On 2006-08-31 19:12:03 -0700, Xue Liangliang [EMAIL PROTECTED] said:
Hi, all.
I have a Adit 3104, and I configured it to work with Asterisk, the
voice quality is quite good, however it just randomly restart, I don't
know whether you guys have the same experience, is it due the firmware
I'm sorry all... But I knew it would take me asking the questions before
the answer would present itself.. I found a reference to the version of
mpg123 needing to be r not s and that was my problem. Had to load
mpg123 from src and fix a few typos in the makefile, but it plays very
nice now.
Hi,
You must have 9001 in voicemail.conf
Extensions.conf should have :
[from-sip[
exten = 9001,1,Ringingexten = 9001,2,Voicemail(u9001)
exten = 9001,3,Hangup
2006/9/1, raviprakash sunkara [EMAIL PROTECTED]:
Hi Users,I'm new to Asterisk, and I'm working with openSER ,
For Call Routing I'm
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Actually it's kind of the opposite... When a call comes in to the FXO,
and it rings the FXS, if the FXS answers on the first ring, the call
goes somewhere but who knows where.
The picking up party hears a dial tone, and the caller
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I tried that image for about 5 minutes. Kept getting errors in asterisk
from the phone and it wouldn't stay registered. Rolled back to 8.0.2
and that works fine for us for now.
Hi Aaron!
When you define all 8 line buttons (on the right
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Does the 8.0.3 image has the same flaws as 8.0.4?
Where did you find 8.0.3 SIP image?
Wasn't even able to register with * at all since
most configuration examples from voip-info.org wouldn't
work...
Do you have any example config for
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
MWI has been working on our (2) 7970's, as far as I can tell. My boss
usually complains when his doesn't work, so it seems to be working fine
as far as that's concerned. The 8.0.4 firmware attempted to register,
but asterisk threw an
Hello,
You should try to setup a proxy/presence server for IM
and presence !
Harry
--- Juraj Bednar [EMAIL PROTECTED] a écrit :
Hello,
I would like to somehow get the presence of IAX2
and SIP users from
Asterisk Manager API or using any other means.
I tried watching for PeerStatus
Hi Mike,
yes we have one and it is working good.
Giorgio Incantalupo
mike wrote:
Dear list,
it is possible to have one quadbri (with only two ports connected) and
one TDM400P card with only one FXO module connected, coexist on the same
machine ?
googling, and voip-infoing (tm) the answer
I've been having some problems with a couple of 7970G so I decided to
factory them. In doing so, it seems to have rendered the phones
unbootable as they (as I understand it) are factory'd with some
version of SCCP code, which makes it constantly looking for
term70.default.loads file which
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I've been having some problems with a couple of 7970G so I decided to
factory them. In doing so, it seems to have rendered the phones
unbootable as they (as I understand it) are factory'd with some
version of SCCP code, which makes
ThanksMarco,
this is my CLI when received a call from number
beginning wwith 0:
-- Executing Macro("VISDN/visdn1.2/76.I", "hangupcall") in new
stack -- Executing ResetCDR("VISDN/visdn1.2/76.I", "w") in
new stack -- Executing NoCDR("VISDN/visdn1.2/76.I", "") in
new stack -- Executing
ThanksMarco,
this is my CLI when received a call from number
beginning wwith 0:
-- Executing Macro("VISDN/visdn1.2/76.I", "hangupcall") in new
stack -- Executing ResetCDR("VISDN/visdn1.2/76.I", "w") in
new stack -- Executing NoCDR("VISDN/visdn1.2/76.I", "") in
new stack -- Executing
Hi
it's possible to create a group of outgoing dial ?
For exemple:
exten = _0.,1,Dial(SIP/voip1/${EXTEN:1},90,rt)
exten = _0.,2,Hangup
exten = _0.,1,Dial(SIP/voip2/${EXTEN:1},90,rt)
exten = _0.,2,Hangup
and when my user call, if voip1 are used, he use voip2
and use not the
Hello *,
is there anybody that has already used the Grandstream configurator
tool? I have to configure a lot of GXP-2000s and BT-102s, and I'd like
to automatize this process.
