1. The phone has not the DND active, i checked it several times
2. Outbound calls always success, the problem is when the phone receive a call,
it repsnds with busy signalling.
3. The firmware i just the lastest one 1.1.5.15 and i cannot upgrade asterisk.
Thanks for all
-Messaggio
Thanks Henry,
anyway the phone is always registered when i get the busy tone
* Name : 502
Secret : Set
MD5Secret: Not set
Context : local
Language : it
FromUser :
FromDomain :
Callgroup: 1 (2)
Pickupgroup : 1 (2)
Mailbox :
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
darren wrote:
An analog DID trunk is a line (typically part of a group) that has a group
of
numbers assigned to it at the telco side. They work in a variety of ways
depending on the telco. One example is the trunks as Telus provides them.
In the UK, to make us match the rest of Europe, it's also possible to
access the Emergency Services on 112.
Again, although a few calls were made around the right time, none of
them were 999 or 112. The I've examined the master.csv for 30 mins
before the Police said the call was made, and can't
Hi Tilghman
As far as I can see from both master.csv and the account log, no number
was dialled beginning 999 (or 112 - both numbers connect to the
Emergency Services, and the Police couldn't tell me which had been
called).
Unfortunately, my Telco (British Telecomsigh) can't tell me exactly
Can I just say I'm grateful for all the replies - this list is
invaluable.
Thanks for the suggestion Razza, I've been back again to the logs and no
call was placed that contained the string 999 or 112 at the right
time!
Glad it made you smile, said it was a fun one for the list. Looking
like
Hi Mark!
13 feb 2008 kl. 23.42 skrev Mark Quitoriano:
Is it possilble for a single context to have multiple host=
something like this
First context is something we use to describe a segment of the
dialplan. I would call this section.
[carrier]
host=ip address1
host=ip address2
I've had the opposite problem. Press mute while the call is still
ringing and it will say MUTE on the display but the microphone is
not muted. It was very embarrassing to discover this bug.
On Wed, Feb 13, 2008 at 2:03 AM, Thomas Kenyon
[EMAIL PROTECTED] wrote:
Lutgring, Sam wrote:
I take it
Hi Amit
OK, the majority of our calls go out via zaptel fxo and pstn lines.
When these are all busy, calls are routed via a VOIP provider here in
the UK. All activity is recorded in our logs, and I can find no trace
of either 999 or 112 (if since been reminded that in the UK, you can now
also
Thanks Michael,
that's a *huge* thing you're telling me, in the wiki details for the
PCI-X bus I've read about retrocompatibility, but I just wanted to be
100% sure. I can go on and order my server, now!
Thanks again
Marco
ps. This proves also the complete unaccuracy of the information
On Thu, 14 Feb 2008, Phil Knighton wrote:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten =
The 481 Call Leg/Transaction Does Not Exist response to the
NOTIFY makes me think that you might need to configure the
phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages
from the phone when it is booted?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hi,
i use a ser, as proxy sip server(authentication), then a cisco router
as sip2h323 gw(authorization and accounting). i want to start asterisk
as sip statefull b2bua server, any suggestion to howto or documentation
to asterisk integration and b2b use?
ty in advance.
--
Riccardo Cupardo
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten = _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
Just
Hi there dear users and dear developers of Asterisk!
I've got a maybe strange idea, let's see if you laugh or think it's reasonable J
I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering
machine).
Each analog line can be reached through a phonenumber, since they are each
On Thu, Feb 14, 2008 at 01:17:45PM +0100, Johan Sandgren wrote:
Hi there dear users and dear developers of Asterisk!
I've got a maybe strange idea, let's see if you laugh or think it's
reasonable J
I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering
machine).
Try switching your DTMF mode on both asterisk and the phone to RFC2833. I have
not seen the issue that you are describing, but I had some very strange issues
like call hang-ups when I was using INFO. After switching the issues were gone
and I have had no further troubles.
Hope this helps
Steve Langstaff wrote:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten =
Hi there dear users and dear developers of Asterisk!
I've got a maybe strange idea, let's see if you laugh or think it's
reasonable J
I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering
machine).
Each analog line can be reached through a phonenumber, since they
Hi Tim,
Imagine the scenario where we had 10x Asterisk servers, with calls
presenting sequentially starting from the first server, then server two,
etc.
If we took down the first server for maintenance with 'asterisk -rx stop
gracefully' we then will block all incoming calls to all servers as
Gordon Henderson wrote:
On Thu, 14 Feb 2008, Phil Knighton wrote:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten =
Steve Langstaff [EMAIL PROTECTED] writes:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten =
On Wed, 13 Feb 2008 22:26:16 -0500, Russell Bryant
[EMAIL PROTECTED] wrote:
The arguments to System() are a bit different. Put it in just like you would
type at the command line.
