Hi,
I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install. The
add-ons pack has been installed for a while, so now I'm trying to add
the Mysql config.
I've created a mysql database, added the grants for a user acces, and
can run a mysql -u asteriskcdruser -p and can connect to the
;
Reading table information for completion of table and column names
You can turn off this feature to get a quicker startup with -A
Database changed
mysql select Host from user where User = 'asteriskcdruser' ;
ERROR 1146 (42S02): Table 'asteriskcdrdb.user' doesn't exist
mysql
Adrian Marsh
-Original
ast_config_engine_register: Registered Config Engine mysql
MySQL RealTime driver loaded.
res_config_mysql.so = (MySQL RealTime Configuration Driver)
This box also das Cacti installed on it, which makes use of the MySql
server as well (and all is ok there).
Adrian Marsh
non-zero on
'IAX2/ubigradin-2' in macro 'ext-group-home'
== Spawn extension (macro-ext-group-home, s, 2) exited non-zero on
'IAX2/ubigradin-2'
-- Hungup 'IAX2/ubigradin-2'
Adrian Marsh
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Hmm... He swears he heard a voice saying he'd dialed the number
incorrectly.. But that's no-where in the dialplan, and I do see the
incoming calls correctly for the times he's saying..
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
),
but the important content of the Register message seems the same. I've
ruled out ISP/firewall interference, as its happened on a number of
users.
Obviously there are fixes in 3.8.6, so I don't want to downgrade the
phones again, but I can't see why they'd fail...
Adrian Marsh
Thanks for that Arnaud, saw it myself this morning, but the download
link takes me to a page not found cisco page :( I've reported it on
their broken links page...
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arnaud
Ligot
Sent: 22
...
Adrian Marsh
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at 100% CPU, 1000's of log files
In article
[EMAIL PROTECTED],
Adrian Marsh [EMAIL PROTECTED] wrote:
Hi All,
Twice now in the past few weeks I've walked into the office to find
that
our 1.2.24 Asterisk process is sat at 100%, and that hundreds of
thousands of log files in /var/log/asterisk
,${EXTEN},1)
Adrian Marsh
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extension number. Ideally I need to capture
the international combinations too:
SOMEVAR=+44179344
And
SOMEVAR=0044179344
Is there regexp * in Asterisk ? :
exten = _.${SOMEVAR}.,1,NoOp(Dialled own number)
Adrian Marsh
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deal with this.
Adrian Marsh
Why not just use Exten = _+.,1,Goto(011${EXTEN:1})
Exten = _011.,1,Dial(..
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Hi All,
If I had a large block of code, eg:
[outgoing-pstn-gradwell]
; the caller ID convertion assumes that the last two digits of the
callers id
; are mapped to the last two digits of the PSTN number.
exten =
_0.,1,ExecIF($[${RECORDOUTBOUND}=TRUE],Monitor,wav|${TIMESTAMP}-${CA
I'm not sure if a macro would work, as it's the exten = _0 bit that
we'd be looking at:
[macro_test}
Exten =
_0.,1,ExecIF($[${RECORDOUTBOUND}=TRUE],Monitor,wav|${TIMESTAMP}-${CA
LLERID(num)}-${EXTEN}-${UNIQUEID}.WAV)
exten =
_0.,2,ExecIF($[${LEN(${CALLERID(num)})}=4],Set,CALLERID(number)=${PSTN_G
Could you give me a short example? I've always been confused on the usage
of s.. How would you pass _0 or _**777 to it ?
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil Reynolds
Sent: 31 August 2007 12:13
To: Asterisk Users Mailing
Subject: Re: [asterisk-users] How to handle + prefix
On Thu, 2007-08-30 at 10:17 -0500, Brian West wrote:
On Aug 30, 2007, at 10:11 AM, Jared Smith wrote:
On Thu, 2007-08-30 at 15:42 +0100, Adrian Marsh wrote:
Is there a way of using variables within the dialplan, eg:
[globals]
SOMEVAR
What logs are coming out to /var/log/messages?
