Hi Patrick,
You are in the UK, right (at least DDI strongly suggests that)? This is
the commonest signalling for a DDI line on an analogue pair. The line is
behaving just like the main exchange is a telephone. It picks up the
line, by applying a 600ohm loop, and dials (with pulses per second or
Hello,
Thanks, that helped a bit:
-- Executing Dial("SIP/1011-506f", "modem/g1/V0412463080") in new
stack
-- Couldn't call g1/V0412463080
-- Hungup 'Modem[i4l]/ttyI0'
== Everyone is busy at this time
-- Executing Congestion("SIP/1011-506f", "") in new stack
== Spawn extension
You are missing a $ in front of {EXTEN:1}. It should be ${EXTEN:1}
On Sat, 2003-11-01 at 22:55, Matthew Enger wrote:
> Hello,
>
> I am trying to use 2 netjet cards under asterisk and isdn4linux. I am
> having a hard time trying to get them to work in terms of dial out. Does
> anyone have a worki
Hello,
I am trying to use 2 netjet cards under asterisk and isdn4linux. I am
having a hard time trying to get them to work in terms of dial out. Does
anyone have a working config I could look at for even one card (tried
that, not much luck either).
When i dial out:
-- Accepting AUTHENTICATED
At 05:03 PM 11/1/2003, you wrote:
> Netfinity 4000R
> >servers that do not support X windows under RH8.x and I
> prefer not to go
> >back to RH7.3...
I recall in the archives somewhere, and through someone's post earlier
today, that there is some sort of problem with RH9 with Zaptel (hardware)
dr
as my first project with *, i would like to replace our old
neax2400(sds) with an * server.
i've got an X100p and a TDM400 on hand already.
for the CO lines, the X100p works ok with fxsks signaling though there
are still strange
things happening every now and again but more testing is on the way
There are links to several other Asterisk related sites at the bottom of
the page at http://www.fnords.org/~eric/asterisk/
On Sat, 2003-11-01 at 18:26, Brancaleoni Matteo wrote:
> > Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
> > servers that do not support X windows und
> Netfinity 4000R
> >servers that do not support X windows under RH8.x and I
> prefer not to go
> >back to RH7.3...
I recall in the archives somewhere, and through someone's post earlier
today, that there is some sort of problem with RH9 with Zaptel (hardware)
drivers and that RH8 is preferred
At 05:15 PM 11/1/2003, you wrote:
Apologies if there is a cleanly written and searchable FAQ that I could be
directed to. I have no problem to RTFM if I can find the FM...
Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
servers that do not support X windows under RH8.x and I
On Sat, 1 Nov 2003, Brian West wrote:
> OH great idea... feature request on bugs.digium.com?
>
> On Sat, 1 Nov 2003, John Brown (CV) wrote:
>
> >
> > How does one send a broadcast message to all voice mail boxes?
> >
> > I want to send a single message to every mailbox on the system
> > informi
> Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
> servers that do not support X windows under RH8.x and I prefer not to go
> back to RH7.3...
yes, asterisk under rh 9.0 works good. I have 3 systems running with
that distro.
> BTW, where would I find a useful FM?
http://ww
OH great idea... feature request on bugs.digium.com?
On Sat, 1 Nov 2003, John Brown (CV) wrote:
>
> How does one send a broadcast message to all voice mail boxes?
>
> I want to send a single message to every mailbox on the system
> informing them of changes, etc.
>
> any thoughts ??
>
>
>
>
Apologies if there is a cleanly written and searchable FAQ that I could be
directed to. I have no problem to RTFM if I can find the FM...
Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
servers that do not support X windows under RH8.x and I prefer not to go
back to RH7.3..
How does one send a broadcast message to all voice mail boxes?
I want to send a single message to every mailbox on the system
informing them of changes, etc.
any thoughts ??
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hi fred,
i don't know if this question has been already answered...
i haven't tested it whit asterisk YET, (i have to)
check the following links:
http://luxik.cdi.cz/~devik/qos
http://www.ibiblio.org/pub/Linux/docs/HOWTO/other-formats/html_single/ADSL-Bandwidth-Management-HOWTO.html
and tell
David J Carter wrote:
I have a FWD number and wish to connect it to Asterisk to receive my FWD
calls.
See the Asterisk FAQ at
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
You'll find pointers to several Asterisk - FWD configurations there.
/Olle
I can also confirm chan_h323 and g.729 work well to 5300s, but we had
to build on RH8 not RH9. Haven't tried 5300 to Asterisk except via SIP
which works fine--even to i4l interfaces.