The configurator doesn't work for me... Probably I don't know how to
make it work. I did the following:
1. Connect the
Hello *,
is there anybody that has already used the Grandstream configurator
tool? I have to configure a lot of GXP-2000s and BT-102s, and I'd like
to automatize this process.
The configurator doesn't work for me... Probably I don't know how to
make it work. I did the following:
1. Connect the
Hi,Please Post your from-trunk context or clever than that create a context in extensions_custom.conf:[justtotest]exten=_X.,1,Answerexten=_X.,2,Playback(vm-goodbye)exten=_X.,3,hangup
and in your [VISDN1.2]context=justtotest .Be aware to write only in extensions_custom.conf , the other
perfect !
thank you very much !
On Fri, 2006-09-01 at 09:12 +0200, Giorgio Incantalupo wrote:
Hi Mike,
yes we have one and it is working good.
Giorgio Incantalupo
mike wrote:
Dear list,
it is possible to have one quadbri (with only two ports connected) and
one TDM400P card
Hi friends, Thank you to all for your response and cooperation to me. I have a doubt.We have registered with Teliax and got DID number. We are making calls to USA successfully using your service. But, We are unable to receive incoming calls to our DID. Here I am sending my config files and error
Try putting a comma between the e-mail address and |attach=yes
On September 1, 2006 00:51, Benjamin Jacob wrote:
Tim and guys,
The sendmail daemon was indeed down. So i turned it on, but messages not
going thru(maybe some sendmail config, will investigate on that)
I do see entries in
On Sep 1, 2006, at 3:36 AM, Jason Lixfeld wrote:
I've been having some problems with a couple of 7970G so I decided
to factory them. In doing so, it seems to have rendered the phones
unbootable as they (as I understand it) are factory'd with some
version of SCCP code, which makes it
I know every second counts in a real 911
situation, but what about adding a pause in the call flow. Maybe a 1 second
pause before actually passing the digits to the provider. This gives the user 1
second to realize the mistake and hang up, longer than 1 seconds is a real
emergency.
hi,
i want try Grandstream GXV-3000 video part. i'm looking for GXV users.
i have asterisk-trunk available.
please contact me privately (or at jabber:[EMAIL PROTECTED])
---
Marek Cervenka
===
Hi Marco,
my from-trunk context in extensions.conf
is:
[from-trunk]
include = from-pstn
[from-pstn]include =
from-pstn-custom
; create this context in extensions_custom.conf to include
customizationsinclude = ext-findmefollow; MODIFICATOIN (PL)
for findmefollow if enabled, should be
Hi Marco,
my from-trunk context in extensions.conf
is:
[from-trunk]
include = from-pstn
[from-pstn]include =
from-pstn-custom
; create this context in extensions_custom.conf to include
customizationsinclude = ext-findmefollow; MODIFICATOIN (PL)
for findmefollow if enabled, should be
I have enabled outside extension '911' and '11' for emergency
service. This way users can either dial '9911' or '911' to get to a
PSAP. I would rather have a couple accidential 911 calls than a
death because someone forgot to dial a 2nd 9. When people are
freaking out they fall back
Hello everybody,
are there any particolar guidelines to follow in order to make Asterisk
act as a SER client?
I see that asterisk fails to register, while X-Lite registers
correctly. Looking at the packets, it seems that the major difference
is that X-Lite packets include the display name in the
Andrea Spadaccini wrote:
Hello *,
is there anybody that has already used the Grandstream configurator
tool? I have to configure a lot of GXP-2000s and BT-102s, and I'd like
to automatize this process.
The configurator doesn't work for me... Probably I don't know how to
make it work. I did the
I have a question on configuration of SPA3000 with asterisk.