System(/tmp/netcid.py 2000 Joe)
That did it :-) Thanks guys.
BTW, for those interested, I didn't have to
On Thursday 14 February 2008 03:39:33 Phil Knighton wrote:
OK, the majority of our calls go out via zaptel fxo and pstn lines.
When these are all busy, calls are routed via a VOIP provider here in
the UK. All activity is recorded in our logs, and I can find no trace
of either 999 or 112 (if
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Benny Amorsen
Sent: 14 February 2008 13:57
Steve Langstaff [EMAIL PROTECTED] writes:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN}) exten =
_0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten =
Bill Andersen wrote:
Has anyone tried to used VB6 to communicate with the Asterisk Manager?
If so, would you be willing to share some basic code showing your
approach to getting connected and parsing results?
I've got a Telnet control that is allowing me to connect, authenticate
and see
I had GXP-2000's running on 1.0 versions of asterisk even earlier. So I
know it does work. I upgraded one of my customers GXP's to the latest
firmware in it still works. Can you post the output of the CLI with verbose
set to 99 and the the output from the asterisk log file that has the call
Thank you to all those who replied to my last query. For them and for the
suggestion, I can monitor asterisk using the asterisk -r -x command
option. What I would like to know is that using asterisk -r -x way I can
only use the *CLI commands. Is there any other way in which I can monitor
asterisk?
Thank you to all those who replied to my last query. For them and for the
suggestion, I can monitor asterisk using the asterisk -r -x command
option. What I would like to know is that using asterisk -r -x way I can
only use the *CLI commands. Is there any other way in which I can monitor
asterisk?
Quoting Steve Langstaff [EMAIL PROTECTED]:
The 481 Call Leg/Transaction Does Not Exist response to the
NOTIFY makes me think that you might need to configure the
phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages
from the phone when it is booted?
Yeah, sure. And there are some
Thought I would post this experience to the list so it's archived for
posterity... My company is deploying Asterisk-based PBX's to all of our
branch offices. Each office has 2 analog Voice lines and a fax line.
We didn't want to go to the expense of using TDM400's in the servers
(which run
Soumya Kat wrote:
Thank you to all those who replied to my last query. For them and for
the suggestion, I can monitor asterisk using the asterisk -r -x
command option. What I would like to know is that using asterisk
-r -x way I can only use the *CLI commands. Is there any other way in
Riccardo Cupardo wrote:
Hi,
i use a ser, as proxy sip server(authentication), then a cisco router as
sip2h323 gw(authorization and accounting). i want to start asterisk as
sip statefull b2bua server, any suggestion to howto or documentation to
asterisk integration and b2b use?
Well,
Hello,
I've seen that many solutions concerning asterisk dimensioning and load
balancing involve the use of sip proxy like openser.
Is there any recommended way to balance IAX load?
BRs,
Claudio
Internet Email Confidentiality Footer
Hello,
reading iax2 draft, I'm not sure if the protocol supports peer 2 peer
calls (e.g. like SIP).
If it doesn't, is Asterisk the only server side iax2 implementation?
I also would like to understand if it's possible for asterisk (by means
of some configuration rules) to translate a iax2
I don't know if it would be of any use to you but we have some C# code
that handles the basics of communicating the the Asterisk Manager
Interface. It doesn't do anything fancy just sends single commands and
checks the responses. We don't use it for monitoring.
Regards,
Greyman.
Thanks
Hi,
I have been working with asterisk to make ivr calls (outbound and inbound). I
have the functionality -
Read(variable|file_name)
used in my dialplan. Now i need to pass the variable to my ruby file to compare
the data entered with the database (mysql).
How can i pass the arguments from my
Hi All,
I've been reading up on 1.4 snmp integration. When I try and compile
asterisk with a -with-netsnmp option it complains about net-snmp
installation being broken. However, the net-snmp-devel rpm is installed,
and snmpd on the machine runs fine.
Anyone have a guide for the
On Thursday 14 February 2008 07:55:08 SIP wrote:
Gordon Henderson wrote:
On Thu, 14 Feb 2008, Phil Knighton wrote:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
On 13/02/2008, Raj Jain [EMAIL PROTECTED] wrote:
SIP over TCP is included in 1.6.
http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co
Thanks all! :o)
___
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On Thu, 14 Feb 2008, Brent Davidson wrote:
That 3 foot cable run passes behind a
21 monitor that was connected to the server. When the line tests
showed everything OK, I decided the monitor might be a long shot but I
could understand how the degaussing coil coil could possibly induce a
Working with asterisk 1.4; using the AMI Originate command, it is possible to
do something like:
Variable: CDR(accountcode)123456
Or must the variable names be var[n] where n is a number?
