I ended up writing a basic parsing script that lets me search the full log,
based on some unique identifier (eg, my own extension "vlog 2027"). It then
digs out the associated A*k log number for each line that's it, and lists them
out. Then I choose the 'call' and it re-filters by that call only
Ok,
This is something to do with folder layouts.
I have:
/var/lib/asterisk/sounds - uk files
/var/lib/asterisk/sounds/digits -uk/us digits
/var/lib/asterisk/sounds/jp - Japanese files
/var/lib/asterisk/sounds/jp/digits - Japanese digits
I read the 1.4 notes on :
http://www.voip-
8633] logger.c: --
Playing 'vm-dialout' (language 'jp')
[2012-08-24 11:33:34] VERBOSE[18633] logger.c: == Spawn extension
(TokyoReception, 9222, 13) exited non-zero on 'SIP/XXX-0005'
Thanks,
Adrian
-Original Message-
From
Hi Chris, Thanks for replying,
I've got it set in the context in extensions.conf:
[TokyoReception]
exten => s,1(TOKYORECEPTION),Answer
exten => s,n,Set(CHANNEL(language)=jp) ; set japanese by default
exten => s,n,SET(LOOP=0)
exten => s,n,SET(LANG=JP)
It could be something fixed between 1
Hi Guys,
I've a few questions around languages I'm on 1.4.18 (old yes I know, but
upgradings not an option just yet).
I've downloaded the gsm Japanese files from
ftp://ftp.voip-info.jp/asterisk/sounds/ and put them in place
I've found that when I switch to jp, and play some of my own voice
Discussion
Cc: Adrian Marsh
Subject: Re: [asterisk-users] Meetme and MOH
Adrian Marsh wrote:
Thanks all,
I realised after posting 2 things.. 1) I needed to also cover MOH
outside of meetme. And that 2) theres a bug in 1.4.18 where the
defaults aren't reloaded properly for MOH, an
Thanks all,
I realised after posting 2 things.. 1) I needed to also cover MOH
outside of meetme. And that 2) theres a bug in 1.4.18 where the
defaults aren't reloaded properly for MOH, and you have to do a server
stop/start to get them to reload.
Thanks,
Adrian
--
___
Hi Gary,
I went through this process a few times over the past few years.
Theres a few short guides for securing Asterisk, but much of it depends
on your design. If it's a traditional POTs-type PBX then locking down
IPs using firewalls is a great thing, however if you make use of
inbound-SIP
Hi,
I'm testing an upgrade from 1.4.18 to 1.4.37 in a VM prior to putting it
into production.
Ive done this by installing 1.4.18 onto the VM, putting my config files
in place and then installing 1.4.37 over the top (which is what I'd have
to do on production).
I've found a few issues in the c
Hi,
With a dynamic Meetme using: MeetMe(|DsMrc)
How do I control which context MOH uses, other than "default" ?
Asterisk: 1.4.15
Thanks,
Adrian
--
_
-- Bandwidth and Colocation Provided by http://www.api-d
Hi,
I'm trying to create a link between two PBXs. One is Asterisk 1.4.15,
the other is an unknown 3rd party PBX.
In my internal testing, beween two A*k servers, I found that if I
created two sip accounts from the same IP, one as peer and one as user
(intending to give an -IN and -OUT setup
How odd...
If I specify the host=dynamic then it goes to the wrong context.
If I specify the host=192.168.50.132, then it goes to the correct
context.
If I don't specify the host at all, then it also goes to the correct
context... (but then of course I can't use that account for outbound
cal
Hi,
Running 1.4.15. I've a SIP user as below. My default context in
sip.conf is [incomming_pstn]
I'm having trouble with inbound calls going to the wrong context.
[test-ubi]
username=test-ubi
type=friend
secret=XXX
host=dynamic
canreinvite=no
context=testinbound
nat=yes
a
Hi,
Am running 1.4.18 at the moment, and am trying to implement inline blind
transfer.