1. I want all incoming calls are redirected from SPA3000 to my
asterisk server.
2. Asterisk then should direct this call to my SIP phones (including
Sipura)
3. In case asterisk server is down I want that call
add this to [justtotest]exten=s,1,Answerexten=s,2,Playback(vm-goodbye)exten=s,3,hangupreply your results and asterisk cli
On 9/1/06, Andrea infoteam [EMAIL PROTECTED] wrote:
Hi Marco,
my from-trunk context in extensions.conf
is:
[from-trunk]
include = from-pstn
[from-pstn]include =
Ciao Dave,
I have used the config tool with a BT-102 that I have. I am using
HTTP provisioning. My procedure was as follows:
[cut]
It might be possible to make this work more streamlined by using
tftp, but grandstream uses some extensions to the basic tftp server
included with the linux
Hi
how can i do balance anouncement by using asterisk
take example, i have table balance , user name 9, balance 200$
user dial *98 or what ever, then i need anouce his balance is 200$, by reading from that row
any clues how can i achive this or is this possible ?
Ram
HiIt's possible to define one MHO for calls that are wainting in the queue and another different MOH to calls that are waiting to be tranfer?Thanks Esteban
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
I play a recording that starts as soon as the second 1 is pressed:
If this is an emergency, please hang up and dial 9-911.
Short, simple, and to the point.
--
Strom Carlson
http://www.stromcarlson.com/
___
--Bandwidth and Colocation provided by
Try register=user:[EMAIL PROTECTED]/user
David Hindmarsh
vCall Communications
M: 04111 72 66 7
E: [EMAIL PROTECTED]
W: www.vcall.com.au
GPO Box 1658, Sydney NSW 2001
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrea
On Friday September 01 2006 9:27 am, ram wrote:
Hi
how can i do balance anouncement by using asterisk
take example, i have table balance , user name 9, balance 200$
user dial *98 or what ever, then i need anouce his balance is 200$, by
reading from that row
any clues how can i achive
When SendDTMF is used on a channel, which party is the DTMF being sent
to, the callee or the caller?
What is the syntax for using SendDTMF in an AGI command?
F Church
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
I have been thinking of sending a page when asterisk dials 911.
beep Attention, 911 has been dialed by dialing-extension whose callerID is
callerID.
I think an email would also be appropriate for HR/facility records.
Does anyone know how to send an email from the dialplan?
I assume I would
We used some way back (a year ago) when they first came out. Had
several issues which they were very helpful in working with us on.
They resolved many, had to upgrade and load patches. Unfortunately
they were lacking a couple features we required so they have been
replaced.
Give tech
Andrea Spadaccini wrote:
Ciao Dave,
I have used the config tool with a BT-102 that I have. I am using
HTTP provisioning. My procedure was as follows:
[cut]
It might be possible to make this work more streamlined by using
tftp, but grandstream uses some extensions to the basic tftp server
On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote:
I have a question on configuration of SPA3000 with asterisk.
1. I want all incoming calls are redirected from SPA3000 to my
asterisk server.
2. Asterisk then should direct this call to my SIP phones (including
Hi all,I've just installed vim70, looking for vim syntax highlighting( for Asterisk.conf files) , http://voip-info.org/tiki-index.php?page=vim+syntax+highlighting
, and i notice that both: asterisk.vim and filetype.vim
already refer asterisk configurations.But unfortunately i couldn't get yet the
Hi Avi,
I had a similar problem. Have extension 405 put the call on hold (after
the transfer) and then off hold. I am willing to bet it will bring back
the audio stream. I posted something similar a few weeks ago and if
anyone thought it was a bug, to let me know what information I needed to
The correct syntax would be sip show agents or sip show agent (agentname)
bp
On 8/31/06, Delca [EMAIL PROTECTED] wrote:
Hi, i'm using Asterisk 1.2.9.1 and i'm needing the AGENT functionsince i need something to offer the agents a way to check if they are
logged in or not. i was specting to use
Yea, but even with that you can't run a macro from the CLI. You'd need the manager API to do this.
