Hello,
I'm using Asterisk with an ISDN30e PRI line (only 16 channels active).
Every now and then I get a CONGESTION error even-though there are only
1 or 2 channels in use out of the 16 at that time.
When this happens, the user just needs to re-dial and the call goes
through OK.
On a SNOM phone
That's not actually true. SER is very much alive and well and under
constant development.
How do I KNOW it's constant development (other than the chatter on the
mailing list)? Because things keep changing in CVS, but there never
seems to be a 'release' version. Just a release candidate. ;)
when i wnat to working with asterisk realtime and mysql
for any context i have to insert (switch = Realtiem/[EMAIL PROTECTED])
statment into extensions.conf
for example if i want to have 10 context,i have to insert these lines into
extension.conf :
[context1]
switch = Realtiem/[EMAIL
when i wnat to working with realtime and mysql
for any context i have to insert (switch = Realtiem/[EMAIL PROTECTED])
statment into extensions.conf
for example if i want to have 10 context, i have to insert these lines into
extension.conf :
[context1]
switch = Realtiem/[EMAIL PROTECTED]
On Tue, Oct 21, 2008 at 1:46 PM, morteza kashani [EMAIL PROTECTED] wrote:
when i wnat to working with realtime and mysql
for any context i have to insert (switch = Realtiem/[EMAIL PROTECTED])
statment into extensions.conf
for example if i want to have 10 context, i have to insert these lines
Hi Peter,
Thailand is similar to China, except for two things.
- Some places require that billing pulses be generated.
- There may be places using DTMF instead of MFC.
The first issue is definitely the case. The second might just be the
false reporting of issues. People definitely used
SIP wrote:
Seriously, though... this seems to be a popular misconception. I hear it
a lot. Where did you come across the information that SER is no longer
developed?
That seems to be a consequence of looking at the releases.
Anyway, I spoke too soon in saying that there's absolutely
Hello Rodolfo,
I see you have experience with Panasonic, and I have a new challenge of
integrating Asterisk in an enterprice where they have a KX-TDE200 without
the VoIP card so they can' t have voip with the pana-PBX, and that's why
they want *, so do you have any advices for me :) ? I need to
Alex Balashov wrote:
SIP wrote:
Seriously, though... this seems to be a popular misconception. I hear it
a lot. Where did you come across the information that SER is no longer
developed?
That seems to be a consequence of looking at the releases.
Anyway, I spoke too soon in
I am now using a Cisco phone for the second extension (102). I am able
to contact 102 from 101 but not the other way around. The error seems
less severe now:
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-0825b118,
SIP/101/20) in new stack
== Using SIP RTP CoS
SIP wrote:
I see a lot of parellels there with OpenSIPS and SER. OpenSIPS is a
stable plaform that has dozens of modules and documentation galore on
how to mesh the system with this, that, and the other. SER has
rock-solid, incredibly innovative core code, but prefers to leave the
writing
On Tue, Oct 21, 2008 at 9:56 AM, Juan Rodríguez [EMAIL PROTECTED] wrote:
Try changing:
exten = 101,1,Dial(SIP/101/20)
to
exten = 101,1,Dial(SIP/101|20) or exten = 101,1,Dial(SIP/101,20)
because exten = 101,1,Dial(SIP/101/20) means you are trying to contact ext.
20 on through a trunk called
On Monday 20 October 2008 20:42:30 jordan pan wrote:
I have encountered a hard problem that when i connect my anology phone
to channelbank ,I found that i dial a number and create the call,then ,I
hangup the call,but ,very quickly,I listen the ringing im my phone,I pick
it up ,and found it
Hi, I use Asterisk-1.2.26 (with Trixbox-2.1.12) and Portech MV-370 and my
problem is that when I try to call an external mobile phone via Portech I
have alway busy and in log file:
Called Portech/348xxx -- Got SIP response 486 Busy Here back from
192.168.1.2-- SIP/Portech-086e5ee0 is busy ==
Hi Duncan,
when my Cisco phone is started I don't view nothing in my tftp logs, in
other words when cisco phone startup it don't call my tftp server for to try
search configuration files.
Regards.
--
Salvatore.