I have :
[featuremap]
blindxfer => *6 ; Blind transfer
in features.conf
And in extensions .conf under [globals] :
DYNAMIC_FEATURES=automon#blindxfr
So what am I missing ??
Anyone have any experience with a Japanese local VoIP termination
supplier?
I've emailed a few companies looking to setup some PSTN to SIP and SIP
to PSTN termination, but no luck so far.
Thanks,
Adrian
--
_
-- Ban
It says in the readme from that link you provided:
" This patch adds AMR-NB support to Asterisk 1.4
(for Asterisk 1.6 check out asterisk 1.6 branch and use the
asterisk-1.6-AMR.patch patch (provided by Ivelin Ivanov))"
Did you use the 1.6 branch and patch ??
I'll have t
Hi James,
Thanks for the help. 3.10 registers into my SIP server just as a normal SIP
client.
Yes, qualify=yes. I just tried setting that to no on my end, and I still get
the message. I'll try turning it off on 3.10 too tomorrow and capture some
trace too
Adrian
> Hi All,
>
>
>
> I've two
Hi All,
I've two asterisk servers on the same LAN, both 1.4, and I keep getting
"Got SIP response 489 "Bad event" back from 192.168.3.10"
No idea whats causing it. The only references I can find mentions NATing
issues, but these are on the same LAN so NAT shouldn't be an issue.
3.10 does auth
Hello,
I'm looking for some advice on securing Asterisk.
Recently my servers been under several brute-force SIP attacks.
I have several remote sites, as well as many roaming users, who may have
PC softclients and/or SIP based hardphones.
My first step will be to strengthen the password
Anyone have any idea on how to force marker bits on in RTP ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: 10 June 2009 14:50
To: Asterisk Users Mailing List - Non-Commercial
(resend as apparently I was blocked)
Hi All,
I'm looking for how to enable SIP Markers, or specifically, how to have
the TIME reset when a call route changes.
I'm debugging an issue, where a sip client we have switching to
one-way-audio, when an asterisk server fruther down the call
Hi All,
I'm looking for how to enable SIP Markers, or specifically, how to have
the TIME reset when a call route changes.
I'm debugging an issue, where a sip client we have switching to
one-way-audio, when an asterisk server fruther down the call path dials
out to the PSTN. Scenario is:
users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 16:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug
On 2 Jun 2009, at 14:14, Adrian Marsh wrote:
> I'
e else is on
the phone, without QOS, they are really going to be fighting for that
bandwidth.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: Tuesday, June 02, 2009 9:59 AM
To: Asterisk Users Ma
this.
On 06/02/2009 08:58 AM, Adrian Marsh wrote:
> Hi,
>
> It's a 2mb dedicated leased fibre line, with<50% utilisation.
> My first thoughts were the internet link, but that wouldn't explain
why
> the client transmit (other channel), which is on the same LAN as
Scratch that, my inventory tool says the system has 256Mb not 1Gb.
I wonder if a memory upgrade would help it out...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: 02 June 2009 14:59
To
al Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: Tuesday, June 02, 2009 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug
Hi,
Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 14:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug
On 2 Jun 2009, at 1
Hi All,
I've a 1.4.15 A*k server supporting several users (approx 80 total, but
<10 sim calls usually). I've one user who complains of intermittent bad
calls, though I suspect the bad calls are across the board, but
intermittent.
Inbound calls are via in IAX trunk from Gradwell. CPU stats
I'd like to see that link too!
I use Cisco 7940s at the moment, and would like to see how to hook them into AD
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews
Sent: 26 May 2009 15:56
To: Asterisk
Noone can give me a clue on this ?
How Domains are used within Asterisk ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: 26 May 2009 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [
Noone can give me a clue on this ?
How Domains are used within Asterisk ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: 26 May 2009 12:14
To: Asterisk Users Mailing List - Non
Hi,
I'm trying to understand an issue I'm seeing between two Asterisk
servers. I think it has to do with Domain definitions.
