,esf,b8zs
This should be 1,1,0
You get your timing from ATT
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve
On 4 Sep 2009, at 07:53, Armin Schindler wrote:
On Thu, 3 Sep 2009, Tilghman Lesher wrote:
On Thursday 03 September 2009 02:47:05 Armin Schindler wrote:
Hello,
I try to move our asterisk installation (3 Asterisk servers in
different
offices connected using IAX and a lot of SIP phones, as
Title: Jason Baker - Signature
Thank you. I will edit that.
Jason Baker
IT Coordinator
Glastender,
Inc.
5400 North Michigan Road
Saginaw, Michigan 48604 USA
Phone: 989.752.4275 ext. 228
Fax: 989.752.4276
www.glastender.com
Doug Lytle wrote:
Jason Baker wrote:
- Barry Miller asterisk-us...@notanet.net wrote:
On Wed, Sep 02, 2009 at 09:44:05AM -0500, Doug Bailey wrote:
- Barry Miller asterisk-us...@notanet.net wrote:
Hi,
Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840
work
fine.
With 1.6.1.[45] same
- Jason Martin jmar...@metrixmatrix.com wrote:
Hello,
The company I work for recently purchased 2 Rhino CB24s and a Rhino
PCI-E R4T1. The channel banks are plugged into the R4T1, as well as 2
PRIs from our telco. The CB24s are for all internal analog phones.
Most of the phones
Hello All
I am looking for the cable I need to create to connect a TE121 card with a
TN2464BP card (AVAYA ISDN Card), please let me know if someone have the
information about this cable, my asterisk CLI show this:
pri show span 1
Status: In Alarm, Down, Active
And the card is in red,
Juan Cardoza escribi:
Hello All
I am looking for the cable I
need to create
to connect a TE121 card with a TN2464BP card (AVAYA ISDN Card), please
let me
know if someone have the information about this cable, my asterisk CLI
show
this:
pri show span 1
Status: In
On Fri, 4 Sep 2009, Olle E. Johansson wrote:
4 sep 2009 kl. 08.05 skrev Gordon Henderson:
On Thu, 3 Sep 2009, Asterisk Security Team wrote:
+
+
| Discussion | A lot of time was spent trying to come up with a
way to
In this moment I have asterisk rvvv
First I had the alarms below:
Sep 4 17:44:51] NOTICE[19916]: chan_dahdi.c:9915 handle_init_event: Alarm
cleared on channel 1
[Sep 4 17:44:51] NOTICE[19916]: chan_dahdi.c:9915 handle_init_event: Alarm
cleared on channel 2
[Sep 4 17:44:51]
On 4/09/09 6:22 PM, Olle E. Johansson wrote:
4 sep 2009 kl. 00.44 skrev Matt Riddell:
On 4/09/09 10:41 AM, Doug Lytle wrote:
Todd Routhier wrote:
Trying to do something like this in the sip.conf under my incoming
provider profiles:
setvar=CDR(accountcode)=${EXTEN}
On 4/09/09 10:31 PM, Stanisław Pitucha wrote:
2009/9/4 Olivieroza-4...@myamail.com:
From an off-list comment, I think the explanation is :
- some apps are opening and closing log files before and after each writing,
- some are leaving log files open.
When a log file is currently rotated,
On 5/09/09 10:03 AM, Gordon Henderson wrote:
I've been hanging out with IAX, thinking it's the right thing, but more
and more I'm thinking of moving to SIP, and I think this will be the straw
that tips the balance as it were. I've a few 100 boxes out there which
would all eventually need
On 4/09/09 10:31 PM, Stanisław Pitucha wrote:
2009/9/4 Olivieroza-4...@myamail.com:
From an off-list comment, I think the explanation is :
- some apps are opening and closing log files before and after each writing,
- some are leaving log files open.
When a log file is currently rotated, both
On Fri, 4 Sep 2009, Gordon Henderson wrote:
example, if multiple peers use the same authentication details, and
they have not all upgraded to support call token validation, then the
ones that do not support it will get locked out. Once an upgraded
client successfully completes an
Hello,
I have a issue between asterisk and windows based VoIP system (Client).
Vendor SIP Server -- My asterisk -- Client
Here is ethereal trace between asterisk and client.
1 0.00 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE
sip:1978525...@192.168.4.23
72 matches
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