Should that not say parkinglot and not parkinglog in features.conf?
-
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On 12 Jan 2010, at 06:03, Michael Wyres wrote:
Has anyone managed to get multi-parking lot
At 23:57 1/11/2010, Michael Wyres wrote:
Content-Language: en-US
Content-Type: multipart/related;
boundary="_004_11FDCFCDD2B4B0439630AEC725D1635D1BAC56FDC4ssyd10exinter_";
type="multipart/alternative"
Has anyone managed to get multi-parking lot call
parking working correctly? Ive had
Has anyone managed to get multi-parking lot call parking working correctly?
I've had several attempts at it, and never seem to be able to get it to go
properly - (actually, at all):
I've most recently done this with 1.6.1.x, and now 1.6.2.x, with no luck in
either case. What I've been "trying
hi,all
when in talking status, agent log out automaticly, why? following are
output in CLI*
do you know the reason？
== Agent '1002' logged in (format ulaw/ulaw)
-- Executing [...@tutorial:1] Queue("SIP/ivan-0013", "queue1")
in new stack
-- Started music on hold, class 'default', on
Dear all,
I can't understand the diff between roundrobin and rrmemory strategy.
Could you explain for me ?
and is roundrobin means each available interface ring once or several
times and ring another?
; A strategy may be specified. Valid strategies include:
;
; ringall - ring all available chan
Hi,
We are currently in the process of installing an Asterisk box between
our legacy PBX (NEC Xen Master) and the Telco (Telstra). Originally the
PBX was connected to the telco via 8x BRIs. Our new Asterisk box has
been setup with 2 OpenVox B800P cards. Each card has the 4 ports in TE
mode and
Hi all.
I have a (new) customer who is describing symptoms that I've not seen before.
They have 12 Polycom 430's behind a NAT, which is working OK. When phone A is
on a call and phone B attempts to transfer another call to phone C, the
conversation on phone A is interrupted for 15-20 seconds..
The VPN termination is a Cisco/Linksys RV042.
> We have that solution running fine...
>
> Is your VPN termination a Linux box? Is it also the office router? Is it
> also the firewall?
--
_
-- Bandwidth and Colocation Provided b
Are you using openvpn? If so, there's an option in the server config
file that allows vpn clients to talk to other vpn clients, otherwise
they can only talk to the server. Using canreinvite=no is just forcing
the traffic to go through the server, which is why that makes it work.
I must say VPN + V
We have that solution running fine...
Is your VPN termination a Linux box? Is it also the office router? Is it
also the firewall?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan McCormack
Sent: Monday,
canreinvite=no did the trick! Thanks!!
> [Cary Fitch]
>
> One thought: if you are using "reinvite" try turning that off. That will be
> a clue.
>
> It would seem that both phones are on the local net via VPN, and should be
> able to talk to each other if they can talk to anyone in the office. (As
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan McCormack
Sent: Monday, January 11, 2010 1:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP over VPN -- no audio to other
remote/VPNco
Hello,
I am having a problem with my current SIP over VPN setup.
We have a server running asterisk at our office. All the phones in the
office are on the same network / local to this server. We also have two
employees with home offices using SIP phones over VPN to connect to the
asterisk ser
If you don't want guest using your asterisk box, make sure sip.conf
'allowguest=no', by default it's 'yes' when commented out.
If you do want guests, make sure the default context cannot dialout, this
allows you to publish your ip address, and allow anyone to dialyou, as one
senario.
Refer https:
- "Barry Miller" wrote:
> On Mon, Jan 11, 2010 at 04:42:44PM +, --[ UxBoD ]-- wrote:
> >
> > - "Barry Miller" wrote:
> >
> > > On Mon, Jan 11, 2010 at 03:23:33PM +, --[ UxBoD ]-- wrote:
> > > > Hi,
> > > >
> > > > why would Asterisk core dump with the following test dialplan
Hello
I have a simple configuration to allow the admins to listen the agents
calls:
exten => _654,1,ChanSpy(Agent)
exten => _654,2,Hangup()
The problem is... even when the agents hung up... it seems the channels
remain active:
asterisk*CLI> show channels
SIP/211-b3042018 6...@default:1
On 11 Jan 2010, at 17:06, listu...@spamomania.co.uk wrote:
> On Mon, 2010-01-11 at 16:52 +, Steve Howes wrote:
>>
>> Try asking Sipgate what settings you should use? If they are sending
>> it as audio, make sure you are using suitable codecs etc. Try SIP
>> traces to see what you can see.