- Original Message -
From: Duncan Turnbull [EMAIL PROTECTED]
To:
Sasa,
It sounds like you need to set the dhcp option for 'next-server' in order to
point the phone to the ip address of the tftp server. Alternatively, if the
load on the phone is new enough, there is an 'alternate tftp server' setting
that you can point to the ip address of your tftp server
when i wnat to working with realtime and mysql
for any context i have to insert (switch = Realtiem/
[EMAIL PROTECTED]) statment into extensions.conf
See http://bugs.digium.com/view.php?id=6019
It looks like this will be available in Asterisk 1.6.1. You can try
out the beta if you would
Hi,
We are processing lots of calls and I want to filter these that have
incomplete numbers sent
with a proper SIP response. These numbers are not in the local dialplan
by themselves, so
I'm trying to find a way to generate 484 Address Incomplete SIP
response based on the
length of the
Am Dienstag, den 21.10.2008, 06:54 -0600 schrieb César García:
Hello Rodolfo,
I see you have experience with Panasonic, and I have a new challenge
of integrating Asterisk in an enterprice where they have a KX-TDE200
without the VoIP card so they can' t have voip with the pana-PBX,
and
I'm trying to find a way to generate 484 Address Incomplete SIP
response based on the
length of the extension called.
See the allowoverlap option in sip.conf. It should cause a 484 to be
sent if the address is potentially matchable, but not yet matching.
Hi,
Since I'm using ubuntu 8.04.1 and asterisk 1.4.22 I cannot have color in
console.
Do I miss a package or compilation option ?
Thanks
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asterisk-users mailing list
To
- Armand Fumal [EMAIL PROTECTED] wrote:
Since I'm using ubuntu 8.04.1 and asterisk 1.4.22 I cannot have color
in console.
Do I miss a package or compilation option ?
Edit your terminal foreground and background color scheme to white on black
-Dwayne.
Hi all,
I know when doing a Dial, when the called party hangs up, we have a few
different ways to redirect the calling party to other parts of the
dialplan.
In this case, I have someone who would like to do the opposite... When
the calling party hangs up after a Dial(), redirect the called party
I haven't got any reply on this question. I did some more googling but still
couldn't find what I am looking for. Does anybody know if this is possible
at all?
Zeeshan
On Fri, Oct 17, 2008 at 2:12 AM, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Hi everybody,
How can we add new contexts in
On Tuesday 21 October 2008 11:43:31 Armand Fumal wrote:
Since I'm using ubuntu 8.04.1 and asterisk 1.4.22 I cannot have color in
console. Do I miss a package or compilation option ?
Neither. The startup scripts changed slightly, such that they no longer
output to a terminal. Because there is
Martin Smith wrote:
Hi all,
I know when doing a Dial, when the called party hangs up, we have a few
different ways to redirect the calling party to other parts of the
dialplan.
In this case, I have someone who would like to do the opposite... When
the calling party hangs up after a
How can we add new contexts in asterisk realtime module? All I could
figure out after googling is that a new context HAS to be declared
in extensions.conf with 'switch = Realtime/@databasetable' under
the context name declaration. This works fine as long as we are
adding extensions
What about zfone project???
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam Tam
Sent: Tuesday, October 21, 2008 12:49 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How Secure Is Asterisk
There are no
The problem is in asterisk -r
My terminal (ssh client) handle color correctly but with asterisk in verbose
level 3 for debug, I have not color, it is difficult to read.
Does is the startup script also ?
-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de
Install a T1 between the Panasonic and Asterisk and program the T1 in the
Panasonic as a other custom PBX. VOIP card would be the best.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodolfo Alcazar
Portillo
Sent: Tuesday, October 21, 2008 10:55
It's not 100% secure. Like any dual-key encryption, it's subject to the
classic man-in-the-middle attack. This is why the Windows Zfone app has
the addition of a visual key you can read and coordinate with the
recipient to determine if a MITM attack is occurring. But only if you
know what you're
I'm trying to send CallerID info to a MetaSwitch system over a PRI. The
MetaSwitch gets the info exactly as it is sent by Asterisk, but I think
it might be having trouble with the Hexadecimal b1 that is being sent
just before the first character of the CallerID Name.
Does anyone know what the
On Tuesday 21 October 2008 13:33:42 Armand Fumal wrote:
The problem is in asterisk -r
There is no such problem. All 'asterisk -r' does is to take output from
a pipe and sends it to your terminal. It is the core Asterisk process which
is failing to generate terminal color codes.
My terminal
The visual eky in Windows Zfone is good, but it can be broken too. There
is no way to be 100% secure. Nothing is 100% secure. What you need to
do is a standard business C/B analysis and based on that determine how
much security is good enough.
