Paul Hales wrote:
I would love to see the agent login/logout stuff working - but that's
just me.
I'd like to see the damn web interface become usable.
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asterisk-users
* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND
VIDEO TO MICROBLOGGING!
In a surprising move, Digium in partnership with Edvina today released
a new channel driver for Asterisk, chan_tweet. The driver connects
seamlessly to several microblogging platforms, including
haw haw haw...
April Fools Day is over in this part of the world.
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Are these functions what you are looking for?
QUEUE_MEMBER_COUNT: Count number of members answering a queue
QUEUE_MEMBER_LIST: Returns a list of interfaces on a queue
QUEUE_WAITING_COUNT: Returns the number of callers currently waiting in a
queue
Just my two eurocents,
l.
2009/3/31 Steve
2009/4/1 Michael mich...@networkstuff.co.nz
haw haw haw...
April Fools Day is over in this part of the world.
Hey dont kill the magic ! :)
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To
On Wed, 01 Apr 2009 21:01:28 you wrote:
2009/4/1 Michael mich...@networkstuff.co.nz
haw haw haw...
April Fools Day is over in this part of the world.
Hey dont kill the magic ! :)
April Fools Day ends at 12.00pm (mid day) here. It is now 9:07pm.
On Wed, 2009-04-01 at 09:18 +0200, Olle E. Johansson wrote:
snip
For more information, please do not contact Digium sales.
To be released: 2009-04-01
Should say enough...
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Karl,
I echoed your comment about a one button hit exit anywhere in the
menu, that is so lame, although you can fake it by lifting the handset
or hitting menu twice.
I think a serious ergonomic study of the entire Polycom Soundpoint
menuing interface is needed. It appears that little thought was
Nice one, Olle ! :)
On Wed, Apr 1, 2009 at 9:18 AM, Olle E. Johansson o...@edvina.net wrote:
* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND
VIDEO TO MICROBLOGGING!
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-- Bandwidth and Colocation Provided by
Hi friends.
Can you help me to use SIPP to stress my asterisk voicemail? I want to send my
own recorded media file to the voicemail system.
Thanks.
--
Linux User Registered #232544
Jabber : p...@jabberes.org
Ekiga : p...@ekiga.net
GnuPG-key :
ContactTel Business wrote:
People should use .020 ms sample rates for RTP as it's the standard. 0.030
was i think the old SPA implementations which caused MR, Roboto kind of
grabling.
You should find a way to patch your sip core i assume, but dev's could tell
you where.
We offer 0.020 ,
Hi,
i got quiet the same problem, but with g711.
Zoiper wan't really work if you got an ISDN Call, so Zoiper told me that the
Asterisk send 16ms packets to zoiper and he can't handle 16ms.
so if have to set 20ms, so what and how can i do this?
Thx
Timm
-
Many thanks - That is exactly what I want - I must have been using
poor search terms as I failed to find them on the Wiki previosuy :)
Regards,
Steve
2009/4/1 Lenz Emilitri lenz.lo...@gmail.com:
Are these functions what you are looking for?
QUEUE_MEMBER_COUNT: Count number of members
Marco Sambo wrote:
Mhmm. Thaht's strange!
modinfo oslec
--
modinfo: could not find module oslec
and
modinfo dahdi_echocan_oslec
--
filename: /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko
license:GPL
author: Tzafrir Cohen
But I don't have also echo
modinfo echo
modinfo: could not find module echo
2009/4/1 Dave Fullerton dfullertaster...@shorelinecontainer.com
Marco Sambo wrote:
Mhmm. Thaht's strange!
modinfo oslec
--
modinfo: could not find module oslec
and
modinfo dahdi_echocan_oslec
Hi,
Why does this warning occur and what are the implications of it? I'm concerned
about calls never getting hung up.!
chan_sip.c:12890 handle_response: Remote host can't match request CANCEL to
call '2f197e56611061a678c13b881b269...@411.2.139.106'. Giving up.
One thing!
I saw that I use kernel 2.6.26 in my asterisk machine. I should use kernel
2.6.28 or newer to use oslec with DAHDI???
2009/4/1 Marco Sambo derwid...@gmail.com
But I don't have also echo
modinfo echo
modinfo: could not find module echo
2009/4/1 Dave Fullerton
Hello all,
Probably a bad news for all...
The Undercompetent Olle E Johansson decided to leave the asterisk team
to create his own Voip server.
The server will be called Minisk (due probably to his poor competence in
Voip).
Following that, Digium decides to stop any development on Asterisk and
Marco Sambo wrote:
One thing!
I saw that I use kernel 2.6.26 in my asterisk machine. I should use kernel
2.6.28 or newer to use oslec with DAHDI???
You don't need to, if you read me previous email you'll notice I'm
running 2.6.27.19. Rebuild DAHDI with the instructions I linked to and
2009/4/1 Shaun Wingrin voi...@gmail.com
Hi,
Why does this warning occur and what are the implications of it? I'm
concerned about calls never getting hung up.!
chan_sip.c:12890 handle_response: Remote host can't match request CANCEL to
call
At least fake your from email to make it believable..
hh174 wrote:
Hello all,
Probably a bad news for all...
