...and that's saying something! ;)
Then again, we just need to remember the old saying... to err is human,
but to really foul things up requires a computer.
Paul Hales wrote:
It was one of those moments in life where I felt a lot less smart than I
usually do...
PaulH
On Fri, 2008-03-07
That's because A is the joining point between B and C. If either B or
C hung up, the remaining party would still be left.
This is a phone function, not an Asterisk one. From Asterisk's
perspective, phone A is simply on two simultaneous calls to B and C - it
has no idea that A is bridging the
Hi there,
Googling through the archives it looks like I'm the ferst person to want
this...
My aim is to set up a voicemail application with a custom greeting before
*AND AFTER* the punter has left the message.
Right now the relevant section of my dialplan is like this:
exten =
I still enjoy the saying to err is human, effective chaos requires root
access
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis
Sent: 07 March 2008 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie
In article [EMAIL PROTECTED],
Lee, John (Sydney) [EMAIL PROTECTED] wrote:
I was successful to control the max users (10) if I hardcode the
conference room number (in this case 101) as follows:
exten = 8600,1,Playback(conf-thereare)
exten = 8600,2,MeetMeCount(101)
exten =
Hello,
Just find this file in /var/lib/asterisk/sounds and change it to anything
you like.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Horwich IT
Dear all
I'm writing to the list for help as a last resort. I've exhausted all
other options, so please forgive me. I've lurked here for years but
never actually posted.
I'm trying to get Asterisk Realtime SIP configuration working, but it
refuses to do so. I have all the necessary
Hi,
Thanks for your reply.I got what is happening in that scenario.
Could you please tell me how the Astrisk pbx identifies a confernce? What
is the sequence of sip messages supported by the astrisk pbx conference?
Any help will be highly appreciated.
preethy
On Fri, Mar 7, 2008 at 5:50 AM, Faraz Khan [EMAIL PROTECTED] wrote:
Gonzalo,
Please let us know what you mean by 'stops working' - it should spit
out errors or wrong queries to ldap.
Basically what I mean by that is that in the slapd debug, no activity was
going on when I tried to
Hello users,
actually we are tyring to setup a dialer to test outbound autodialer
and we are uanble to bridge the answered outbound calls to the
local agents and the debug in asterisk is showing the follwoing
error message:
chan_sip.c:2918 auto congestion
can anybody have any idea where might
What does show config mappings show ?
Julian
Stuart Ford wrote:
Dear all
I'm writing to the list for help as a last resort. I've exhausted all
other options, so please forgive me. I've lurked here for years but
never actually posted.
I'm trying to get Asterisk Realtime SIP
Julian Lyndon-Smith wrote:
What does show config mappings show ?
zateteis*CLI core show config mappings
zateteis*CLI
Config Engine: mysql
Seems mysteriously ... light :)
Stu
--
Stuart Benjamin Ford
Chief Technical Officer
Zimo Communications Ltd.
Landmark House
Station Road
Cheadle Hulme
On Fri, 7 Mar 2008 12:10:37 +0200, Mindaugas Kezys [EMAIL PROTECTED]
wrote:
Just find this file in /var/lib/asterisk/sounds and change it to anything
you like.
But that will break other applications that use the auth-thankyou sound,
Authenticate() for a start (which I use elsewhere in order to
Julian Lyndon-Smith wrote:
your extconfig is wrong.
try
[globals]
[settings]
sipusers = mysql,valid db,telephones
sippeers = mysql,valid db,telephones
Oh you're a life-saver! I have other problems with it now but they're
unrelated to this config and I don't expect they're
On Fri, Mar 7, 2008 at 8:46 AM, Gonzalo Servat [EMAIL PROTECTED] wrote:
On Fri, Mar 7, 2008 at 5:50 AM, Faraz Khan [EMAIL PROTECTED] wrote:
Also please keep this list in your replies. I have no problems
answering personal emails but both of us might get more feedback if we
post our
It does work. Did you do the switch statement in extensions.conf?
If not check voip-info for Asterisk Realtime Extensions
Quoting Gonzalo Servat [EMAIL PROTECTED]:
On Fri, Mar 7, 2008 at 8:46 AM, Gonzalo Servat [EMAIL PROTECTED] wrote:
On Fri, Mar 7, 2008 at 5:50 AM, Faraz Khan [EMAIL
Hi All;
I am using Background in my configuration, and I
noticed the following so if any can help:
1) If I pressed 1 twice (11), so it runs the step
related to first 1 and then it runs the step related
to second 1, so it does buffering for my input and run
two steps, how can I make it run only
Hello,
Then you can change channel language in front of VoiceMail() app and in
appropriate place put auth-thankyou file which is recorded/made by you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL
The standard G.732 :
*Multiframe Aligment Signal:
*
When Timeslot 16 of the E1 frame is used for Channel Associated Signaling
purposes, Frame 0 contains information that is used by the receiver to
identify the incoming frame. Specifically, this pattern in Timeslot 0,
Frame 0 is called the
Well, then it is irrelevant what version of unicall, spandsp, asterisk
etc you are using.