bp
On 8/30/06, Tim St. Pierre [EMAIL PROTECTED] wrote:
Sort of.There is a command line argument to the asterisk process that runsit's arguments as CLI commands.You could write a shell script that
exten = 0.,1,Dial(SIP/Voip1/${EXTEN:1}SIP/voip2/${EXTEN:1},90,rt)
exten = 0.,2,hangup
bp
On 9/1/06, Noc Phibee [EMAIL PROTECTED] wrote:
Hiit's possible to create a group of outgoing dial ?For exemple:exten = _0.,1,Dial(SIP/voip1/${EXTEN:1},90,rt)
exten = _0.,2,Hangupexten =
Hi
thanks for the quick reply
any documents to read to achive this
or any examples would be great to read
Ram
On 9/1/06, John Millican [EMAIL PROTECTED] wrote:
On Friday September 01 2006 9:27 am, ram wrote: Hi how can i do balance anouncement by using asterisk
take example, i have table
Steven wrote:
I have been thinking of sending a page when asterisk dials 911.
beep Attention, 911 has been dialed by dialing-extension whose callerID is
callerID.
I think an email would also be appropriate for HR/facility records.
Does anyone know how to send an email from the dialplan?
I
Hi marco,
this is my cli when i receive a call beginning with 0,i have done two
tests:
First test the cli is:
-- Executing Answer("VISDN/visdn1.2/10.I", "") in new
stack -- Executing Playback("VISDN/visdn1.2/10.I",
"vm-goodbye") in new stack -- Playing 'vm-goodbye'
(language 'it') --
My 3000 does this natively without config.
Kevin Collins
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy
Sent: Friday, September 01, 2006 10:03 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sipura SPA3000
On Fri,
On Friday September 01 2006 10:19 am, ram wrote:
Hi
thanks for the quick reply
any documents to read to achive this
or any examples would be great to read
Ram
On 9/1/06, John Millican [EMAIL PROTECTED] wrote:
On Friday September 01 2006 9:27 am, ram wrote:
Hi
how can i do
Hi
I've got the exact same problem on Fedora Core 5 with vim70.
Best regards,
Henrik Woffinden
Marco Mouta wrote:
Hi all,
I've just installed vim70, looking for vim syntax highlighting( for
Asterisk.conf files) ,
http://voip-info.org/tiki-index.php?page=vim+syntax+highlighting
I once worked for a big accounting firm who eliminated this problem very
simply -- they used 7 to get a trunk. 7911 and 911 would still get you
an emergency operator, but accidental 911 calls were all but a thing of
the past.
Aaron Daniel wrote:
On Wed, 2006-08-30 at 20:10 -0700, George
I just verified it here as well. Running Asterisk 1.2.11 and two polycom
phones running 1.6.7 firmware with canreinvite=yes. Putting the call on
hold and then off does bring the audio back. From what I can tell by
looking at the lights on my switch, something is still sending the RTP
traffic
Asterisk is the least of your problems here. You first need to talk to
your country's telephone operator and ask if it's possible to get a
toll-free prefix or area code. IF they can, I'm sure they'll charge you
handsomely for that privilege -- initially, per month, per call and per
minute.
Steve Kennedy wrote:
On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote:
I have a question on configuration of SPA3000 with asterisk.
1. I want all incoming calls are redirected from SPA3000 to my
asterisk server.
2. Asterisk then should direct this call to my SIP phones
Steven wrote:
Does anyone know how to send an email from the dialplan?
Use the System() command to launch your script before the dial. Also,
append an ampersand sign to the end of the script name so it will detach
from the script and not hold up the dial command.
Doug
--
Ben Franklin
Andrea Spadaccini wrote:
are there any particolar guidelines to follow in order to make Asterisk
act as a SER client?