> St
On Mon, 2010-01-11 at 16:52 +, Steve Howes wrote:
> On 11 Jan 2010, at 16:26, listu...@spamomania.co.uk wrote:
> > This has made no difference. I've tried a range of settings (auto,
> > rfc2833,info) but no matter what, it plain refuses to pick up key
> > presses.
> >
> > It's extremely frustr
Hi all,
I'm trying to get the MeetMe system to take a caller and announce to them
they've joined the conference in addition to the other members of the
conference assuming previous members of the conference >= 1.
I can see where the meetme.c app actually processes it using the
ast_pthread_crea
I have Asterisk 1.6.2 running on CentOS 5.4. I would like to use
mode=new in say.conf but when I do, SayUnixTime doesn't play any sounds.
i.e., SayUnixTime(,CST,ABd 'digits/at' IMp) says the date / time as
expected with mode=old, but changing to mode=new causes that to do
nothing at all.
Is t
On Mon, Jan 11, 2010 at 04:42:44PM +, --[ UxBoD ]-- wrote:
>
> - "Barry Miller" wrote:
>
> > On Mon, Jan 11, 2010 at 03:23:33PM +, --[ UxBoD ]-- wrote:
> > > Hi,
> > >
> > > why would Asterisk core dump with the following test dialplan
> > extension ?
> > >
> > > exten => 8100,1,An
On 11 Jan 2010, at 16:26, listu...@spamomania.co.uk wrote:
> This has made no difference. I've tried a range of settings (auto,
> rfc2833,info) but no matter what, it plain refuses to pick up key
> presses.
>
> It's extremely frustrating and I would be grateful if anyone could
> offer
> some he
- "Barry Miller" wrote:
> On Mon, Jan 11, 2010 at 03:23:33PM +, --[ UxBoD ]-- wrote:
> > Hi,
> >
> > why would Asterisk core dump with the following test dialplan
> extension ?
> >
> > exten => 8100,1,Answer()
> > exten => 8100,n,Set(CALLERID(all)="")
> > exten => 8100,n,PrivacyManager
I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to
recognize digits pressed on a keypad coming in from a Sipgate trunk.
There answer was to set this:
dtmfmode=rfc2833
in the general section of sip.conf
This has made no difference. I've tried a range of settings (auto,
rfc283
On Mon, Jan 11, 2010 at 03:23:33PM +, --[ UxBoD ]-- wrote:
> Hi,
>
> why would Asterisk core dump with the following test dialplan extension ?
>
> exten => 8100,1,Answer()
> exten => 8100,n,Set(CALLERID(all)="")
> exten => 8100,n,PrivacyManager()
> exten => 8100,n,GotoIf(${[${PRIVACYMGRSTATUS
11 jan 2010 kl. 16.23 skrev --[ UxBoD ]--:
> Hi,
>
> why would Asterisk core dump with the following test dialplan extension ?
>
> exten => 8100,1,Answer()
> exten => 8100,n,Set(CALLERID(all)="")
> exten => 8100,n,PrivacyManager()
> exten => 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} = FAILED]}?:noci
- Original Message -
From: jonas kellens
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, January 11, 2010 2:30 PM
Subject: Re: [asterisk-users] Some minor configuration issues with queues
To answer my own question :
I had the following in my dia
Hi,
why would Asterisk core dump with the following test dialplan extension ?
exten => 8100,1,Answer()
exten => 8100,n,Set(CALLERID(all)="")
exten => 8100,n,PrivacyManager()
exten => 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} = FAILED]}?:nocid)
exten => 8100,n,NoOp(Number is ${CALLERID(num)})
exten =>
Here is an exert of my speed dial system that pulls a phone number from a
database, and then connects the caller.