The Undercompetent Olle E Johansson decided to leave the asterisk team
to create his own Voip server.
The server will be called Minisk (due probably to his poor competence in
hh174 wrote:
Hello all,
Probably a bad news for all...
The Undercompetent Olle E Johansson decided to leave the asterisk team
to create his own Voip server.
Oh Ma GOSH! I guess I'll trash all my installs and move over to Avaya!
*snicker*
--
Ben Franklin quote:
Those who would
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten =
Hello,
In our previous PBX we have an option to turn off or on outside calls with
a pincode..
Like, user is able to get calls or dial local lines by default, but when
he/she uses a password entrance via dtmf, he can dial long distance calls
etc.And at anytime he can logoff from outside call
Check this:
http://www.voip-info.org/wiki/index.php?page=Call+Quality+Metrics
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Here is a simple control for what you are asking:
Exten = s,1,noop(Dial Long Distance #)
exten = s,n,Set(LDACCESS=${DB(LD/Access)})
exten = s,n(readacct),Read(digitacc,record/entercode,8,skip,1,10])
exten = s,n,Gotoif($[${LEN(${digitacc})} 4]?readacct)
exten = s,n,Gotoif($[${digitacc}
hello,
I am beginning to asterisk.
I have a sip trunk access to operator and VPN access with operator.
i booked 10 sda numbers.
IP adress asterisk : 192.168.600.1
IP adress operator : 192.168.700.50
i can ping on 192.168.700.50
# cat sip.conf
[general]
context=default
srvlookup=yes
port =
I wish we could have this for real
- Original Message -
From: Olle E. Johansson o...@edvina.net
To: Asterisk Non-Commercial Discussion Users Mailing List -
asterisk-users@lists.digium.com
Sent: Wednesday, April 01, 2009 10:18 AM
Subject: [asterisk-users] FOR IMMEDIATE RELEASE: NEW
On Wed, Apr 01, 2009 at 06:52:55PM +0300, Dovid Bender wrote:
I wish we could have this for real
Micro-video-blogging: Limited to 140B ?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8
bits each, and that is 2^140^8, a nearly inexhaustible key number which is
related to audio and video data simultaneously stored on a Google Database,
which is then sent to the user.
Thus with the 140 byte message, full
Cary Fitch wrote:
It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8
bits each, and that is 2^140^8, a nearly inexhaustible key number which is
related to audio and video data simultaneously stored on a Google Database,
which is then sent to the user.
Thus with the 140
On Wed, Apr 01, 2009 at 11:27:17AM -0500, Cary Fitch wrote:
It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8
bits each, and that is 2^140^8, a nearly inexhaustible key number which is
related to audio and video data simultaneously stored on a Google Database,
which is
Ok, this is where it gets interesting. Consider the case of a PBX
which has its own MOH source and is talking via Asterisk to another
PBX.
If that PBX wants to put the call on hold while sending its own MOH,
you would probably argue that it should not send a re-INIVTE at all,
but should
Regarding compression with g.729/gsm/etc. and Asterisk
If we convert all the voice files to the corresponding format g.729/gsm/etc.
and we send digits using RFC 3261 and we do not need silence detection, is
there still a need to decompress the media stream ?
If doable how to make
If half-duplex audio is good enough for you, sure.
You've lost me there. I am not aware of a modem that is for sale today that is
half duplex. (OK some support a couple of minor half duplex modes). All state
of the art modem protocols send and receive simultaneously using the full 300 -
3000
Hi Tony
I can see where you guys are coming from on this and have already
enumerated your argument in my own email.
But there are very real reasons for a PBX to signal the hold even when
it wants to send its own MOH:
1. Bandwidth: under your scheme the PBX would continue to receive
- Wilton Helm wh...@compuserve.com wrote:
If half-duplex audio is good enough for you, sure.
You've lost me there. I am not aware of a modem that is for sale today that is
half duplex. (OK some support a couple of minor half duplex modes). All state
of the art modem protocols send and
I don't think a off the shelf modem has the necessary DSPs to convert
voice to codecthat is why a Voice Gateway/Analog Telephony Adapter
or FXO/FXS cards exist instead of modem having a second life.
I do recall a few that worked as a answering machine allowing your home
computer to answer
Tim Nelson wrote:
- Wilton Helm wh...@compuserve.com wrote:
If half-duplex audio is good enough for you, sure.
You've lost me there. I am not aware of a modem that is for sale
today that is half duplex. (OK some support a couple of minor half
duplex modes). All state of the art
Fred wrote:
Hello
Considering how cheap PCI modems are compared to even the cheapest
PCI hardware from Digium, OpenVox, Sangoma, etc I was wondering
why Zaptel can't be used with those to connect an Asterisk server to
a POTS line for low-level use? It just seems overkill for SOHO
1 apr 2009 kl. 16.24 skrev Grygoriy Dobrovolskyy:
2009/4/1 Shaun Wingrin voi...@gmail.com
Hi,
Why does this warning occur and what are the implications of it? I'm
concerned about calls never getting hung up.!
chan_sip.c:12890 handle_response: Remote host can't match request
Hello,
I don't speak english very well but i think.