Server A), has extension 5550 defined. Has a sip client 2000 defined,
and has guest-invites enabled.
Server B), Dials to server A for any 5550 dialled. Has sip clie
Hi,
I've a few working asterisk servers, all seeing the same symptom, but
they are all based on the same configs.
A SIP inbound INVITE message is coming in to an extension (not a peer)
eg 5...@ourserver.com
A tcpdump clearly shows the INVITE coming in, but asterisk seems to be
ignorin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: Wednesday, May 13, 2009 8:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Proxying from one server to another
Hi All,
I'm trying to find a software p
Hi All,
I'm trying to find a software package to do the following sip proxy
work:
I've an A*k server A that needs to be decommissioned, from the USA, and
replaced by server B, in the UK. Both servers are on public internet
IPs.
Whilst the client migration happens, I want to divert all the
anyone has any pointers.
Adrian
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: 07 May 2009 09:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [aste
Hi All,
Looking to gauge some opinions on redirect/proxy software.
I've two existing A*k servers out on the 'net. I need to redirect the
traffic going to those two servers, over to a new 3rd one.
Unfortunately, when the servers and clients were built, they used
hardcoded IPs, rather th
Lesher
Sent: 07 May 2009 15:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Understanding Codecs
On Thursday 07 May 2009 03:33:14 Adrian Marsh wrote:
> So where are the codec translations set?
I assume you're talking about the numbers within t
translations set?
Thanks
Adrian
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: 06 May 2009 18:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
A2s on 10. I
cant see why that would make a difference though.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: 06 May 2009 17:53
To: asterisk-users@lists.digium.com
Subject: [asterisk
Hi,
I'm having problems with an asterisk server that's not offering Codecs
for ulaw and alaw as it should.
I've three servers in total: a1, a2 and "b"
A1 and A2 have pretty much the same config files, except IP address info
changes
Server B is configured to accept all inbound invites
Thanks Brian, I do remember seeing references to that AGI, but I've not
used AGI much yet either so was looking for something simple to setup
(hence the original SIPbroker config). Will try to find it though.
Adrian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I'll be sure to post back if I think of anything as I go
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: 14 August 2008 14:32
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-
Hi All,
For a 1.4 version asterisk, whats the recommended mechanism for dialling
with ENUM lookup? At the moment I user SIPbroker, but am getting tired
of it hanging on certain numbers, so I was thinking about implementing
it myself.
I've seen various vo-ip.info pages
(http://www.voip-info
ngup
The "[" before "0-9]" is needed.
Felippe Silvestre
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Sent: Thursday, August 07, 2008 07:46
To: Asterisk Use
[pbx_config]
104. Dial(${TRUNK}/${EXTEN}||Wr)
[pbx_config]
206. Busy()
[pbx_config]
The page at voip-info isn't too clear in the differences between 1.2 and
1.4
(http://www.voip-info.org/wiki/view/Asterisk+config+e
Why would you need to to that anyway?
Just set them to one port, but use different contexts to handle the
inbound traffic differently.
Adrian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: 25 July 2008 14:40
To: Asterisk Users Mail
Hi All,
When I use re-invite, does the Asterisk server stay in the SIP
conversation, and just RTP traffic diverts, or does the SIP transfer
away from the A*k server too ?
Thanks,
Adrian
___
-- Bandwidth and Colocation Provided by http://www.a
Most SIP clients have a logging ability.. you can use those.. but
turning on debug on the server is the best mechanism, as its whats going
on there that counts.
sip set debug
And if you want to get really into the lower levels, then tcpdump will
let you capture the packets for offline analysis i
I've got to agree.. I've never given it much thought either...
All of my calls are SIP/IAX based, coming in from the PSTN from a peer
like that too..
I've never tracked the total number of conference users... But I'll bet
we've hit at least 10.. And I've never seen the CPU go above 10%.. And
that
- Non-Commercial Discussion
Subject: Re: [asterisk-users] Default ringtone
Adrian Marsh wrote:
> Hmmm..