I'd like to set the accountcode for a Local channel that originates a call.
Thanks. -A
--
Anthony -
Hi;
Sorry, I forgot to post the zapata version, it is 1.4
but I do not know the release and I do not know how to
know the exact release.
Regards
Bilal
--
Hi All;
I am facing a problem that the telephon line in
Egypt
does not work with the FXO port at the digium card
(TDM22B), and
Hi, all. I've got a PoS Emagen VM system tied in with our Telrad PBX. I
hate 'em both, but I'm stuck with the Telrad for the time being. That
being said, does anyone know of a way to replace the VM solution with
Asterisk? I'd -love- to get an Asterisk box in the loop, here.
Thanks,
-Ken
Hi;
The PBX located in Egypt at Cairo city.
I am able to receive calls on the FXO ports at 3rd and
4th ports, but I am not able to place outgoing call
(it gives busy tone that coming from the service
provider, or it gives an voice message from the
service provider that the dialed number is
Honestly.. this sounds like a telco issue.I understand what the other
person is saying about the PRI still being technically up... BUT... if the
channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to
the next available channel, which they clearly are not doing.
On Thu,
I considered doing just that, but since I didn't have my scope with me
and it's an hour's drive away it didn't seem worth it at this point. If
we have trouble again I may take the scope down there and test it.
-Brent
Steve Edwards wrote:
On Thu, 14 Feb 2008, Brent Davidson wrote:
That
bilal ghayyad wrote:
[channels]
rxgain=15.0
txgain=15.0
Wow! Is this necessary? Is this something you took from a sample
config somewhere, or numbers that you arrived at through trial and
error? They seem a bit high in my experience, *but* I've never been to
Egypt before, and I sure
Quoting Brent Davidson [EMAIL PROTECTED]:
I considered doing just that, but since I didn't have my scope with me
and it's an hour's drive away it didn't seem worth it at this point.
If we have trouble again I may take the scope down there and test it.
unless the cable is in the same spot
--- Fons van der Beek [EMAIL PROTECTED]
wrote:
What phone do you use?
Linksys ?
SIP softphones and Alcatel analog phones behind ATA
gateways (Grandstream). However, I'm having a hard
time reproducing the problem. It doesn't happen often.
That does sound like what is happening.. Telco knows channel 1-23 are not
busy (so far as they are concerned), however.. so far as you are concerned,
they are busy.. so telco sends the call down... but the equipment doesn't
take it.
I would *think* the Telco could keep trying channels down the
I think the problem is that the telco presents the call on a specific
channel, then zaptel tells it that the channel is busy.
We need to be able to tell the telco that each unused channel on a given
span is unavailable, and it will determine that the others are in use and
will present the call
Hi list,
I have been experiencing a strange behavior with asterisk and i would
like to know if someone else has face it.
This is my scenario,
3 extensions created on sip.conf: 121 | 123 | 123
Everything work just perfect except for the following issue:
I have this block on my
I had the same problem some time ago...
You got to install also this packages:
net-snmp-devel
newt-devel
lm_sensors-devel
bzip2-devel
That should do it!
Regards,
Ricardo Carvalho.
On Thu, Feb 14, 2008 at 5:30 PM, Adrian Marsh [EMAIL PROTECTED]
wrote:
Hi All,
I've been reading up on
Steve Langstaff [EMAIL PROTECTED] writes:
Oops! Yes, I see that now - my fault for confusing Asterisk pattern
matching with RFC3435 pattern matching. Sorry.
Unfortunately inventing a new regex syntax seems to be a favourite
pastime.
Perhaps it would be possible to allow exten = /00.*/,Dial...
Matthew J. Roth [EMAIL PROTECTED] writes:
Yes, asterisk -rx will only allow you to execute CLI commands. It
also tends to spew out a bunch of garbage that makes parsing difficult
unless verbosity is always set to 0.
It would be very handy if it was possible to turn off messages that
aren't
On Thu, 2008-02-14 at 22:32 +0100, Benny Amorsen wrote:
Perhaps it would be possible to allow exten = /00.*/,Dial... It might
cause problems with the ex-GF syntax. Another starting character could
mean RFC3435 pattern matching.
I've been suggesting that for about four years now (long before I
Hi,
i've just finished setting up gtalk connection with asterisk. it works
nice, audio is full duplex.
i just have one question which i could not find an exact answer to. Is
gtalk able to send dtmf codes? Because i'd like to listen to my
voicemails while away from home.
I've been googling
If you want to flush your disk cache to see how much memory is being eaten
cache pages, try this:
echo 3 /proc/sys/vm/drop_caches
- ast erisk [EMAIL PROTECTED] wrote:
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
If you take Asterisk down, the PRI should go down as the D channel is
down. Then the telco should KNOW that there is trouble with the PRI and
those channels are in trouble busy and not availible. If the telco
still tries to push a call to a channel on a PRI that is down, then the
telco is at
Ricardo Carvalho wrote:
I had the same problem some time ago...