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
Jacob
Sent: 04 September 2007 07:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] stop log
All,
I've been experimenting with shortcodes for SIPgate etc. Passing calls
to SIPbroker seems a good way to go, but the message I've had back from
SIPgate is we don't support SIPBroker...
So whats the easiest way to support SIP SIP network calling?
At the moment, I've setup some local
- Notice Error
Tzafrir Cohen wrote:
On Tue, Sep 04, 2007 at 10:43:15AM +0100, Adrian Marsh wrote:
What logs are coming out to /var/log/messages?
Ask asterisk
logger show channels
EMAIL DISCLAIMER : This email and any files transmitted with it are
confidential and intended
..
A.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SIP
Sent: 04 September 2007 15:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIPBroker vs SIPgate
Adrian Marsh wrote:
All,
I've been experimenting with shortcodes
,${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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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,
I can see that now after
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 2007
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
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).
/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4a6d0d34;rport=5060
From: asterisk sip:[EMAIL PROTECTED];tag=as35c7a074
To: sip:[EMAIL PROTECTED]:5060;tag=1624959632
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Contact: Adrian Marsh sip:[EMAIL PROTECTED]:5060
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
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:
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..
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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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
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
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
.
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
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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 Redundancy
On Tue, 2007-09-25
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
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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: Asterisk
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 hardphones?
Adrian Marsh
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
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,
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 back:
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
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 Ciscos re-download SIPDefault.cnf periodically, or
only on boot ?
As far
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
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
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
Hi All,
Whats the SLN file format (for import/export to Audacity)?
Need to avoid Sox if I can
Adrian Marsh
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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:
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
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
.
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 suppose
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
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
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
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 SS7 action I could take, which send
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 SS7 action I could take, which send
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]
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( $[ $[ ${PSTN_NUM:0
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
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't have any E1/T1 cards at all (am
- 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:
[uk]
description = United Kingdom
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
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 options
And if you want to get really into the lower levels, then tcpdump will
let you capture the packets for offline
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
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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
. 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+extensions.conf+sort
ing) so I'm not sure where I've gone wrong.
Adrian Marsh
] is needed.
Felippe Silvestre
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Sent: Thursday, August 07, 2008 07:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
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
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-users
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
Hi All,
I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used
for PSTN calls via IAX2.
Our 'net link is a dedicated 2Mb fibre connection (of which we have ever
used 50% max bandwidth). We've no E1/T1 links, everything is IP based.
My boss complains that many of the calls he
, that would help the Meetme function? Maybe
different codecs?
Thanks,
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed W
Sent: 20 April 2007 19:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can I
2007 19:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can I improve call quality?
Try turning the jitterbuffer off, I found that often the endpoints can
do better on their own.
On 20 Apr 2007, at 19:01, Adrian Marsh wrote:
Hi All,
I've a single
So which is the best quality?
Gradwells www site lists g711u and g729a, but we currently use ulaw/alaw
with them too..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: 22 April 2007 09:18
To: Asterisk Users Mailing List -
Hi,
I'm having complaints from some users about calls into dynamic meetme
sessions failing. I'm guessing that they are dialling the wrong DTMF
keys, OR that DTMF is hearing the digits entered wrong (or not hearing
some).
I've put debug = debug into logging.conf, and searched through the
file,
line in logger.conf then do a logger
reload.
That will get you DTMF.
-Chris
On 6/4/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Hi,
I'm having complaints from some users about calls into dynamic meetme
sessions failing. I'm guessing that they are dialling the wrong DTMF
keys, OR that DTMF
/log/asterisk/full.HCCAPP1.usad File Enabled- Debug DTMF
Verbose Warning Notice Error
Console Enabled- Notice Error
[HCCAPP1 0.01]--
Make sure you see DTMF listed in your logger channel
On 6/5/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Thanks
All,
I've a question on A*k hardware.