[operador]
qualify=yes
nat=yes
host=192.168.700.50
insecure=invite,port
canreinvite=no
context=default
disallow=all
allow=ulaw
allow=g729
in your extensions.conf
exten = _00X,1, Dial (SIP/operador/${EXTEN},60,tT)
Best Regards
On Tue, Mar 31, 2009 at 10:27:45AM +0100, Steve Davies wrote:
Most commonly, if DNS is not ready to resolve a hostname, IAX can
stall and/or fail to register.
DNS was the cause. Replacing the hostname with its IP address fixed it.
Thanks!
-Yahya
Okay, I am not understanding if I have this correct or not.
I have a requirement to allow guests into a PBX from different domains.
However, I can not allow the guests into the default context because each
domain has its own IVR. So I end up setting the domain context. I also need
to
Duuh guys - it's so easy. Ever thought of simply compressing the compressed
data AGAIN???
Do that the necessary amount of times and - tadaa - it's done.
Chris
2009/4/1 Brent Davidson br...@texascountrytitle.com
Cary Fitch wrote:
It uses proprietary EDC. (Extreme Data Compression) The 140
Yeah got it down to 1 bit that way.
exten byte1 = (dataflag=(${byte1}:bit1)?had-data:didn't-have-data))
If dataflag returns had-data recovering the data you call and parse an
external subroutine the same size and composition of the original data.
Otherwise no external routine is needed.
If you had done it once more you would have had it down to half a bit.
Quantum computing?
j
On Wed, 1 Apr 2009, Cary Fitch wrote:
Yeah got it down to 1 bit that way.
exten byte1 = (dataflag=(${byte1}:bit1)?had-data:didn't-have-data))
If dataflag returns had-data recovering the data
For those looking for the faq on that page :)
http://unix.derkeiler.com/Newsgroups/comp.os.vms/2003-07/2406.html
Tzafrir Cohen wrote:
On Wed, Apr 01, 2009 at 11:27:17AM -0500, Cary Fitch wrote:
It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8
bits each, and that
Computing used to be fun. Now we have to make the buttons on the phone
blink, even if the manufacturer didn't put an LED or circuit behind the
button.
:-)
CF
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
On Wed, Apr 1, 2009 at 3:18 AM, Olle E. Johansson o...@edvina.net wrote:
What a shame about the loss of chan_hype. I was really hoping to build
a .com around it.
At least I'm feeling better since starting the placebo treatment for
my allergies.
___
--
The Asterisk Development Team is pleased to announce the release of Asterisk
1.6.0.7. Asterisk 1.6.0.7 is available for immediate download at
http://downloads.digium.com/pub/asterisk/
This release resolves an issue where IMAP voicemail message retrieval and
Message Waiting Indication (MWI) would
On Wed, 2009-04-01 at 11:41 -0500, Brent Davidson wrote:
Cary Fitch wrote:
It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8
bits each, and that is 2^140^8, a nearly inexhaustible key number which is
related to audio and video data simultaneously stored on a Google
The Asterisk Development Team is pleased to announce the first beta of
Asterisk-Addons 1.6.2.0. Asterisk-Addons 1.6.2.0-beta1 is available for
immediate download at http://downloads.digium.com/pub/asterisk/
This beta fixes a several issues with chan_mobile from the chan_mobile
refactor branch,
On Wed, Apr 01, 2009 at 11:16:50PM +0200, Hans Witvliet wrote:
Wasn't that patented under the name of I2CA (Infinite Impropability
Compression Algorithm)...
It was far to technical for me, but afaicr is uses a key with a base of
42, Or was the exponent 42. can't remember, since then too busy
Sorry for replying for the second time, but this issue is interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this: http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data
I wonder why people don't get it ? X100P is a winmodem was and always will be.
Martin
On Wed, Apr 1, 2009 at 12:26 PM, Tim Nelson tnel...@rockbochs.com wrote:
If the primary purpose is to drive down cost, why not simply buy any one of
the existing 'Wildcard X100P' clone cards that are
Dear All,
Is anyone having luck with using some code for SIP network topology
hiding + NAT traversal (SBC functionality) with Asterisk ?
I tried OpenSBC but it didn't do NAT from Asterisk to ATA correctly.
It's in plans for OpenSIPS but it's not implemented yet ... checked
their svn.
Martin
We are planning to run an outbound only campaign. A 20-second voice message
will be played to callers and our dialer on machine1 will send to
machine2-asterisk (1.4) instructions to dial 400 calls, play the message and
hang up. This will be done for about 1 million phones.
The asterisk box will
Asterisk max call estimation doesn't scale linearly ... it might in the future
with some fixes they're adding.
For your application you could use some other open PBX that is known not to have
'Asterisk' limitations.
Anyways most people will tell you to simply buy a box and make a test.
Noone
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