On Friday, October 31, 2003, at 01:57 AM, Jeremy McNamara wrote:
John Todd wrote:
I've done some reviewing of
If it doesn't find a host named ciscoccm1 then it will try to connect to
whatever host it got it's DHCP lease from. (assuming it's using DHCP, of
course)
On Sat, 2003-11-01 at 15:07, Brian West wrote:
> Last I checked skinny firmware would try to connect to a host that would
> resolve to CiscoCM1
Last I checked skinny firmware would try to connect to a host that would
resolve to CiscoCM1
bkw
On Sat, 1 Nov 2003, Ray Burkholder wrote:
> I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm
> a little unsure as to how get the phone to figure out which ip address it
> s
Title: Leterhead
Sip.conf
[general]
register=>FWDNUMBER:[EMAIL PROTECTED]/EXTENSION
[fwd]
type=friend
username=FWDNUMBER
secret=FWDPASSWORD
host=fwd.pulver.com
context=YOURINBOUNDCONTEXT
extensions.conf
[inboundcontext]
exten => EXTENSION,1,Dial(SIP/SOMEOTHEREXT
Title: Making a Skinny phone talk to Asterisk
I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm a little unsure as to how get the phone to figure out which ip address it should register with when it boots.
How do I do that?
I already have a tftp server for my SIP
dear all gurus,
i am looking into setting a fair size conference room system and would
be most
grateful for any advise, experience, recommendation on the following:
1) what is a reasonable real world max. channels conference room that 1
* server
can handle? with what kinda h/w in server?
2) is
Title: Leterhead
You have to input your info in your sip.conf -- it’s in your examples
Checkout this site for examples… www.fnords.org/~eric/asterisk
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J Carter
Sent: Saturday, November
As far as I know they do only SIP. If your Asterisk box is behind a NAT
firewall then you probably will have problems.
> Hi All,
>
> I have a FWD number and wish to connect it to Asterisk to receive my FWD
> calls.
>
> How I do?
>
> Is it a register in sip.conf or iax.conf?
>
>
> Regards
>
> Dave
Title: Leterhead
Hi All,
I have a
FWD number and wish to connect it to Asterisk to receive my FWD calls.
How I do?
Is it a
register in sip.conf or iax.conf?
Regards
Dave
Registered Office: - 23 First Street, Low
Moor, Bradford, West Yorksh
[EMAIL PROTECTED]
>
> > However, voice as heard on X-Lite is just fine from Cisco,
> but voice as
> > heard on Cisco from X-Lite has random silent breaks of one
> or two or
> > three second duration on a very regular basis.
> > Any ideas on how to get rid of the random silent breaks?
>
> "X-Lite" (build 1082
At 10:42 AM 10/31/2003, you wrote:
I have the same problem and it was solved setting:
# Uncomment for aggressive residual echo supression under
# MARK2 echo canceller
#
KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
This creates a very nasty "click" when I talk into the SNOM (but no long
echo!). It's like havin
At 08:54 AM 11/1/2003, you wrote:
P.S.: Looks like I have to post this once a day now.
You should post this (or I'll do it for you, with permission, as I already
have an account) on the Asterisk wiki at www.voip-info.org. You might still
have to post, but at least it will be out there...
Thanks,
At 07:42 AM 11/1/2003, you wrote:
Hi All,
Is it possible to show which line a call has come in on in *.
Yes, absolutely. In asterisk each line is a channel. The channel
information is VITAL to the call and is available (and used) everywhere in
asterisk. Channels look like this: "ZAP/1-1", which
i m interested too?
[EMAIL PROTECTED] wrote:
Any details yet?
-Dan
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At 10:23 PM 10/31/2003, Bryan Nolen wrote:
System execute "asterisk -rx reload"
?
Yes, correct.
--Ernest
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
> Sent: Saturday, 1 November 2003 5:18 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk
Hi!
> in sending you my mgcp.conf file, my ip10s mostly working fine...
Could you explain "mostly" in your sentence, and maybe - if you can -
give quick overview of Grandstream vs. SwissVoice (except for the pending
SIP implementation, of course)?
Thanks, Philipp!
Hi!
> However, voice as heard on X-Lite is just fine from Cisco, but voice as
> heard on Cisco from X-Lite has random silent breaks of one or two or
> three second duration on a very regular basis.
> Any ideas on how to get rid of the random silent breaks?
"X-Lite" (build 1082 and possibly lat
Default User Password is 123
Default Admin Password is 456
-sb
-Original Message-
From: Roman Pelikh [mailto:[EMAIL PROTECTED]
Sent: Friday, October 31, 2003 11:54 PM
To: '[EMAIL PROTECTED]'
Subject: [Asterisk-Users] Polycom Soundpoint IP600
Does anyone have the Admin password for the
Javier Rios wrote:
hello
you can help me with a problem
I have dlink DG-104S already and this registered in asterisk
but not to call... between in ports
you can help with an example the configuration me of
mgcp.conf
extensions.conf
; MGCP Configuration for Asterisk
;
[general]
Title: Message
>By
default X-Lite now has silence supression turned on..
>Go
to Advanced System Settings > Audio Settings > Silence Settings
>>and change Transmit Silence to "Yes"..
I
played with this. Still problems.
Where do I
check for PT 13 or 19?
Could be comfort noice ? Ch
Title: Huge silence breaks between Cisco 7960 phone & X-Lite
I hear
no ring back tone when I place a call using Nikotel
as my outbound provider to a PSTN telephone number. When I call to a Vonage telephone number
I get a ring back tone. Any
suggestions as to why I do not receive ring bac
Could be comfort noice ?