I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz)
with 512mb RAM.
I'm supporting 60 users (Cisco 7940s each + Xlite PCs).
Call loads are low, max of about 10 simultaneous SIP/IAX calls.
CPU for A*k rarely goes above 2% as I can tell.
Its
To: Adrian Marsh
Subject: Re: [asterisk-users] Hardware spec comparison
On Tue, Jun 05, 2007 at 06:51:40PM +0100, Adrian Marsh wrote:
All,
I've a question on A*k hardware.
I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz)
with 512mb RAM.
I'm supporting 60 users (Cisco
Hi,
Are there any decent (commercial or free) LOG parsers for A*k.
Its *really* hard to debug issues involving multiple calls (eg meetme)
when all of the messages are interlaced with each other. There must be
an easier way. (A*K 1.2.18)
Adrian
___
to the procedures, I should be able to upgrade, but once the
phones loaded and reboots it says it downgrades again and reboots, then the
cycle starts again.
Anyone had any success in doing this?
Adrian Marsh
smime.p7s
Description: S/MIME cryptographic signature
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
Salas M.
Sent: 17 June 2007 15:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Upgrade cisco SIP phone 7940
On Sun, 2007-06-17 at 13:45 +0100, Adrian Marsh
Hi All,
I'm wondering if anyone can share any info on why I frequently get peer
timeouts like below, and receive 489 messages from another A*k server on
the same LAN.
For the peers, we've one L2 switch. ICMP is 1ms. The CPU of the main
A*k server is usually 2%. So I can't see why we'd get
Hi All,
Is there a way to have A*k record calls on-the-fly, at the users
request? i.e. a possible scenario:
Party A calls Party B
During the call, Party A wants to start recording the call, so presses
*, A*k announces recording.. and starting MixMonitor to a file.
Once the call is finished,
I made some progress on this issue...
It seems that I now see logs of DTMF for IAX/SIP outbound calls, but not
for internal SIP calls (aka meetme).
Not sure why.
A.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Sent: 05 June 2007 18:40
and it will complete the recording.
Rob
Drew Gibson wrote:
Adrian Marsh wrote:
Hi All,
Is there a way to have A*k record calls on-the-fly, at the users
request? i.e. a possible scenario:
Party A calls Party B
During the call, Party A wants to start recording the call, so presses
*, A*k announces
Scrap that... Tried the Set() method and it worked, so then I moved it from
[general] to [globals] and it does now record the calls.
A.
-Original Message-
From: Adrian Marsh
Sent: 20 June 2007 10:06
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk
it?
A.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Marsh
Sent: 20 June 2007 10:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inline record
Scrap that... Tried the Set() method and it worked, so then I moved
Or you could just show him... :)
Heres one I added, I have a global variable defined (RECORDSIP), so that
I can switch the record on/off without having to hack the code all over
the place... This uses Monitor instead of mixmonitor as I only want one
file. In the dialplan I have:
exten =
Hi All,
Anyone know if theres a way to share a Cisco 7940 between hot-desk
users?
My phones get their setup via SIP .cnf files, that load at boot via
tftp, so I'm assuming the configs a failry static. However if I want a
phone to be hot-desked, I could have different users sitting there. Is
they can change.
- I don't really want a www-based system, as most of my users are usually
mobile, and might not have access to the corporate intranet.
Thanks,
Adrian Marsh
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asterisk-users
Thanks Jason,
Could you post some sample code? What do you do if the CLI is not present ?
(i.e. international callee..)
Thanks,
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Fuermann
Sent: 02 April 2007 14:32
To: Asterisk Users
How do I do that? Doesn't Ds create dynamic room numbers?
Thanks,
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: 01 April 2007 11:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
Anyone know of any tools for interpreting master.csv call logs?
(Excel is kind of basic)
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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.
and 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
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
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 the table
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
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: [asterisk-users
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
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