All that it could matter is your zaptel version and zaptel.conf
configuration. I ignore why you need to control the bit pattern at
that level, but AFAIK zaptel only let you control the CAS ABCD bits in
E1
I have a Hitachi WirelessIP5000 phone (the original 1st gen hardware). It's
running the latest firmware (v2.2.6) and my Asterisk server is running
1.2.10. This setup has been working great for me for a long time, and then
last week I started having a problem where the Hitachi phone loses
For outbound trunking we go directly from Asterisk to the terminating
gateway no SIP Proxy involved. For inbound trunking we do go through
the SIP Proxy for the same reason you get users to. Incoming calls are
going to be more reliable if they are not tied to a single Asterisk
server (I
There are a few gotchas with a SIP Proxy the main one being transfers.
But if you can get away with not allowing transfers then you are best
to do so as the CDR's Asterisk produces are wrong anyway.
What is the transfer problem? Is it the Asterisk native type using
features.conf or the SIP
Asterisk Voicemail for iPhone
Download is available. Also, live demo is available.
Many features are still missing and I would love to get some help from
the community.
Developers, please download and submit fixes and improvements.
http://chriscarey.com/projects/asterisk/iphone/
--
Asterisk
Asterisk 1.2.26.2
On an ACD call, I can press 0 to do attendant transfer. After talking to the
transfered party, I want to cancel the transfer and get back to the original
party. If I press *, it will disconnect me and complete the transfer. How can I
set it up so I can press * and get the call
I am looking at the asterisk dialout context and trying to set a invalid
extension (i).
The question I have is how to get the user back into the VoiceMail menu after
leaving it. Do I need to have them login again through VoicemailMain?
I know it does not come back like a gosub.
Thanks in
I am using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18
Spectralink wireless IP phones.
Most of the Spectralink phones have entries in 'sip show channels'
that do not go away. None of the other phones do this.
Is there anyway to remove these entries without restarting Asterisk?
Does anyone on the list have information regarding using the Motorola SBV5220
to connect to an asterisk box? The modem is leased through the cable provider,
but I have signed up for their voice plan. What I would like to do instead is
still use the RJ11 jacks on them and tell the modem to
On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote:
http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html
Trouble is, you'll need 7 32-port units to cover your needs and I'm not
sure if USB2 is up to driving that many ... Tzafrir?
One USB connector
My equipement
: 2x tdm 400p /4FXO/4FXS
16 PORT ASTRIBANK
My first TDM400p is the sync master, so i set astribank to sync to it, but
the quality is bad, like it is going fine and second after ROBOVOICE ;)
any other devices (isdn/sip/tdm cards) are working fine, looks like
astribank dont like to be
On Fri, Mar 7, 2008 at 3:00 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote:
http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html
Trouble is, you'll need 7 32-port units to cover your needs and I'm
On Fri, Mar 7, 2008 at 5:43 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Mar 7, 2008 at 3:00 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote:
http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html
200 extensions, take 100 PAP2 and you´re set.
The trouble would be configuring them all.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I have a similar problem with UT Starcom F1000g, after a while it loses
registration, sip show peers says its IP address is unknown. But the phone
still says it's registered, I can still see SIP packets and I'm able to make a
call from this phone. But if i try to call back, it dies with Unable
On Fri, Mar 07, 2008 at 03:00:03PM -0500, Jay R. Ashworth wrote:
On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote:
http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html
Trouble is, you'll need 7 32-port units to cover your needs and I'm not
Hi
On Fri, Mar 07, 2008 at 11:23:03PM +0100, Grygoriy Dobrovolskyy wrote:
My equipement
: 2x tdm 400p /4FXO/4FXS
Any other Zaptel devices on that box? (you mentioned isdn cards)
16 PORT ASTRIBANK
My first TDM400p is the sync master, so i set astribank to sync to it, but
the quality is bad,
On Fri, Mar 7, 2008 at 8:02 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Mar 07, 2008 at 03:00:03PM -0500, Jay R. Ashworth wrote:
On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote:
http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html
As expected, Jim took care of me WRT the Cisco upgrade. It is now far more
usable than when it was SCCP... I gave up on trying to get SCCP working in
Asterisk after upgrading to 1.4 from 1.0. Due to his generosity, I feel I
owe him to recommend his termination\origination services. The one
On Wednesday 05 March 2008 12:05:40 Joshua Kinard wrote:
That'd be ASCAP (I think there's another one too). They're the ones known
for calling up places, asking to be put on hold to listen to the hold
music, then querying on whether it's been licensed or not (among other
tactics).
BMI
Same problem over here
I use KIRK-Telecom ip600v3
This only happens on calls between SIP en MiSDN, anyone any clue?
As far as i can see these dead calls once in while occur when the
remote party first hangs up (remote=MiSDN channel)
Keith do you also have error messages in
Even if I have s in defult it is not work.
2008/3/6, Noah Miller [EMAIL PROTECTED]:
Hi -
Thank you all for answers. As I understand s - i and others is
device specific.
I will not need them in my SIP configuration.
The s extension is not zap-specific. You can use
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