Check for PT 13 or 19
Michael
Ray Burkholder wrote:
Huge silence breaks between Cisco 7960 phone & X-Lite
Does any one else have problems with
huge, random silence breaks between an X-Lite and Cisco 7960 SIP
phone? Both are running g.711. Softphone t
rnc Info Lists wrote:
Hi,
-Original Message-
The portion of extensions.conf is:
exten => 3001,1,Dial(MGCP/aaln1,20)
exten => 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20)
Or aaln/1@ should do just fine. However this doesn't explain why there
is no dialtone on the phone..
Oh, one thought: D
On Saturday 01 November 2003 09:21, Rich Adamson wrote:
> I'm somewhat confused with the caching though. My iax.conf had:
> register => npi:[EMAIL PROTECTED]
> in it (which worked fine prior to their IP change). On the same
> * machine, if I ping iaxtel.com now, the dns resolves to 69.73.19.178
>
Been meaning to ask this for some time... no big deal, but curious.
I have a single register statement in my iax.conf for iaxtel like:
[general]
port=5036
register => npi:[EMAIL PROTECTED]
However, when I restart *, I see:
Registered to '69.73.19.178', who sees us as 205.221.193.101:5036
Hi All,
Is it possible to show which line a call has come in on in *.
My scenario is 8 incoming lines, 6 lines are trunked to one number and the
other 2 are individual lines.
I would like to pass the trunked lines to one set of extensions, and the
other lines to two other set of extensions.
Als
> >I've not been able to register with iaxtel.com for the last couple
> >of days. Is anyone else seeing this, or did I miss something?
>
>
> Same here I have not been able to get any calls nor do any calling
> through them!
Mark indicated yesterday that digium changed the IP address of the
iaxt
Jim,
> Off-premise SIPs are all behind simple NAT routers.
>
> Off-premise SIPs have been able to receive calls from and make calls
> through the PSTN. No problem. Calls between on-premise SIPs, not a problem.
> Calls between off-premise SIPs and any other SIPs connected to the server
> are a p
From: Rich Adamson <[EMAIL PROTECTED]>
>
>I've not been able to register with iaxtel.com for the last couple
>of days. Is anyone else seeing this, or did I miss something?
Same here I have not been able to get any calls nor do any calling through them!
___
James,
You can make mirror of your site at our facilities. To support Asterisk
community we can host mirror of your site, or make it primary hosting
whatever is more convinient for you. We can do it duting this weekend.
Let me know
Alexander
Unofficial Asterisk Forums
*
Look at RTP (/etc/asterisk/rtp.conf) packets, and its firewall configuration.
Jim Greenfield, Computer Troubleshooters Metro NY/NJ wrote:
Our network is connected to a cablemodem using a dynamic DNS service
to resolve our address. The Asterisk server has been alternately set
up behind a NAT router and without a NAT router -- that is, with two
NICs, one of which is provid
Our network is
connected to a cablemodem using a dynamic DNS service to resolve our
address. The Asterisk server has been alternately set up behind a
NAT router and without a NAT router -- that is, with two NICs, one of
which is providing NAT to the rest of the network; the office SIPs are
Found the answer.
It was not codec, but instead missing "[" in "local" context.
Ta
Senad
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Yes, we have!
Nice device, works fine on public IP but behind
NAT it has problems.
PCphoneline are sorting out NAT problems as far as I know.
Ta
Senad
I'll see what I can do to upgrade the speed of www.voip-info.org
Traffic has been going up as it gets more popular.
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: "Michael Wood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, October 31, 2003 6:44 AM
Subjec
Does any one know what below means?
-
DEBUG[6151]: File chan_sip.c, Line 4904 (handle_request): Check for res
for 2298
DEBUG[6151]: File chan_sip.c, Line 973 (find_user): Call from user
'2298' is 1 out of 0
---
Ray Burkholder wrote:
Does any one else have problems with huge, random silence breaks
between an X-Lite and Cisco 7960 SIP phone? Both are running g.711.
Softphone to/from softphone works, softphone to/from iax2 works, iax2
to/.from cisco phone works.
However, voice as heard on X-Lite is j
Hi,
At 05:03 30-10-2003 +0300, you wrote:
== Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))
-- MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate
[chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099
(find_subchannel): Gateway '192.168.0.5' (and thus its end
At 01:43 1-11-2003 +0300, you wrote:
Is msn messenger capable of using asterisk as it's gateway?
Yes, provided you are using MSN 4.7, and not 5.0 or higher. Configure the
Communications Service under the Options/Accounts pane.
Florian
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Any details yet?
-Dan
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At 23:49 31-10-2003 +0100, you wrote:
Hi!
> MGCP works on IP basis, it has no userid's or passwords.
Ouch - that means MGCP and NAT w/ dynamic IP (of the router) is a No-No?
Correct. Use IAX :)
Florian.
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I noticed tonight, when doing a demo of the Directory app, that
something mighty odd is going on.
I have one Zap FXS channel and a SIP channel (Grandstream B101).
When I invoke that app on the Zap phone things work normally.
When I invoke it from the GS phone, the CLI shows that it is playing t
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