No. I have the following config:
register = account:[EMAIL PROTECTED]/asterisk-extension
and
[ser-out]
type=peer
secret=fump
host=serbox.com
callerid=MyMyselfAndi 123456
hi
reading http://www.digium.com/en/docs/misc/compatibility_notes.php, I
see that some rather cool servers from IBM are listed unsupported.
can someone please explain what's so bad about these?
roy
--
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
(+47) 98013356
---
In space, loud sounds, like
Just get one of the number-lookup deals and install my cid-rewrite
script -- http://muware.com/asterisk. Has been working quite well for me.
Robert DeVries wrote:
They are a bit on the expensive side - I'm looking for something along
the usual 2-3 cents per minute.
On 8/31/06,
Does/could the command interrupt call flow? Or does it get fired off and not
wait for it to finish.
I know calls to agi scripts will wait for them to finish. (unless this is
optional)
--
--
Steven
http://www.glimasoutheast.org
Dave Fullerton [EMAIL PROTECTED] wrote in message news:[EMAIL
Hi,
I have a strange problem that started some few days back. Every so often my
asterisk (version Asterisk 1.2.11) comes up with an error and either does a
segmentation fault or dumps core, but then it restarts automatically. Here is
the error when that happens:
Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in:
http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22
By default, if my asterisk went down after the SPA3000 was already
registered, the in-bound PSTN call was lost. I probably did not wait
HiCalls from Cellphones and from landline phones arriving into your asterisk using the same PSTN provider? Or do you have GSM gateway?It seems to me your problem is you are not receiving any DID or MSNs when call comes from landline (i mean start with 0)
M.On 9/1/06, Andrea infoteam [EMAIL
I suppose I could use TrySystem()
http://www.voip-info.org/wiki/view/Asterisk+cmd+TrySystem with a command that
spawns its own shell
to send the email.
Therefore it would always return 0 and be as fast as possible in the dialplan.
My concern is that any priority jumping, out timeouts will
Doh, I should have thought of that one.
Thanks, I think I will throw in some emails with my 911 calls. ( of course,
tested with diff exten. first.)
--
--
Steven
http://www.glimasoutheast.org
Doug Lytle [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Steven wrote:
Does anyone
On 1-Sep-06, at 3:41 AM, Tomislav Parčina wrote:
In article [EMAIL PROTECTED], jason
[EMAIL PROTECTED] says...
I've been having some problems with a couple of 7970G so I decided to
factory them. In doing so, it seems to have rendered the phones
unbootable as they (as I understand it) are
Hi,Please read bellow:On 9/1/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi friends, Thank you to all for your response and cooperation to me. I have a doubt.We have registered with Teliax and got DID number. We are making calls to USA successfully using your service. But, We are unable to receive
Hi
iam trying like this in my extension.conf
some one refered in the news group past
error in messages
Sep 1 21:31:42 WARNING[28610] file.c: File Goodbye does not exist in any formatSep 1 21:31:42 WARNING[28610] file.c: Unable to open Goodbye (format ulaw): No such file or directorySep 1
Does anyone knows what could be the cause for asterisk not listening in post 5060 if SIP interfaces is loaded with no problems?I am using Fedora Core 3.I have followed the instructions in several tutorials and tried several soft phones and the SIP interface seem to be dead.1. When
Is there a card that supports analog DID trunks, alosi known
as ILT trunks or Incoming Loop Trunk. They work by providing talk battery to
the CO, incoming calls happen by pulling loop sending a wink accepting the DID
dtmf digits for the station being called.
Jonn
Pardon me for jumping into the middle of the thread,
but how does one actually test 911 (or 9911)?
I called Verizon, and after speaking with four different departments and
supervisors, they said to just call 911 and tell them that I am going to
be calling them to test. (Well, duh, too late, I'd
Calls from Cellphones and from landline phones arriving into my asterisk
using the same PSTN provider.I don't have a GSM gateway.