$AGI->verbose("Record found in database.",3);
$AGI->exec('Playback','/var/lib/asterisk/agi-bin/speeddial/trsf-call');
my $stmnt = $db->prepare("select phone from phonebook where
;")
You might find this helpful:
http://www.voip-info.org/wiki/view/Asterisk+AGI+php
Regards,
On Mon, Jan 11, 2010 at 2:19 AM, Zhang Shukun wrote:
> hi,
>
> i want to use $agi -> exec_dial() to dial .
>
> this is in extention.conf
>
> [tutorial]
> exten => 1234,1,Dial(SIP/ivan)
>
> is that i use
>
U alrigth!
The number begins with 8 the TELCO sent this number like DID
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*-Developer DataBase
*-JQ Microsistemas
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*-www.jqmicrosistemas.com
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*---
I have the issue where they hit me, get no where, and then my box tells them
invalid context, and it timeouts connecting back to them..
And I get these :(
[Jan 10 19:49:06] WARNING[4103] chan_sip.c: Maximum retries exceeded on
transmission 209673377-00012714169-309054...@117.34.72.42 for seqno 10
11 jan 2010 kl. 12.25 skrev ahmed magdy:
> Hello,
>
> I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know
> how can i monitor the extension status?
> when i wrote sip show peers on asterisk
> Extension Domain port Status
> 111
To answer my own question :
I had the following in my dialplan : Queue(VC_support_queue,r)
The 'r' option replaces the moh with a dialtone...
I have now replaced the 'r' with 'R', so that there is moh and a
ringtone when an agent is ringing...
(source: voip-info.org)
However the caller keeps hea
- "Robert Lister" wrote:
| On Mon, 2010-01-11 at 10:45 +, --[ UxBoD ]-- wrote:
| > Hi,
| >
| > I am starting to see a lot of these:
| >
| > [Jan 10 01:18:56] NOTICE[5627] chan_sip.c: Call from '' to extension
| '33155786056' rejected because extension not found.
| > [Jan 10 01:52:47] NO
On Mon, 2010-01-11 at 10:45 +, --[ UxBoD ]-- wrote:
> Hi,
>
> I am starting to see a lot of these:
>
> [Jan 10 01:18:56] NOTICE[5627] chan_sip.c: Call from '' to extension
> '33155786056' rejected because extension not found.
> [Jan 10 01:52:47] NOTICE[5627] chan_sip.c: Call from '' to exten
You're invited to the first test of the Global Asterisk bimonthly
evening meetings at BerkeleyTIP-Global. :)
Join in tonight, Monday Jan 11, 5-6P Pacific, 8-9P Eastern, = Tues Jan
12 1A-2A UTC.
http://sites.google.com/site/berkeleytip/schedule
On #berkeleytip on irc.freenode.net,
& on voip - wha
On Wed, 30 Dec 2009 11:43:59 -0800, vijay.go...@alliance-infotech.com
wrote:
>
>case 2: This skype account (rexesbposolutions) has been assigned with a
>online virtual number (00 44 20 ). If somebody dial this number
>from their landline/cellphone, call is transfered to Asterisk queue bu
ahmed magdy wrote:
> Hello,
>
> I am new in Asterisk Community, i am working on Asterisk 1.6, i need
> to know how can i monitor the extension status?
> when i wrote sip show peers on asterisk
> Extension Domain port Status
> 111/111(
Hello,
I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know
how can i monitor the extension status?
when i wrote sip show peers on asterisk
Extension Domain port Status
111/111(Unspecified)D 0
Hi, I'm trying to diagnose a particularly elusive problem, and am
wondering if anyone else here has seen anything similar and can offer
any ideas.
I have a conference bridge running Asterisk 1.2.32 (with slight mods),
in a colo talking via a LAN to an ITSP using SIP/RTP. It is dedicated
to a singl
Hi,
I am starting to see a lot of these:
[Jan 10 01:18:56] NOTICE[5627] chan_sip.c: Call from '' to extension
'33155786056' rejected because extension not found.
[Jan 10 01:52:47] NOTICE[5627] chan_sip.c: Call from '' to extension
'033155786056' rejected because extension not found.
[Jan 10 02:
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