>
> Well indications.conf does have:
>
> country=uk
>
> But I've definitly just hearing a long-tone tone, long break, long
tone
>
> But the file is set to:
>
&g
iling List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Default ringtone
Adrian Marsh wrote:
>
> Hi All,
>
> I've trying to force on the ringtone generated for outbound calls with
> Dial,r but want the tone to be the UK standard.
>
> I use Zaptel, but don&#x
Hi All,
I've trying to force on the ringtone generated for outbound calls with
Dial,r but want the tone to be the UK standard.
I use Zaptel, but don't have any E1/T1 cards at all (am completely IP
based). So I don't think zaptel.conf will come into this (am I right??)
I've tried editing
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: 25 May 2008 15:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Logical AND
On Sunday 25 May 2008 07:10:22 Adrian Marsh wrote:
> exten => s,n,ExecIf( $[ $[ "
Hi Steve,
I can see what yours does, but I still get the same end result (even
though theres only a single "0" result now)
:
exten => s,n,ExecIf( $[ $[ "${PSTN_NUM:0:1}" != "0" ] & $[
${LEN(${PSTN_NUM})} = 10 ] ] |Set|PSTN_NUM=001${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp("SIP/
: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfer
Adrian Marsh wrote:
>
> Hi All,
>
> In my old telco days (SS7), if I was wanting to hand back a call to
> the network for transfer to a different PSTN number, there was a
> specific SS
Hi All,
I'm trying to figure out why in the below code, the PSTN_NUM variable is
always amended
exten => s,n,NoOp(${PSTN_NUM})
exten => s,n,ExecIf( $[ "${PSTN_NUM:0:1}" != "0" ] & $[
${LEN(${PSTN_NUM})} = 10 ]|Set|PSTN_NUM=001${PSTN_NUM})
exten => s,n,NoOp(${PSTN_NUM})
-- Executing [EMA
: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfer
Adrian Marsh wrote:
>
> Hi All,
>
> In my old telco days (SS7), if I was wanting to hand back a call to
> the network for transfer to a different PSTN number, there was a
> specific SS
Hi All,
I'm trying to figure out why in the below code, the PSTN_NUM variable is
always amended
exten => s,n,NoOp(${PSTN_NUM})
exten => s,n,ExecIf( $[ "${PSTN_NUM:0:1}" != "0" ] & $[
${LEN(${PSTN_NUM})} = 10 ]|Set|PSTN_NUM=001${PSTN_NUM})
exten => s,n,NoOp(${PSTN_NUM})
-- Executing [EMAIL PR
Hi All,
In my old telco days (SS7), if I was wanting to hand back a call to the
network for transfer to a different PSTN number, there was a specific
SS7 action I could take, which send the call back to the network, which
in turn then routed the call appropriately. It added a transfer-number
in
.
Murrell
Sent: 17 May 2008 21:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Googles 411 services
On Sat, 2008-05-17 at 18:38 +0100, Adrian Marsh wrote:
> All,
>
> Does anyone know of a SIP URI direct to googles 800-GOOG-411 service?
Yeah, I
All,
Does anyone know of a SIP URI direct to googles 800-GOOG-411 service?
When I put calls via sipbroker, half the time the calls fail. An enum
lookup shows 3 URIs listed, none of them seem to be google directly, and
I think 1 of them fails 100%, and the remaining one fails at other
random t
Hi,
Can anyone confirm if calls placed via sipbroker have their NUM CLI
changed by sipbroker??
I'm testing between two asterisk servers in seperate locations. When I
place a call directly, the CLI is fine. When the call is placed via
sipbroker lookup, the NAME stays the same, but the NUM is recie
My exact requirement.. to edit out some recorded hiss and then put the
file back...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russell
Bryant
Sent: 08 May 2008 15:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-u
Hi All,
Whats the SLN file format (for import/export to Audacity)?