SIP wrote:
It's not 100% secure. Like any
On Oct 20, 2008, at 12:01 PM, Steve Anness wrote:
I am sure this has been discussed prior, however, I am sitting here
and being asked this very question by my superiors. They are loving
what I have done with our two Asterisk servers here; however, they
keep asking me if it is secure or
I'm wondering if the SIP header join, RFC 3911, is supported in the
asterisk stack?
--
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
___
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chan_sip.c's sip_options[] array o' struct cfsip_options says:
/* RFC3911: SIP Join header support */
{ SIP_OPT_JOIN, NOT_SUPPORTED, join },
Bruce Reeves wrote:
I'm wondering if the SIP header join, RFC 3911, is supported in the
asterisk stack?
--
Alex
I had seen that and figured as much. Thanks Alex.
On Tue, Oct 21, 2008 at 5:18 PM, Alex Balashov
[EMAIL PROTECTED] wrote:
chan_sip.c's sip_options[] array o' struct cfsip_options says:
/* RFC3911: SIP Join header support */
{ SIP_OPT_JOIN, NOT_SUPPORTED, join },
On 10/21/08, Eric Chamberlain [EMAIL PROTECTED] wrote:
..snip..
Yes, it's possible to encrypt
voice traffic between SIP phones, but there is no standard that works across
vendors.
..snip..
Incorrect. SIP TLS/SDES/SRTP works across
Cisco/Polycom/Snom/FreeSWITCH/SER family/and more. Asterisk
Kristian Kielhofner wrote:
Asterisk has experimental support for TLS and I know SDES/SRTP is on the
roadmap.
According to the chatter and heated discussions on asterisk-dev, I'd say
TCP/TLS is fairly untested and potentially very broken and will remain
that way through at least 1.6.1.
I also tried downgrading to version 1.4-current but that didn't help.
Oh, typo, but that still didn't cure it
Successful call from from 101 to 102
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-08220318,
SIP/102,20) in new stack
== Using SIP RTP CoS
On 10/21/08, RE Kushner List Account [EMAIL PROTECTED] wrote:
According to the chatter and heated discussions on asterisk-dev, I'd say
TCP/TLS is fairly untested and potentially very broken and will remain
that way through at least 1.6.1.
-Ron
I'm on asterisk-dev too. That's why I said
Hi,
Can someone tell me what causes asterisk to Auto-congest a phone on
a SIP channel?
Is it just a lag issue to the phone or is there something else going on?
Thanks.
-- James
___
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need asterisk tech to relocate to riyadh
this is a permanent position
arabic speaking not necessaryurdu and tagalog ok
ENGLISH required
shukran, shukria, salamat
_
When your life is on the go—take your life with you.
hi;
I'm curious, is there a way, after i set up an admin ivr one needs a
password to enter, to make for say
option 7
for say give me current bandwith usage information? Let me tell you
what i do now and you could give me your ideas.
First of all i have a user, mbabcock that i ssh with, I'm
On Tue, Oct 21, 2008 at 04:39:14PM -0400, Singer Wang wrote:
The visual eky in Windows Zfone is good, but it can be broken too. There
is no way to be 100% secure. Nothing is 100% secure. What you need to
do is a standard business C/B analysis and based on that determine how
much security
Hi Tilghman,
Thanks for you beautiful reply,but i can not understading the
sentences, you bounce the receiver in the cradle,and creating a possible
second leg,can you tell me the principle,and how to solve it, Thanks in
advance.
--
Best regards!
jordan pan
Location:Shenzhen China
hi
for any context ,you must to open /etc/asterisk/extensions.conf and insert this
line : exten =Realtime/[EMAIL PROTECTED]
and (reload) or (restart now) your asterisk
From: Zeeshan Zakaria [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
The flash hook essentially puts the first call on hold while the second is
dialed. try hanging up a bit longer to see what happens.
Eric
On Tue, Oct 21, 2008 at 10:23 PM, jordan pan [EMAIL PROTECTED] wrote:
Hi Tilghman,
Thanks for you beautiful reply,but i can not understading the
yes my problem is like your problem but i am tring to :
http://bugs.digium.com/view.php?id=6019
http://downloads.digium.com/pub/asterisk/asterisk-1.6.1-beta1.tar.gz
From: Terry Wilson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
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