I post my reports of call where i see DID, it contain caller numbers without
0 (asterisk do it):
CALLDATECHANNELSOURCE
Elpidio Ramos wrote:
Does anyone knows what could be the cause for asterisk not listening
in post 5060 if SIP interfaces is loaded with no problems?
I am using Fedora Core 3.
I have followed the instructions in several tutorials and tried
several soft phones and the SIP interface seem to be
At 02:16 AM 9/1/2006, you wrote:
it's possible to create a group of outgoing dial ?
For exemple:
exten = _0.,1,Dial(SIP/voip1/${EXTEN:1},90,rt)
exten = _0.,2,Hangup
exten = _0.,1,Dial(SIP/voip2/${EXTEN:1},90,rt)
exten = _0.,2,Hangup
and when my user call, if voip1 are used, he
which softphone?On 9/1/06, Elpidio Ramos [EMAIL PROTECTED] wrote:
Does anyone knows what could be the cause for asterisk not listening in post 5060 if SIP interfaces is loaded with no problems?I am using Fedora Core 3.I have followed the instructions in several tutorials and tried several
Jim Rice wrote:
Pardon me for jumping into the middle of the thread,
but how does one actually test 911 (or 9911)?
I called Verizon, and after speaking with four different departments and
supervisors, they said to just call 911 and tell them that I am going to
be calling them to test.
Its telling you the sound file Goodbye
does not exist in the directory it looks for sounds. If you indeed have a sound
file called Goodbye then you need to either move it to the default sounds
directory or add the path line to the command. If you dont have the
sound file youll need to
If pstn call ring thru line 1 is enabled, all incoming pstn calls will
ring through to the fxs port (and not to asterisk). The OP was looking
for a auto fail over function that essentially would be pstn call ring
thru line 1 on sip failure. That doesn't exist.
Bob Chiodini wrote:
Probably
I am not up to speed on networking in linnux/fedora core 3. How can I verify if I have iptables turned on or the port 5060 is blocked?ThanksSteven Ringwald [EMAIL PROTECTED] wrote: Elpidio Ramos wrote: Does anyone knows what could be the cause for asterisk not listening in post 5060 if SIP
I have tried X-Lite, Express Talk, snom 360 and sipXphone I assume it may be a configuration problem in my linux boxElpidio Marco Mouta [EMAIL PROTECTED] wrote: which softphone? On 9/1/06, Elpidio Ramos [EMAIL PROTECTED] wrote: Does anyone knows what could be the cause for asterisk
Jim Rice wrote:
Is there a proper protocol or procedure? I didn't want the local
sheriff showing up to verify that it wasn't a real emergency...
Depends on your area, you may want to call the local law enforcement
offices and ask.
Doug
--
Ben Franklin quote:
Those who would give up
Jeremiah Millay wrote:
For those of you out there with SPA-942s in production, do any of you
have issue with the sound quality when using g.729 as the preferred
codec? We are noticing terrible sound quality on the other end of a
call made to a spa-942. Everything sounds crystal clear at the
Elpidio,
Is it truly not listening or is maybe a firewall blocking port 5060.
What does netstat -an | grep 5060 tell you? I get this:
netstat -an | grep 5060
udp0 0 0.0.0.0:50600.0.0.0:*
iptables -L will list any firewall restrictions.
Bob...
On Fri, 2006-09-01
Derek Whitten wrote:
Jim Rice wrote:
Pardon me for jumping into the middle of the thread,
but how does one actually test 911 (or 9911)?