You got to install also this packages:
net-snmp-devel
newt-devel
lm_sensors-devel
bzip2-devel
That should do it!
Why would this depend on newt? net-snmp and lm-sensor headers and
libraries make sense. newt doesn't make
Correct me if I'm wrong, but as I understand it your issue is that when
you give Asterisk the stop gracefully command it waits until all
active calls have finished before it takes the ISDN down but gives busy
signals to new incoming calls on idle channels. If this is the case
then it would
Maybe you'r right and newt isn't really necessary. I just read somewhere
that those dependencies were needed, I've installed them and it worked...
Try to only install the other ones and if res_snmp gets compiled without it,
great!
Regards,
Ricardo Carvalho.
On Fri, Feb 15, 2008 at 12:01 AM,
Andrew wants to take the system down softly-there are active calls on some
channels. He doesn't want to accept additional calls on the idle channels.
He can't take the D channel down without disruption to the active calls.
--Don
Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651
On Thu, Feb 14, 2008 at 10:12 AM, Henry Devito [EMAIL PROTECTED] wrote:
I had GXP-2000's running on 1.0 versions of asterisk even earlier. So I
know it does work. I upgraded one of my customers GXP's to the latest
I'm not sure you are right, since I have had Polycoms that didn't
work, it's
I'm struggling to get my dialplan to work with a simple analog fax
machine.
I have TDM400B zaptel card with an FXO and FXS port. I have the FXO
port connected to the POTS machine and the FAX machine connected to the
FXS port.
The FAX machine itself works fine, I can FAX outgoing messages
We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30) and
the BLF indicators no longer function.
Has anyone had a similar issue? And a solution?
PaulH
___
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Just wondering how your experience is with HPEC,
Is it just for analog interfaces or we can use it on TE122 as well?
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options
Hi,
Since the original codec negotiation patch (
http://bugs.digium.com/view.php?id=4825 report) just closed yesterday,
and as well as my report (http://bugs.digium.com/view.php?id=11998), I had
nothing to do but send my patches to the list.
It might be good if my patches are placed at
Always rely on free -m to see how much free memory you have not top.
in terms of memory leak, i have asterisk running on servers with uptime of
400 days (CentOs), if there was any leak, i'm guessing i would have crashed
server long time ago.
On Thu, Feb 14, 2008 at 4:23 PM, Doug Bailey [EMAIL
On Thu, Feb 14, 2008 at 8:38 PM, Al lists [EMAIL PROTECTED] wrote:
Always rely on free -m to see how much free memory you have not top.
You could install and use htop - it's a much more functional (and
informative) version of top. It shows the difference between
shared/buffer/cache memory.
Al lists wrote:
Always rely on free -m to see how much free memory you have not top.
in terms of memory leak, i have asterisk running on servers with
uptime of 400 days (CentOs), if there was any leak, i'm guessing i
would have crashed server long time ago.
On Thu, Feb 14, 2008 at 4:23 PM,
On Thu, Feb 14, 2008 at 09:32:04PM -0600, Erik Anderson wrote:
On Thu, Feb 14, 2008 at 8:38 PM, Al lists [EMAIL PROTECTED] wrote:
Always rely on free -m to see how much free memory you have not top.
You could install and use htop - it's a much more functional (and
informative) version of
On Thu, Feb 14, 2008 at 9:37 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
It also consumes more CPU.
True, a fraction more. If you have that little overhead on your
server, though, that this would cause a problem, you probably should
upgrade your hardware, IMHO.
-eriik
2008/2/13, Johansson Olle E [EMAIL PROTECTED]:
In SIP, there's a specification for how I as a domain owner can
request all calls to my domain to use
SIP/TLS by using DNS NAPTR and SRV records.
Which one ?
Does it also deal with SPIT ?
But how do I as a caller
request a secure service?
I
2008/2/13, Atis Lezdins [EMAIL PROTECTED]:
On 2/13/08, Rob Hillis [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
If it is being removed in 1.6, I'm a little concerned since there's no
mention of this when you show the application, nor on voip-info.org
.
Hi,
I am using asterisk-1.4.15
I have a queue with one agent added using AddQueueMember
(FAO|Local/[EMAIL PROTECTED]|0||Agent/602).
Once this command executes queue show FAO shows:
FAO has 0 calls (max unlimited) in 'roundrobin' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within
The other day my asterisk local SIP clients got hung when my provider had a
DNS failure. All registrations went dead (even the ones that were IP
addresses) and all sip peers went offline. I know this was know problem at
one point is there any update on this when using a FQDN for one of the peer
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