Need to avoid Sox if I can
Adrian Marsh
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
Basic process:
1) Build the A*k server so that it has tftp installed (or another box
that does)
2) Build up the SIPdefault.conf and get the firmware files in place (see
Cisco docs on this, plus theres loads on the wikis).
3) Test with a single phone, change its tftp server to the asterisk.
Check t
3rd attempt.. get the right list...
Hi All,
When I hairpin calls out to some networks (eg international or mobiles),
there can be a long delay until the PSTN starts sending audio ring tones
back. Is there a way I can have asterisk play ringtones until the PSTN
really answers??
I've loo
What version of Asterisk are you running? What setup ?
I've just hit issues with 1.4.19.1 (see previous post)... Same
symptoms-ish.
Adrian
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian
Sent: 30 April 2008 10:34
To: Asterisk Users M
Any ideas? I've downgraded my primary system back to 1.4.18 and will
upgrade the backup to 1.4.19.1 tomorrow to test a bit more.
Very confusing..
Adrian
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Sent: 29 Apri
Hi All,
I'm trying to debug DTMF issues I have with certain endpoint
conferencing systems (external, 3rd party).
On our A*k server I log DTMF, and I see that coming through in the log.
What I'd like to see is what is sent onto our VoIP carrier over SIP.
I can do a tcpdump of the packet
sers Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] UK GMT/BST settings
On Wed, Mar 26, 2008 at 10:43:13AM -, Adrian Marsh wrote:
> Ah ok,
>
> Those settings do seem to work (test phone was going to a different
> tftpd server..)
>
> Anyone know if the Ci
Ah ok,
Those settings do seem to work (test phone was going to a different
tftpd server..)
Anyone know if the Ciscos re-download SIPDefault.cnf periodically, or
only on boot ?
Thanks,
Adrian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Hi,
Anyone know what the settings in SIPDefault.cnf should be for Cisco 7940
phones this year?
Came in today to find they'd all moved one hour ahead (NTP server is
correct and ok). Found the "day" was set to "26", but on trying to
change the settings to the below, my test phone isn't changing bac
Anyone have an idea on this?
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Marsh
Sent: 17 March 2008 17:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Turn off MusicOnHold for individual User
Hi All,
I might of got my wires
Hi All,
I might of got my wires crossed here, but I'm looking for a way to
disable musiconhold for individual users.
I had thought that putting the sip.conf entry to:
[690]
type=friend
context=from-sip
secret=*
qualify=yes
host=dynamic
canreinvite=no
nat=yes
mailbox=2090
calle
Thanks guys,
On two cloned machines, on one I tried:
yum install lm_sensors-devel bzip2-devel
(ignoring newt, and these were the only ones missing)
..and it compiled ok. Then on the other I just added lm_sensors-devel
and the configure -with-net-snmp worked ok, but it didn't compil
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 pre-requisit
Vytenis,
As far as I understand it 1.2 zaptel should be used with 1.2 Asterisk,
1.4 with 1.4.
As for 1.2 vs 1.4, it depends on if you want new features and any
bug-fixes. 1.2 is a closed project (I think).
Just compile from source if its not available as an RPM in 1.4 for
Debian.
Adrian
-O
Hi All,
I'm going to be upgrading our 1.2 Asterisk system. At the moment we use
the Enicomms SLN files. Are there major differences in the 1.4 default
voicefile packs, or will I be able to re-use Enicomms??
In the Make menuselect, I noticed theres no .SLN voicefile selection for
the basic audiof
Correction:
netstat -an|grep 5060
udp0 0 0.0.0.0:50600.0.0.0:*
Adrian Marsh
From: Adrian Marsh
Sent: 27 November 2007 15:24
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-
Zaheer,
If a "netstat -an|grep -I LISTENING" shows that a LISTENING port for
5060 is there, then the problem isn't Asterisk, but some firewall system
on the server is blocking access from outside. If its not there, then
come back to the group..
Hi,
Can anyone recommend any company that can provide PSTN termination for
SIP calls, at least USA based, preferably California area. One of my A*k
servers is US based and I don't want my traffic flowing back and forth
via my current UK PSTN provider for US<>US calls.