I saw a while back some people were testing 911 and they would say something to
the effect
of i am a pbx technician and i am testing the system and
Bob,I get the same answer you get when using netstat -anWhen I query the firewall rules I get this: Chain RH-Firewall-1-INPUT (2 references)target prot opt source destination ACCEPT all -- anywhere anywhere ACCEPT icmp -- anywhere anywhere icmp any ACCEPT ipv6-crypt-- anywhere
anywhere
Hi
the intention of writing that line in to extension.conf to read my balance
when iam dialing 888, read caller id get balance from the table
how can i achive this
is this not possible calling from extension directly than writing some small
agi code, as people recomended before
Ram
On 9/1/06,
Yup.. that's all we do. The other thing to do.. is call the
NON-EMERGENCY number first and check with the dispatcher to make sure
they aren't overloaded with 911 calls... then call in. Around here in
Lycoming County, PA.. the dispatchers are more then happy to test if
they are having a
Jim Rice wrote:
PS: Someone posted this earlier. Is this a valid configuration?
exten = _911,1,Dial(Zap/1/911)
exten = _9911,1,Dial(Zap/1/911)
The _ are unneeded because you don't need to use pattern matching.
I suspect you wanted to match 911 and 9911 exactly, because you didn't
include
Elpidio Ramos wrote:
I am not up to speed on networking in linnux/fedora core 3.
How can I verify if I have iptables turned on or the port 5060 is blocked?
Thanks
*//*
/*iptables -A INPUT -s x.x.x.x -i eth1 -d x.x.x.x -p UDP --dport 5060
-j ACCEPT
*/
Check and see if you're 1) actually
Not sure where you got your SIP image, but my SIP files have that particular file in it.
On 9/1/06, Jason Lixfeld [EMAIL PROTECTED] wrote:
On 1-Sep-06, at 3:41 AM, Tomislav Parčina wrote: In article
[EMAIL PROTECTED], jason +lists.asterisk@lixfeld.ca says... I've been having some problems with a
I call once a year unless I make a mojor dialplan change or telco change, then
I call immed. after I am done.
I call 911 and tell them it is a test and I want to verify what location they
see.
Then I let them know that I need to call once more (where I test 9911).
911 oper. may say to call the
Depends on your area, you may want to call the local law enforcement offices
and ask.
Which may be Police, Fire or other; depending on your area's 911 service setup.
They will let you know the first time you call 911 to test.
--
--
Steven
http://www.glimasoutheast.org
Doug Lytle [EMAIL
Elpidio Ramos wrote:
I am not up to speed on networking in linnux/fedora core 3.
How can I verify if I have iptables turned on or the port 5060 is blocked?
/etc/init.d/iptables status
if you see a bunch of ports, make sure that one of them lists 5060 as
ACCEPT, and that the RTP ports
Elpidio Ramos wrote:
Bob,
I get the same answer you get when using netstat -an
When I query the firewall rules I get this:
Chain RH-Firewall-1-INPUT (2 references)
target prot opt source destination
ACCEPT all -- anywhere anywhere
Hi,I am Chandra. I have a doubt related to ports. I have seen my port 5060 status with nmap command and it is showing that 5060 blocked. Afterthat, I stopped firewall also. After stopping the firewall also, it is showing the 5060 port is blocked. Can I need to restart the linux system from boot to
Westany, the Asterisk voice experts, announce their first Italian voice for
the Asterisk PBX. Liona, an Italian female voice, is the latest addition to
Westany¹s growing catalogue of proven, meticulously-crafted voice prompt¹
suites for Asterisk, Freepbx, trixbox, Bicomsystems and Amp.
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Hello all :-)
I am having some problems with my asterisk callback server with mobile
telephones.
When someone from Norway or Sweden call the asterisk server, and it calls
them back, everything works ok using voipjet and ipcb.net.
But when someone from Thailand or Philipines try to call, it
I think all anywhere should allow 5060. Try running service iptables
stop (as root) to shutdown the firewall. See if 5060 then answers.
I'm not running a firewall on my asterisk box so I'm not sure what the
rule would need to be. service iptables start will restore the firewall.
Bob...
I am
looking at CTPX's VP2000 product. I haven't tried it
yet.
Please
let me know if you find a solution that works.
Tim
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Jonn R
TaylorSent: Friday, September 01, 2006 12:15 PMTo:
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