Thanks,
Adrian
_
pting, replacing the tftp config
files for that phone, and then remotely resetting the phone, however
that would be quite clumber sum.
And before I go that route, I wondered if any of the commercial A*k
systems already offer this?
If the Ciscos can't do this.. then can any other h
Sorry - should add - AFTER its been initally tftp'd and firmware changed
to SIP. (i.e. changing existing settings of a working phone).
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Sent: 16 October 2007 11:37
To: Ast
Hi,
Does anyone know if its possible to change configs on a 7940G remotely,
without having to reboot/tftp the device?
I can login via telnet, but can't see how to change settings.
Thanks,
Adrian
___
--Bandwidth and Colocation Provided by htt
, so maybe it's a case of looking at
Linux-HA.
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jared
Smith
Sent: 25 September 2007 15:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Red
h a DNS change could take over
operations.
However I'd like to achieve something more automated if possible.
I know that some of my VoIP trunk providers "cluster" IAX connections, but I'm
not sure how they would do that.
Any ideas?
Adrian Marsh
Hi All,
Can anyone tell me how the below can be happening?
-- SIP/205-08439ee0 is ringing
-- SIP/405-084468f8 is ringing
-- SIP/405-084468f8 is ringing
-- SIP/405-084468f8 is ringing
-- SIP/405-084468f8 is ringing
Where, according to A*k, its ringing the same SIP device at t
I don't think * means anything special to A*k,
But wouldn't it be:
_X.*X.
To match as you ask ?
(number)(wildcard)*(number)(wildcard)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Messina
Sent: 14 September 2007 17:40
To: Asterisk Users Mail
Satish,
Whats your network setup? Do you get bad quality on internal-asterisk calls, or
only on external calls? Are you making pure IP calls (sip2sip), or are there
E1/T1 cards involved? What codecs are you currently using? What devices are you
using?
Adrian Marsh
d and back
again), and would probably only work well for home-users who aren't
mobile at all. Not sure how you'd implement this into Asterisk though.
Adrian Marsh
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But then how do you know which is the "correct" user?
This is where the whole point of secrets/passwords should come into
play. If no-one else knows his details, then no-one else can register.
In the land of IP, you can't even guarantee that a remote ends IP will
be the same from minute to minute..
I believe you can use the host= to configure the allowed IP in sip.conf
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Hisham
Sent: 11 September 2007 11:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk
200 OK
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4a6d0d34;rport=5060
From: "asterisk" ;tag=as35c7a074
To: ;tag=1624959632
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Contact: "Adrian Marsh"
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Accept: application/sdp,applica
Hi,
I noticed today, that there was a stale SIP call on my 1.2.24 A*k
server. One call (X-lite client) started yesterday into a meetme
conference. For some reason the call stayed established.
>From network stats, I see transmit data but no receive (as obviously the
client went offline).
Luckil
Hi All,
I'm working from home today (DSL -> Internet -> 2MB leased line -> A*K
server behind NAT), and trying to pickup voicemail using Zoiper..
I can access the VM system, I hear all the prompts, and I can even hear
part of the message playback.
But then I get silence on the call (call stays
Many thanks for that!! I didn't know that the order worked quite like
that but I see it now... Better go check the other contexts...
(the [56][0-9] worked fine).
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis
Sent: 05 September
me pay per play networks that do peering automagically (such
as XConnect), but it's a cost per connected call (granted, a tiny one,
but still a cost), and it won't guarantee you any better connectivity to
a closed network than, say, SIPBroker.
N.
Adrian Marsh wrote:
> Yeah,
>
5201[56][0-9],1,Goto(local,${EXTEN:-3},1)
exten => _0.,1,Set(CALLERID(num)=${PSTN_GLOBAL}${CALLERID(num):-3})
exten => _0.,2,Dial(${TRUNK}/${EXTEN},,W)
Adrian Marsh
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